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[alsa-devel] [PATCH v2 0/4] ASoC: tlv320aic32x4: DT support
by Markus Pargmann 22 Jan '14

22 Jan '14

Hi, The series adds basic DT support for the TI tlv320aic32x4 codec. v2 adds a new patch to support regulators. The codec now decides on the given regulators if LDO for AVDD should be enabled. 'ti,ldo-enable' was removed. Regards, Markus Changes in v2: - Regulator support - Fix documentation for gpio property Markus Pargmann (4): ASoC: tlv320aic32x4: Use signed int mixer controls ASoC: tlv320aic32x4: DT support ASoC: tlv320aic32x4: Support for master clock ASoC: tlv320aic32x4: Support for regulators .../devicetree/bindings/sound/tlv320aic32x4.txt | 28 +++++ sound/soc/codecs/tlv320aic32x4.c | 126 +++++++++++++++++++-- 2 files changed, 144 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt -- 1.8.5.2

3 8

[alsa-devel] [PATCH] ALSA: cs46xx: Fix memory leak at destructor
by Takashi Iwai 22 Jan '14

22 Jan '14

The release of module object itself was forgotten. Spotted by COVERIY CID 1162828. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/cs46xx/cs46xx_lib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index f18e5878f58b..062398ec5335 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -369,6 +369,7 @@ static void free_module_desc(struct dsp_module_desc *module) kfree(module->segments[i].data); kfree(module->segments); } + kfree(module); } /* firmware binary format: -- 1.8.5.2

1 0

[alsa-devel] [PATCH] alsa-utils: fix timespec to msec conversion
by Olivier Langlois 22 Jan '14

22 Jan '14

There is 1M nsec in a msec. Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> --- aplay/aplay.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index 69e8bda..33a8ed5 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -1509,7 +1509,7 @@ static void xrun(void) timermsub(&now, &tstamp, &diff); fprintf(stderr, _("%s!!! (at least %.3f ms long)\n"), stream == SND_PCM_STREAM_PLAYBACK ? _("underrun") : _("overrun"), - diff.tv_sec * 1000 + diff.tv_nsec / 10000000.0); + diff.tv_sec * 1000 + diff.tv_nsec / 1000000.0); #else fprintf(stderr, "%s !!!\n", _("underrun")); #endif -- 1.8.5.3

2 1

[alsa-devel] [PATCH - my module 1/1] control_hw: fix potential memory leak
by vanyacpp＠gmail.com 22 Jan '14

22 Jan '14

From: Ivan Sorokin <vanyacpp(a)gmail.com> Signed-off-by: Ivan Sorokin <vanyacpp(a)gmail.com> diff --git a/src/control/control_hw.c b/src/control/control_hw.c index 148097f..dfc9dcd 100644 --- a/src/control/control_hw.c +++ b/src/control/control_hw.c @@ -240,8 +240,10 @@ static int snd_ctl_hw_elem_tlv(snd_ctl_t *handle, int op_flag, return -errno; } if (op_flag == 0) { - if (xtlv->tlv[1] + 2 * sizeof(unsigned int) > tlv_size) + if (xtlv->tlv[1] + 2 * sizeof(unsigned int) > tlv_size) { + free(xtlv); return -EFAULT; + } memcpy(tlv, xtlv->tlv, xtlv->tlv[1] + 2 * sizeof(unsigned int)); } free(xtlv); -- 1.8.1.2

2 1

[alsa-devel] [ISSUE] crackles and clicks
by Eric F 22 Jan '14

22 Jan '14

Hello, I tried my DAC (Tenor TE8802 chipset) in USB on Archlinux (LTS 3.10.26). I can hear regularly crackles ans clicks. I've tried all available USB ports, other kernels (3.12.6, 3.12.7), other distro (ubuntu), without luck... The DAC is Teac UD-H01. here is lsusb verbose : http://ix.io/9Q4 Feel free to ask for more precisions, I'll give you all you need. I thank you very much. Eric

4 6

[alsa-devel] HD audio problem (fwd)
by Russ Whitaker 22 Jan '14

22 Jan '14

Still not heard from you. WHY? ---------- Forwarded message ---------- Date: Mon, 30 Dec 2013 20:20:31 -0800 (PST) From: Russ Whitaker <russ(a)ashlandhome.net> To: alsa-devel(a)alsa-project.org Subject: HD audio problem (fwd) Have not heard back from you. What should I do? ---------- Forwarded message ---------- Date: Thu, 19 Dec 2013 23:48:19 -0800 (PST) From: Russ Whitaker <russ(a)ashlandhome.net> To: alsa-devel(a)alsa-project.org Subject: HD audio problem Hi Bought a new mini-itx mb: ECS KBN-I On board sound chip: VT1705CF Installed kernel linux-3.10.24 Otherwise distro is Slackware64 14.1 mplayer plays .mp3 files just fine. No sound from video clips that use Adobe flashplayer, and I get this error message: ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave I think the problem is: card0 and card1 are interchanged as card0 is listed as the HDMI video output. How to fixit, please. Thank you, Russ

2 2

[alsa-devel] [PATCH 2/2] ASoC: fsl_ssi: Do not print 'baud clock' error message all the time
by Fabio Estevam 21 Jan '14

21 Jan '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Currently everytime we get the following message on boot: fsl-ssi-dai 202c000.ssi: could not get baud clock: -2 This is not really useful information to get on every boot, so make it a debug message instead. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 7b1772d..3155611 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1407,7 +1407,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) */ ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); if (IS_ERR(ssi_private->baudclk)) - dev_warn(&pdev->dev, "could not get baud clock: %ld\n", + dev_dbg(&pdev->dev, "could not get baud clock: %ld\n", PTR_ERR(ssi_private->baudclk)); else clk_prepare_enable(ssi_private->baudclk); -- 1.8.1.2

2 3

[alsa-devel] [PATCH] ASoC: fsl: use snd_soc_dai_init_dma_data()
by Xiubo Li 21 Jan '14

21 Jan '14

Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 4d075f1..8f36f49 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -911,8 +911,8 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) { struct fsl_spdif_priv *spdif_private = snd_soc_dai_get_drvdata(dai); - dai->playback_dma_data = &spdif_private->dma_params_tx; - dai->capture_dma_data = &spdif_private->dma_params_rx; + snd_soc_dai_init_dma_data(dai, &spdif_private->dma_params_tx, + &spdif_private->dma_params_rx); snd_soc_add_dai_controls(dai, fsl_spdif_ctrls, ARRAY_SIZE(fsl_spdif_ctrls)); -- 1.8.4

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[alsa-devel] [PATCH 1/3] ASoC: wm5102: Improve EQ coefficient controls
by Charles Keepax 21 Jan '14

21 Jan '14

The EQ coefficient binary controls overlapped with the volume controls for the B4 and B5 volumes, which were controllable from either the coefficient control or the volume control itself. This patch adds controls for the mode and moves the coefficient control to only cover the coefficients. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 17 ++++++++--------- 1 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a08e8bf..ac19f62 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -685,15 +685,8 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21, - ARIZONA_EQ1_ENA_MASK), -SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21, - ARIZONA_EQ2_ENA_MASK), -SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21, - ARIZONA_EQ3_ENA_MASK), -SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21, - ARIZONA_EQ4_ENA_MASK), - +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -705,6 +698,8 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 18), +SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -716,6 +711,8 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 18), +SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -727,6 +724,8 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 18), +SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, -- 1.7.2.5

2 3

[alsa-devel] [PATCH] ASoC: wm5110: Extend SYSCLK patch file for rev D
by Charles Keepax 21 Jan '14

21 Jan '14

Latest evaluation of the the device has given some patch file additions for improved performance. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 48 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 48 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 34fbb03..92f1b40 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -43,6 +43,54 @@ static const struct reg_default wm5110_sysclk_revd_patch[] = { { 0x3133, 0x1201 }, { 0x3183, 0x1501 }, { 0x31D3, 0x1401 }, + { 0x0049, 0x01ea }, + { 0x004a, 0x01f2 }, + { 0x0057, 0x01e7 }, + { 0x0058, 0x01fb }, + { 0x33ce, 0xc4f5 }, + { 0x33cf, 0x1361 }, + { 0x33d0, 0x0402 }, + { 0x33d1, 0x4700 }, + { 0x33d2, 0x026d }, + { 0x33d3, 0xff00 }, + { 0x33d4, 0x026d }, + { 0x33d5, 0x0101 }, + { 0x33d6, 0xc4f5 }, + { 0x33d7, 0x0361 }, + { 0x33d8, 0x0402 }, + { 0x33d9, 0x6701 }, + { 0x33da, 0xc4f5 }, + { 0x33db, 0x136f }, + { 0x33dc, 0xc4f5 }, + { 0x33dd, 0x134f }, + { 0x33de, 0xc4f5 }, + { 0x33df, 0x131f }, + { 0x33e0, 0x026d }, + { 0x33e1, 0x4f01 }, + { 0x33e2, 0x026d }, + { 0x33e3, 0xf100 }, + { 0x33e4, 0x026d }, + { 0x33e5, 0x0001 }, + { 0x33e6, 0xc4f5 }, + { 0x33e7, 0x0361 }, + { 0x33e8, 0x0402 }, + { 0x33e9, 0x6601 }, + { 0x33ea, 0xc4f5 }, + { 0x33eb, 0x136f }, + { 0x33ec, 0xc4f5 }, + { 0x33ed, 0x134f }, + { 0x33ee, 0xc4f5 }, + { 0x33ef, 0x131f }, + { 0x33f0, 0x026d }, + { 0x33f1, 0x4e01 }, + { 0x33f2, 0x026d }, + { 0x33f3, 0xf000 }, + { 0x33f6, 0xc4f5 }, + { 0x33f7, 0x1361 }, + { 0x33f8, 0x0402 }, + { 0x33f9, 0x4600 }, + { 0x33fa, 0x026d }, + { 0x33fb, 0xfe00 }, }; static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Add info print to note that the DSP has shutdown
by Charles Keepax 21 Jan '14

21 Jan '14

It can be useful for debugging purposes to see at what point the DSP has powered down, so add a message to inform us of this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4fbcab6..a15f75b 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1637,6 +1637,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, list_del(&alg_region->list); kfree(alg_region); } + + adsp_info(dsp, "Shutdown complete\n"); break; default: -- 1.7.2.5

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[alsa-devel] [alsa-plugins] [PATCH 2/2] Allow to specify a client name as parameters
by Corfu, ValentinX 21 Jan '14

21 Jan '14

0001-Modify-jack-client-name.patch is spitted in: 0001-Allow-number-of-channels.patch and 0002-Allow-to-specify-a-client-name-as-parameters-in-the-.patch >From 8d6d397306bdd09c6c3fae83543e124c89443e94 Mon Sep 17 00:00:00 2001 From: Valentin Corfu <valentinx.corfu(a)intel.com> Date: Tue, 21 Jan 2014 11:32:51 -0500 Subject: [PATCH 2/2] Allow to specify a client name as parameters The current jack client name contains the process id of the application providing the audio samples. This leads to unpredictable jack client names which makes handling of the connections by a controlling application very hard. This modification now, allows to specify a client name as parameters in the configuration file. The implementation is backward compatible and simply adds a new configuration option to the plugin. Signed-off-by: Valentin Corfu <valentinx.corfu(a)intel.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 4306a13..61ac197 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -32,6 +32,11 @@ typedef enum _jack_format { SND_PCM_JACK_FORMAT_RAW } snd_pcm_jack_format_t; +typedef enum _naming_scheme { + SND_PCM_JACK_STD_NAME, + SND_PCM_JACK_USER_DEFINED_NAME +} snd_pcm_jack_naming_t; + typedef struct { snd_pcm_ioplug_t io; @@ -325,7 +330,8 @@ static int make_nonblock(int fd) static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, snd_config_t *playback_conf, snd_config_t *capture_conf, - snd_pcm_stream_t stream, int mode) + snd_pcm_stream_t stream, int mode, + snd_pcm_jack_naming_t naming) { snd_pcm_jack_t *jack; int err; @@ -356,11 +362,19 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, return -EINVAL; } - if (snprintf(jack_client_name, sizeof(jack_client_name), "alsa-jack.%s%s.%d.%d", name, - stream == SND_PCM_STREAM_PLAYBACK ? "P" : "C", getpid(), num++) - >= (int)sizeof(jack_client_name)) { - fprintf(stderr, "%s: WARNING: JACK client name '%s' truncated to %d characters, might not be unique\n", - __func__, jack_client_name, (int)strlen(jack_client_name)); + if (naming == SND_PCM_JACK_STD_NAME ) { + if (snprintf(jack_client_name, sizeof(jack_client_name), "alsa-jack.%s%s.%d.%d", name, + stream == SND_PCM_STREAM_PLAYBACK ? "P" : "C", getpid(), num++) + >= (int)sizeof(jack_client_name)) { + fprintf(stderr, "%s: WARNING: JACK client name '%s' truncated to %d characters, might not be unique\n", + __func__, jack_client_name, (int)strlen(jack_client_name)); + } + } else { + if (snprintf(jack_client_name, sizeof(jack_client_name), "%s", name) + >= (int)sizeof(jack_client_name)) { + fprintf(stderr, "%s: WARNING: JACK client name '%s' truncated to %d characters, might not be unique\n", + __func__, jack_client_name, (int)strlen(jack_client_name)); + } } jack->client = jack_client_new(jack_client_name); @@ -417,6 +431,8 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) int err; long num_channels = 0; long chan; + const char *local_name = name; + snd_pcm_jack_naming_t naming = SND_PCM_JACK_STD_NAME; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -425,6 +441,11 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; + if (strcmp(id, "name") == 0) { + snd_config_get_string(n, &local_name); + naming = SND_PCM_JACK_USER_DEFINED_NAME; + continue; + } if (strcmp(id, "nchan") == 0) { snd_config_get_integer(n, &num_channels); continue; @@ -449,7 +470,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return -EINVAL; } - if ( (num_channels != 0) && (playback_conf == NULL) && (capture_conf == NULL) ) { + if ( (naming == SND_PCM_JACK_USER_DEFINED_NAME) + && (num_channels != 0) + && (playback_conf == NULL) + && (capture_conf == NULL) ) { err = snd_config_make_compound(&playback_conf, "playback_ports", 0); if (err) { SNDERR("Error during snd_config_make_compound(p): %s\n", strerror(err)); @@ -486,7 +510,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return err; } } - err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + err = snd_pcm_jack_open(pcmp, local_name, playback_conf, capture_conf, stream, mode, naming); if (err) { SNDERR("Error during snd_pcm_jack_open: %s\n", strerror(err)); return err; @@ -513,7 +537,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return err; } } else { - err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + err = snd_pcm_jack_open(pcmp, local_name, playback_conf, capture_conf, stream, mode, naming); } return err; -- 1.8.1.4 -------------------------------------------------------------- Intel Shannon Limited Registered in Ireland Registered Office: Collinstown Industrial Park, Leixlip, County Kildare Registered Number: 308263 Business address: Dromore House, East Park, Shannon, Co. Clare This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

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[alsa-devel] [alsa-plugins] [PATCH 1/2] Allow number of channels
by Corfu, ValentinX 21 Jan '14

21 Jan '14

0001-Modify-jack-client-name.patch is spitted in: 0001-Allow-number-of-channels.patch and 0002-Allow-to-specify-a-client-name-as-parameters-in-the-.patch >From d247e4d675dd68c6f683a8c09ff792172396e870 Mon Sep 17 00:00:00 2001 From: Valentin Corfu <valentinx.corfu(a)intel.com> Date: Tue, 21 Jan 2014 10:10:52 -0500 Subject: [PATCH 1/2] Allow number of channels The modification allows to specify the number of channels as parameters in the configuration file The given number of channels are then created but not connected. Signed-off-by: Valentin Corfu <valentinx.corfu(a)intel.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index c51fe84..4306a13 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -415,6 +415,8 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) snd_config_t *playback_conf = NULL; snd_config_t *capture_conf = NULL; int err; + long num_channels = 0; + long chan; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -423,6 +425,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; + if (strcmp(id, "nchan") == 0) { + snd_config_get_integer(n, &num_channels); + continue; + } if (strcmp(id, "playback_ports") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for %s", id); @@ -443,7 +449,72 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return -EINVAL; } - err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + if ( (num_channels != 0) && (playback_conf == NULL) && (capture_conf == NULL) ) { + err = snd_config_make_compound(&playback_conf, "playback_ports", 0); + if (err) { + SNDERR("Error during snd_config_make_compound(p): %s\n", strerror(err)); + return err; + } + err = snd_config_make_compound(&capture_conf, "capture_ports", 0); + if (err) { + SNDERR("Error during snd_config_make_compound(c): %s\n", strerror(err)); + return err; + } + char value[4]; + for (chan=0; chan<num_channels; ++chan) { + snprintf(value, sizeof(value), "%d", chan); + snd_config_t *newLeafPlayback; + snd_config_t *newLeafCapture; + err = snd_config_imake_integer(&newLeafPlayback, value, chan); + if (err) { + SNDERR("Error during snd_config_imake_string(p): %s\n", strerror(err)); + return err; + } + err = snd_config_add(playback_conf, newLeafPlayback); + if (err) { + SNDERR("Error during snd_config_add(p): %s\n", strerror(err)); + return err; + } + err = snd_config_imake_integer(&newLeafCapture, value, chan); + if (err) { + SNDERR("Error during snd_config_imake_string(c): %s\n", strerror(err)); + return err; + } + err = snd_config_add(capture_conf, newLeafCapture); + if (err) { + SNDERR("Error during snd_config_add(c): %s\n", strerror(err)); + return err; + } + } + err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + if (err) { + SNDERR("Error during snd_pcm_jack_open: %s\n", strerror(err)); + return err; + } + + err = snd_config_delete_compound_members(playback_conf); + if (err) { + SNDERR("Error during snd_config_delete_compound_members(p): %s\n", strerror(err)); + return err; + } + err = snd_config_delete_compound_members(capture_conf); + if (err) { + SNDERR("Error during snd_config_delete_compound_members(c): %s\n", strerror(err)); + return err; + } + err = snd_config_delete(playback_conf); + if (err) { + SNDERR("Error during snd_config_delete(p): %s\n", strerror(err)); + return err; + } + err = snd_config_delete(capture_conf); + if (err) { + SNDERR("Error during snd_config_delete(c): %s\n", strerror(err)); + return err; + } + } else { + err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + } return err; } -- 1.8.1.4 -------------------------------------------------------------- Intel Shannon Limited Registered in Ireland Registered Office: Collinstown Industrial Park, Leixlip, County Kildare Registered Number: 308263 Business address: Dromore House, East Park, Shannon, Co. Clare This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Resume mixer values properly
by Takashi Iwai 21 Jan '14

21 Jan '14

Implement reset_resume callback so that the mixer values are properly restored. Still no boot quirks are called, so it might not work well on some devices. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 18 ++++++++-- sound/usb/mixer.c | 99 ++++++++++++++++++++++++++++++++++++++++++++----------- sound/usb/mixer.h | 7 ++-- 3 files changed, 99 insertions(+), 25 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index d979050e6a6a..025224136129 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -691,12 +691,12 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) } list_for_each_entry(mixer, &chip->mixer_list, list) - snd_usb_mixer_inactivate(mixer); + snd_usb_mixer_suspend(mixer); return 0; } -static int usb_audio_resume(struct usb_interface *intf) +static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) { struct snd_usb_audio *chip = usb_get_intfdata(intf); struct usb_mixer_interface *mixer; @@ -711,7 +711,7 @@ static int usb_audio_resume(struct usb_interface *intf) * we just notify and restart the mixers */ list_for_each_entry(mixer, &chip->mixer_list, list) { - err = snd_usb_mixer_activate(mixer); + err = snd_usb_mixer_resume(mixer, reset_resume); if (err < 0) goto err_out; } @@ -723,9 +723,20 @@ static int usb_audio_resume(struct usb_interface *intf) err_out: return err; } + +static int usb_audio_resume(struct usb_interface *intf) +{ + return __usb_audio_resume(intf, false); +} + +static int usb_audio_reset_resume(struct usb_interface *intf) +{ + return __usb_audio_resume(intf, true); +} #else #define usb_audio_suspend NULL #define usb_audio_resume NULL +#define usb_audio_reset_resume NULL #endif /* CONFIG_PM */ static struct usb_device_id usb_audio_ids [] = { @@ -747,6 +758,7 @@ static struct usb_driver usb_audio_driver = { .disconnect = usb_audio_disconnect, .suspend = usb_audio_suspend, .resume = usb_audio_resume, + .reset_resume = usb_audio_reset_resume, .id_table = usb_audio_ids, .supports_autosuspend = 1, }; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 44b0ba4feab3..aa9bc19aae68 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2299,26 +2299,6 @@ requeue: } } -/* stop any bus activity of a mixer */ -void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer) -{ - usb_kill_urb(mixer->urb); - usb_kill_urb(mixer->rc_urb); -} - -int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) -{ - int err; - - if (mixer->urb) { - err = usb_submit_urb(mixer->urb, GFP_NOIO); - if (err < 0) - return err; - } - - return 0; -} - /* create the handler for the optional status interrupt endpoint */ static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer) { @@ -2417,3 +2397,82 @@ void snd_usb_mixer_disconnect(struct list_head *p) usb_kill_urb(mixer->urb); usb_kill_urb(mixer->rc_urb); } + +#ifdef CONFIG_PM +/* stop any bus activity of a mixer */ +static void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer) +{ + usb_kill_urb(mixer->urb); + usb_kill_urb(mixer->rc_urb); +} + +static int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) +{ + int err; + + if (mixer->urb) { + err = usb_submit_urb(mixer->urb, GFP_NOIO); + if (err < 0) + return err; + } + + return 0; +} + +int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer) +{ + snd_usb_mixer_inactivate(mixer); + return 0; +} + +static int restore_mixer_value(struct usb_mixer_elem_info *cval) +{ + int c, err, idx; + + if (cval->cmask) { + idx = 0; + for (c = 0; c < MAX_CHANNELS; c++) { + if (!(cval->cmask & (1 << c))) + continue; + if (cval->cached & (1 << c)) { + err = set_cur_mix_value(cval, c + 1, idx, + cval->cache_val[idx]); + if (err < 0) + return err; + } + idx++; + } + } else { + /* master */ + if (cval->cached) { + err = set_cur_mix_value(cval, 0, 0, *cval->cache_val); + if (err < 0) + return err; + } + } + + return 0; +} + +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) +{ + struct usb_mixer_elem_info *cval; + int id, err; + + /* FIXME: any mixer quirks? */ + + if (reset_resume) { + /* restore cached mixer values */ + for (id = 0; id < MAX_ID_ELEMS; id++) { + for (cval = mixer->id_elems[id]; cval; + cval = cval->next_id_elem) { + err = restore_mixer_value(cval); + if (err < 0) + return err; + } + } + } + + return snd_usb_mixer_activate(mixer); +} +#endif diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index aab80df201bd..73b1f649447b 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -63,8 +63,6 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid); int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set); -void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer); -int snd_usb_mixer_activate(struct usb_mixer_interface *mixer); int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, struct snd_kcontrol *kctl); @@ -72,4 +70,9 @@ int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *_tlv); +#ifdef CONFIG_PM +int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer); +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume); +#endif + #endif /* __USBMIXER_H */ -- 1.8.5.2

1 0

Re: [alsa-devel] [PATCH] ASoC: da9055: Partially fix device
by Opensource [Adam Thomson] 21 Jan '14

21 Jan '14

On Wed, 8 Jan 2014 20:38:55 +0000, Mark Brown wrote: > It has never been possible to even load the da9055 driver since both the > MFD and CODEC drivers register the I2C device "da9055". Fix this by using > the usual pattern for MFDs with multiple I2C addresses and having the CODEC > driver reference an I2C dummy registered by the MFD. > Since I don't know which I2C address to use for the CODEC a FIXME has been > left in the MFD, this doesn't make anything any worse since the device has > never been able to load in the first place. What you're saying here is incorrect, as previously discussed at length in this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-September/066027.… The two drivers did work together fine until someone in 3.12 changed the PMIC I2C Id back to just 'da9055' and broke it again. Also, I wanted to add a simple patch to tidy up the I2C Id names and avoid this but you would not agree. The codec part can be standalone and by doing what you've done here means that customers would need to now add additional, unnecessary code to be able to use just the codec driver. This is not right to me.

2 11

[alsa-devel] [PATCH] ASoC: omap: Make RX51 depend on GPIOLIB due to jack usage
by Mark Brown 21 Jan '14

21 Jan '14

From: Mark Brown <broonie(a)linaro.org> Since the GPIO jacks are only supported if gpiolib is built and fail to compile otherwise add a build depedency. This is unlikely to have any practical impact outside of coverage testing. Reported-by: Russell King <linux(a)arm.linux.org.uk> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/omap/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 4a07f7179690..22ad9c5654b5 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -30,6 +30,7 @@ config SND_OMAP_SOC_RX51 select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 + depends on GPIOLIB help Say Y if you want to add support for SoC audio on Nokia RX-51 hardware. This is also known as Nokia N900 product. -- 1.8.5.2

2 1

[alsa-devel] ALSA: hda - Fix possible races in HDMI driver - lockup on shutdown when radeon.audio=1 after using audacity
by Arthur Marsh 21 Jan '14

21 Jan '14

I have had reproducible lock-ups on shut-down (at the shutting down ALSA stage) of my AMD64 machine (Asus M3A78Pro motherboard, BIOS 1701 01/27/2011, CPU AMD Athlon(tm) II X4 640 Processor) running the 64 bit Linux kernel more recent than 3.12 when *both* radeon.audio=1 was set and I had been running audacity 2.0.5. (iommu=noaperture is also set). The problem was reproducible with the stock Debian kernel linux-image-3.13-rc6-amd64 version 3.13~rc6-1~exp1. The machine is using an ATI/AMD 3850HD video card with a DVI cable to a DVI input on my monitor, and the default audio device is the motherboard's on-board audio device: 00:14.2 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] SBx00 Azalia (Intel HDA) 01:00.0 VGA compatible controller: Advanced Micro Devices, Inc. [AMD/ATI] RV670 [Radeon HD 3690/3850] 01:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] RV670/680 HDMI Audio [Radeon HD 3690/3800 Series] $ git bisect bad cbbaa603a03cc46681e24d6b2804b62fde95a2af is the first bad commit commit cbbaa603a03cc46681e24d6b2804b62fde95a2af Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Oct 17 18:03:24 2013 +0200 ALSA: hda - Fix possible races in HDMI driver Some per_pin fields and ELD contents might be changed dynamically in multiple ways where the concurrent accesses are still opened in the current code. This patch fixes such possible races by using eld->lock in appropriate places. Reported-by: Anssi Hannula <anssi.hannula(a)iki.fi> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> :040000 040000 0c29281f82a3ebd9a704d481114f9cfcefea07c8 d71fd101125cd29a628cb5e66c7ee4f56e28329b M sound When running audacity from the command line there was the following output: ALSA lib pcm_dsnoop.c:618:(snd_pcm_dsnoop_open) unable to open slave ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave Expression 'stream->playback.pcm' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 Expression 'stream->playback.pcm' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 Expression 'stream->playback.pcm' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 I am happy to supply further information or run further tests to help in isolating the problem and verifying a solution. Arthur.

2 5

[alsa-devel] [PATCH] ASoC: wm8804: Allow control of master clock divider in PLLgeneration
by Daniel Matuschek 20 Jan '14

20 Jan '14

WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using an additional mclk_div divider, it is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. The behavior has to be controlled by the sound card driver, because some sample frequency share the same setting. e.g. 192kHz and 96kHz use 24.576MHz master clock. The only difference is the MCLK divider. Signed-off-by: Daniel Matuschek <daniel(a)matuschek.net> Acked-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Tested-by: Florian Meier <florian.meier(a)koalo.de> --- sound/soc/codecs/wm8804.c | 17 ++++++++++++++--- sound/soc/codecs/wm8804.h | 4 ++++ 2 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 9bc8206..4163786 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -63,6 +63,7 @@ struct wm8804_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; + int mclk_div; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -318,7 +319,7 @@ static struct { #define FIXED_PLL_SIZE ((1ULL << 22) * 10) static int pll_factors(struct pll_div *pll_div, unsigned int target, - unsigned int source) + unsigned int source, unsigned int mclk_div) { u64 Kpart; unsigned long int K, Ndiv, Nmod, tmp; @@ -330,7 +331,8 @@ static int pll_factors(struct pll_div *pll_div, unsigned int target, */ for (i = 0; i < ARRAY_SIZE(post_table); i++) { tmp = target * post_table[i].div; - if (tmp >= 90000000 && tmp <= 100000000) { + if ((tmp >= 90000000 && tmp <= 100000000) && + (mclk_div == post_table[i].mclkdiv)) { pll_div->freqmode = post_table[i].freqmode; pll_div->mclkdiv = post_table[i].mclkdiv; target *= post_table[i].div; @@ -387,8 +389,12 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, } else { int ret; struct pll_div pll_div; + struct wm8804_priv *wm8804; - ret = pll_factors(&pll_div, freq_out, freq_in); + wm8804 = snd_soc_codec_get_drvdata(codec); + + ret = pll_factors(&pll_div, freq_out, freq_in, + wm8804->mclk_div); if (ret) return ret; @@ -452,6 +458,7 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { struct snd_soc_codec *codec; + struct wm8804_priv *wm8804; codec = dai->codec; switch (div_id) { @@ -459,6 +466,10 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, snd_soc_update_bits(codec, WM8804_PLL5, 0x30, (div & 0x3) << 4); break; + case WM8804_MCLK_DIV: + wm8804 = snd_soc_codec_get_drvdata(codec); + wm8804->mclk_div = div; + break; default: dev_err(dai->dev, "Unknown clock divider: %d\n", div_id); return -EINVAL; diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index 8ec14f5..e72d4f4 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -57,5 +57,9 @@ #define WM8804_CLKOUT_SRC_OSCCLK 4 #define WM8804_CLKOUT_DIV 1 +#define WM8804_MCLK_DIV 2 + +#define WM8804_MCLKDIV_256FS 0 +#define WM8804_MCLKDIV_128FS 1 #endif /* _WM8804_H */ -- 1.8.3.2

1 0

[alsa-devel] ASoC updates for v3.14
by Mark Brown 20 Jan '14

20 Jan '14

The following changes since commit 258753361534a40ad7180c742da813fc659e427b: Merge branch 'for-next' into for-linus (2014-01-20 10:20:14 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-3 for you to fetch changes up to ac6d7c48e3a5e82e876114b9ca0b489e96997743: Merge remote-tracking branch 'asoc/topic/compress' into asoc-next (2014-01-20 11:50:41 +0000) ---------------------------------------------------------------- ASoC: Last updates for the merge window A couple more fixes plus some extensions to DPCM for use with compressed audio from Liam which arrived just after my previous pull request. ---------------------------------------------------------------- Arun Shamanna Lakshmi (1): ASoC: dapm: Fix double prefix addition Liam Girdwood (2): ASoC: DPCM: make some DPCM API calls non static for compressed usage ASoC: compress: Add suport for DPCM into compressed audio Mark Brown (3): Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/compress' into asoc-next Xiubo Li (1): ASoC: core: Fix possible NULL pointer dereference of pcm->config include/sound/soc-dpcm.h | 22 +++ include/sound/soc.h | 1 + sound/soc/soc-compress.c | 301 +++++++++++++++++++++++++++++++++- sound/soc/soc-dapm.c | 11 +- sound/soc/soc-generic-dmaengine-pcm.c | 9 +- sound/soc/soc-pcm.c | 29 ++-- 6 files changed, 347 insertions(+), 26 deletions(-)

2 1

[alsa-devel] [PATCH v3 01/16] ALSA: Oxygen: Add the separate SPI waiting function
by Roman Volkov 20 Jan '14

20 Jan '14

This function performs waiting when the SPI bus completes a transaction. Timeout error checking introduced and the timeout increased to 400 from 40. Signed-off-by: Roman Volkov <v1ron(a)mail.ru> --- sound/pci/oxygen/oxygen_io.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 521eae4..a917f37 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -194,6 +194,21 @@ void oxygen_write_ac97_masked(struct oxygen *chip, unsigned int codec, } EXPORT_SYMBOL(oxygen_write_ac97_masked); +static int oxygen_wait_spi(struct oxygen *chip) +{ + /* Higher timeout to be sure: 400 microseconds */ + unsigned int count; + for (count = 100; count > 0; count--) { + if ((oxygen_read8(chip, OXYGEN_SPI_CONTROL) & + OXYGEN_SPI_BUSY) == 0) + return 0; + udelay(4); + --count; + } + snd_printk(KERN_ERR "Oxygen: SPI wait timeout\n"); + return -EIO; +} + void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data) { unsigned int count; -- 1.7.10.4

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[alsa-devel] Thinkpad T520 (Low Volume)
by Raymond Yau 20 Jan '14

20 Jan '14

> 2014/1/16 Axel Braun <axel.braun(a)gmx.de> > same problem on a ThinkPad T520, sound volume is way too low - watching movie via loudspeakers is fairly impossible. > Any hint you could give me? > > post the output of alsa-info.sh , there is a bug in alsa-driver which treat > dock headphone and headphone to support surround 4.0 you get low volume because Amp Out vals are 0x3d instead of 0x4a but speaker volumes are at 100% (0dB) Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 74 [100%] [0.00dB] [on] Front Right: Playback 74 [100%] [0.00dB] [on] Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CX20590 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x3d 0x3d] https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=ca2… thinkpad t520 was supported with CXT5066_AUTO which use cxt5066_update_speaker which check port a and port d Do this mean thinkpad don't support surround 4.0 by dock headphone and headphone ? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… ALSA: hda/conexant - Set up the missing docking-station pins [ 9.956636] ALSA hda_auto_parser.c:393 autoconfig: line_outs=1 (0x1f/0x0/0x0/0x0/0x0) type:speaker [ 9.956639] ALSA hda_auto_parser.c:397 speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 9.956641] ALSA hda_auto_parser.c:401 hp_outs=2 (0x1c/0x19/0x0/0x0/0x0) [ 9.956643] ALSA hda_auto_parser.c:402 mono: mono_out=0x0 [ 9.956644] ALSA hda_auto_parser.c:406 inputs: [ 9.956646] ALSA hda_auto_parser.c:410 Mic=0x1b [ 9.956647] ALSA hda_auto_parser.c:410 Internal Mic=0x23 [ 9.956649] ALSA hda_auto_parser.c:410 Dock Mic=0x1a the driver select this as best config ==> Best config: lo_type=1, wired=1, mio=1 multi_outs = 1f/0/0/0 : 10/0/0/0 (type SP) out path: depth=2 :10:1f hp_outs = 19/1c/0/0 : 11/11/0/0 hp path: depth=2 :11:19 hp path: depth=2 :11:1c https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… ALSA: hda - Fix multi-io channel mode management this patch change the the channel_count from 2 to 4 since there are two headphone + if (cfg->line_out_type != AUTO_PIN_HP_OUT) + spec->const_channel_count = max(spec->const_channel_count, + cfg->hp_outs * 2); if you don't have docking station , the workaround is to remove those pin fix of node 0x1a and 0x1c

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[alsa-devel] valgrind error in snd_tlv_get_dB_range
by Ivan Sorokin 20 Jan '14

20 Jan '14

One program (qasmixer -- alsa mixer) on my system prints lots of valgrind errors. The first one is this: ==25716== Conditional jump or move depends on uninitialised value(s) ==25716== at 0x4E72307: snd_tlv_get_dB_range (tlv.c:170) ==25716== by 0x4E8A746: get_dB_range (simple_none.c:1162) ==25716== by 0x4E8A7E7: get_dB_range_ops (simple_none.c:1176) ==25716== by 0x4E85590: snd_mixer_selem_get_playback_dB_range (simple.c:298) ==25716== by 0x472B2B: QSnd::Mixer_Simple_Elem::set_snd_mixer_selem_id(_snd_mixer_selem_id*) (mixer_simple_elem.cpp:161) ==25716== by 0x4724B8: QSnd::Mixer_Simple_Elem::Mixer_Simple_Elem(QObject*, _snd_mixer*, _snd_mixer_selem_id*) (mixer_simple_elem.cpp:31) ==25716== by 0x479F17: QSnd::Mixer_Simple::create_mixer_elem(_snd_mixer_elem*) (mixer_simple.cpp:318) ==25716== by 0x4797B3: QSnd::Mixer_Simple::create_mixer_elems() (mixer_simple.cpp:217) ==25716== by 0x4795DF: QSnd::Mixer_Simple::open(QString const&) (mixer_simple.cpp:178) ==25716== by 0x4D4754: Tray_Mixer::load_mixer() (tray_mixer.cpp:222) ==25716== by 0x4D4505: Tray_Mixer::set_mdev_setup(Tray_Mixer_MDev_Setup*) (tray_mixer.cpp:137) ==25716== by 0x4DB863: Desktop_Items::tray_mixer_create() (desktop_items.cpp:504) ==25716== Uninitialised value was created by a heap allocation ==25716== at 0x4C2CD7B: malloc (in /usr/lib/valgrind/vgpreload_memcheck-amd64-linux.so) ==25716== by 0x4E8A572: init_db_range (simple_none.c:1114) ==25716== by 0x4E8A70D: get_dB_range (simple_none.c:1159) ==25716== by 0x4E8A7E7: get_dB_range_ops (simple_none.c:1176) ==25716== by 0x4E85590: snd_mixer_selem_get_playback_dB_range (simple.c:298) ==25716== by 0x472B2B: QSnd::Mixer_Simple_Elem::set_snd_mixer_selem_id(_snd_mixer_selem_id*) (mixer_simple_elem.cpp:161) ==25716== by 0x4724B8: QSnd::Mixer_Simple_Elem::Mixer_Simple_Elem(QObject*, _snd_mixer*, _snd_mixer_selem_id*) (mixer_simple_elem.cpp:31) ==25716== by 0x479F17: QSnd::Mixer_Simple::create_mixer_elem(_snd_mixer_elem*) (mixer_simple.cpp:318) ==25716== by 0x4797B3: QSnd::Mixer_Simple::create_mixer_elems() (mixer_simple.cpp:217) ==25716== by 0x4795DF: QSnd::Mixer_Simple::open(QString const&) (mixer_simple.cpp:178) ==25716== by 0x4D4754: Tray_Mixer::load_mixer() (tray_mixer.cpp:222) ==25716== by 0x4D4505: Tray_Mixer::set_mdev_setup(Tray_Mixer_MDev_Setup*) (tray_mixer.cpp:137) I compiled libasound2 from the latest git from git://gitorious.org/alsa/alsa-lib.git with -g -O0, so you can see line numbers. As I understand the problem is that init_db_range initialize invalid rec->db_info, because snd_hctl_elem_tlv_read failed to fully initialize tlv: > tlv = malloc(tlv_size); // line 1114 > if (! tlv) > return -ENOMEM; > if (snd_hctl_elem_tlv_read(ctl, tlv, tlv_size) < 0) > goto error; > db_size = snd_tlv_parse_dB_info(tlv, tlv_size, &dbrec); > if (db_size < 0) > goto error; > rec->db_info = malloc(db_size); > if (!rec->db_info) > goto error; > memcpy(rec->db_info, dbrec, db_size); > free(tlv); I tried to trace problem in snd_hctl_elem_tlv_read, but I failed to do so due to lack of experience with alsa-lib. Could someone more experienced with alsa-lib look at the problem?

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[alsa-devel] Realtec ALC898 supported?
by Haug Bürger 20 Jan '14

20 Jan '14

Hi, I would like to know if the new Realtec 89x chips are supported. I found drivers on the Realtec site but the seem to be older than the alsa-project.org drivers and I'm not sure if they support the 89x chips. thanks Haug

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[alsa-devel] [PATCH] ASoC: core: Use devm_kzalloc() instead kzalloc()
by Xiubo Li 20 Jan '14

20 Jan '14

Makes the code slightly shorter Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 5bace12..bfb012f 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -388,7 +388,7 @@ int snd_dmaengine_pcm_register(struct device *dev, struct dmaengine_pcm *pcm; int ret; - pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); + pcm = devm_kzalloc(dev, sizeof(*pcm), GFP_KERNEL); if (!pcm) return -ENOMEM; @@ -408,7 +408,6 @@ int snd_dmaengine_pcm_register(struct device *dev, err_free_dma: dmaengine_pcm_release_chan(pcm); - kfree(pcm); return ret; } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_register); @@ -433,7 +432,6 @@ void snd_dmaengine_pcm_unregister(struct device *dev) snd_soc_remove_platform(platform); dmaengine_pcm_release_chan(pcm); - kfree(pcm); } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister); -- 1.8.4

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[alsa-devel] [PATCH] ASoC: dapm: Fix double prefix addition
by Arun Shamanna Lakshmi 17 Jan '14

17 Jan '14

The prefix for the codec driver can be used during dual identical codec usecases. However, dapm adds prefix twice for codec DAI widget in snd_soc_dapm_add_route API. This change is to avoid double prefix addition for codec DAI widget and is needed while using identical dual codecs. Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> Signed-off-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> --- sound/soc/soc-dapm.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2a44fe9..dc8ff13 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2476,7 +2476,8 @@ err: } static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_route *route) + const struct snd_soc_dapm_route *route, + unsigned int is_prefixed) { struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; struct snd_soc_dapm_widget *wtsource = NULL, *wtsink = NULL; @@ -2486,7 +2487,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, char prefixed_source[80]; int ret; - if (dapm->codec && dapm->codec->name_prefix) { + if (dapm->codec && dapm->codec->name_prefix && !is_prefixed) { snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", dapm->codec->name_prefix, route->sink); sink = prefixed_sink; @@ -2614,7 +2615,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { - r = snd_soc_dapm_add_route(dapm, route); + r = snd_soc_dapm_add_route(dapm, route, false); if (r < 0) { dev_err(dapm->dev, "ASoC: Failed to add route %s -> %s -> %s\n", route->source, @@ -3670,7 +3671,7 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) cpu_dai->codec->name, r.source, codec_dai->platform->name, r.sink); - snd_soc_dapm_add_route(&card->dapm, &r); + snd_soc_dapm_add_route(&card->dapm, &r, true); } /* connect BE DAI capture if widgets are valid */ @@ -3681,7 +3682,7 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) codec_dai->codec->name, r.source, cpu_dai->platform->name, r.sink); - snd_soc_dapm_add_route(&card->dapm, &r); + snd_soc_dapm_add_route(&card->dapm, &r, true); } } -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ASoC: wm8804: Allow control of master clock divider in PLL generation
by Florian Meier 17 Jan '14

17 Jan '14

On 01/17/2014 07:33 PM, Mark Brown wrote: > On Fri, Jan 17, 2014 at 07:06:24PM +0100, Florian Meier wrote: >> > Intentionally off-list? Oh no - I am sorry! >> If I remember correctly the error was >> "codec can not start from non-off bias with idle_bias_off==true" > >> I think the solution is just to set idle_bias_off = false and everything >> seems to be working with that. I just don't know if there might be any >> side effects. > > Setting it to false increases power consumption since the device is > kept more powered on when idle but reduces startup time from idle. For > digital only devices like the wm8804 there shouldn't be any reason to > keep it powered up when not in use, the startup time is generally > negligable anyway. > So a better solution would be to set SND_SOC_BIAS_OFF instead of SND_SOC_BIAS_STANDBY at the end of probe?

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[alsa-devel] [PATCH 0/4] ASoC: tlv320aic32x4: DT support
by Markus Pargmann 17 Jan '14

17 Jan '14

Hi, Here are some more updates for the tlv320aic32x4 codec. This series fixes the mixer controls to operate on the full range instead of only the positive range. This depends on my other series: "ASoC: core: volume control using signed register values" Patch 3 adds DT support. Patch 4 adds support to control the master clock connected to the codec. Regards, Markus Markus Pargmann (4): ASoC: tlv320aic32x4: Use signed int mixer controls ASoC: tlv320aic32x4: Use gpio_is_valid ASoC: tlv320aic32x4: DT support ASoC: tlv320aic32x4: Support for master clock .../devicetree/bindings/sound/tlv320aic32x4.txt | 25 +++++++ sound/soc/codecs/tlv320aic32x4.c | 83 ++++++++++++++++++---- 2 files changed, 96 insertions(+), 12 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt -- 1.8.5.2

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[alsa-devel] [PATCH v2 0/2] Dynamic routing for compressed streams.
by Liam Girdwood 17 Jan '14

17 Jan '14

Changes for V2 * We no longer copy the compressed params into the BE hw_params. This was bogus, but didn't show up on Haswell testing as the BE params were never used. The DSP ran the BE is a fixed configuration so all BE params were ignored. * The BE hw_params are now zeroed to force the machine driver to fixup the BE configuration. i.e. machine driver can query the DSP encodec/decoder for correct hw_params given the use case. * No longer need to allocate memory for hw_params. Regards Liam

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[alsa-devel] [PATCH v2 1/2] ASoC: DPCM: make some DPCM API calls non static for compressed usage
by Liam Girdwood 17 Jan '14

17 Jan '14

The ASoC compressed code needs to call the internal DPCM APIs in order to dynamically route compressed data to different DAIs. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dpcm.h | 22 ++++++++++++++++++++++ sound/soc/soc-pcm.c | 29 ++++++++++++----------------- 2 files changed, 34 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 047d657..2883a7a 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -11,6 +11,7 @@ #ifndef __LINUX_SND_SOC_DPCM_H #define __LINUX_SND_SOC_DPCM_H +#include <linux/slab.h> #include <linux/list.h> #include <sound/pcm.h> @@ -135,4 +136,25 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute); int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd); int soc_dpcm_runtime_update(struct snd_soc_card *); +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list_); +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list, int new); +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream); +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream); +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int tream); +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd); +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, + int event); + +static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) +{ + kfree(*list); +} + + #endif diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1534115..7023f7e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -58,7 +58,7 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); /* DPCM stream event, send event to FE and all active BEs. */ -static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) { struct snd_soc_dpcm *dpcm; @@ -877,7 +877,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, } /* disconnect a BE and FE */ -static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; @@ -973,7 +973,7 @@ static int widget_in_list(struct snd_soc_dapm_widget_list *list, return 0; } -static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { struct snd_soc_dai *cpu_dai = fe->cpu_dai; @@ -995,11 +995,6 @@ static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, return paths; } -static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) -{ - kfree(*list); -} - static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { @@ -1069,7 +1064,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; /* don't connect if FE is not running */ - if (!fe->dpcm[stream].runtime) + if (!fe->dpcm[stream].runtime && !fe->fe_compr) continue; /* newly connected FE and BE */ @@ -1094,7 +1089,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, * Find the corresponding BE DAIs that source or sink audio to this * FE substream. */ -static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list, int new) { if (new) @@ -1103,7 +1098,7 @@ static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, return dpcm_prune_paths(fe, stream, list); } -static void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1141,7 +1136,7 @@ static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, } } -static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int err, count = 0; @@ -1293,7 +1288,7 @@ be_err: return ret; } -static int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1354,7 +1349,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1419,7 +1414,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret; @@ -1549,7 +1544,7 @@ static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, return ret; } -static int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; @@ -1717,7 +1712,7 @@ out: return ret; } -static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret = 0; -- 1.8.3.2

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[alsa-devel] ProVideo PV 918A - audi o device created but cannot capture sound
by siddharth 17 Jan '14

17 Jan '14

Hi, I have installed a ProVideo PV 981. I was able to get the sound device detected by putting modprobe snd-bt87x load_all=1 digital_rate=48000 enable=1 but am unable to capture sound. I am using Fedora 19 on an intel i5 machine. In dmesg i get the error: [ 235.145932] ALSA sound/pci/bt87x.c:860 unknown card 0x878-0xaa00:0x1460 [ 235.145935] ALSA sound/pci/bt87x.c:863 please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> [ 235.146061] ALSA sound/pci/bt87x.c:931 bt87x0: Using board 0, analog, digital (rate 48000 Hz) Any help will be appreciated. regards Siddharth

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Re: [alsa-devel] [pulseaudio-discuss] Need help with audio gain on an Intel HDA equipt ASUS M2N-SLI Deluxe
by Raymond Yau 17 Jan '14

17 Jan '14

2014/1/16 Axel Braun <axel.braun(a)gmx.de> > Am Donnerstag, 16. Januar 2014, 16:23:19 schrieb Raymond Yau: > > Any hint you could give me? > > > > post the output of alsa-info.sh , there is a bug in alsa-driver which > treat > > dock headphone and headphone to support surround 4.0 > > /sys/class/sound/hwC0D0/init_pin_configs: 0x19 0x04211040 0x1a 0x61a19050 0x1b 0x04a11060 0x1c 0x6121401f 0x1d 0x40f001f0 0x1e 0x40f001f0 0x1f 0x90170110 0x20 0x40f001f0 0x22 0x40f001f0 0x23 0x90a60170 /sys/class/sound/hwC0D0/driver_pin_configs: 0x19 0x042110ff 0x1a 0x21a190f0 0x1c 0x212140ff https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… it was using auto but now the driver modify the dock headphone , dock mic and headphone pin default. there are also changes in the thinkpad_acpi module too. https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… do this thinkpad support headset or headphone ? it is strange that speaker volume is at max but amp out vals at node 0x11 is 0x3d instead of 0x4a (0dB) Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 74 Mono: Playback 61 [82%] [-13.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 0 [0%] [-74.00dB] [off] Front Right: Playback 0 [0%] [-74.00dB] [off] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 74 [100%] [0.00dB] [on] Front Right: Playback 74 [100%] [0.00dB] [on] Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CX20590 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x3d 0x3d] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x80 0x80] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0

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[alsa-devel] [PATCH 1/2] ASoC: DPCM: make some DPCM API calls non static for compressed usage
by Liam Girdwood 17 Jan '14

17 Jan '14

The ASoC compressed code needs to call the internal DPCM APIs in order to dynamically route compressed data to different DAIs. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dpcm.h | 22 ++++++++++++++++++++++ sound/soc/soc-pcm.c | 29 ++++++++++++----------------- 2 files changed, 34 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 047d657..2883a7a 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -11,6 +11,7 @@ #ifndef __LINUX_SND_SOC_DPCM_H #define __LINUX_SND_SOC_DPCM_H +#include <linux/slab.h> #include <linux/list.h> #include <sound/pcm.h> @@ -135,4 +136,25 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute); int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd); int soc_dpcm_runtime_update(struct snd_soc_card *); +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list_); +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list, int new); +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream); +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream); +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int tream); +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd); +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, + int event); + +static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) +{ + kfree(*list); +} + + #endif diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1534115..7023f7e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -58,7 +58,7 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); /* DPCM stream event, send event to FE and all active BEs. */ -static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) { struct snd_soc_dpcm *dpcm; @@ -877,7 +877,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, } /* disconnect a BE and FE */ -static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; @@ -973,7 +973,7 @@ static int widget_in_list(struct snd_soc_dapm_widget_list *list, return 0; } -static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { struct snd_soc_dai *cpu_dai = fe->cpu_dai; @@ -995,11 +995,6 @@ static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, return paths; } -static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) -{ - kfree(*list); -} - static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { @@ -1069,7 +1064,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; /* don't connect if FE is not running */ - if (!fe->dpcm[stream].runtime) + if (!fe->dpcm[stream].runtime && !fe->fe_compr) continue; /* newly connected FE and BE */ @@ -1094,7 +1089,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, * Find the corresponding BE DAIs that source or sink audio to this * FE substream. */ -static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list, int new) { if (new) @@ -1103,7 +1098,7 @@ static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, return dpcm_prune_paths(fe, stream, list); } -static void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1141,7 +1136,7 @@ static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, } } -static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int err, count = 0; @@ -1293,7 +1288,7 @@ be_err: return ret; } -static int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1354,7 +1349,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1419,7 +1414,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret; @@ -1549,7 +1544,7 @@ static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, return ret; } -static int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; @@ -1717,7 +1712,7 @@ out: return ret; } -static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret = 0; -- 1.8.3.2

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[alsa-devel] [PATCH v4 0/9] ASoC: fsl-ssi: offline/online configuration and cleanups
by Markus Pargmann 17 Jan '14

17 Jan '14

Hi, This patch series fixes some fsl-ssi code that does not act exactly as it is described in the reference manuals. The reference manuals before imx51 do mention that some register bits, including TDMAE/RDMAE, should not be changed while the SSI unit is enabled (SSIEN). At the same time the SDMA unit has a undefined request handling if there are DMA requests before the SDMA engine and its channel is configured. The reference manual states that SSI DMA requests are sent as soon as the FIFO and the DMAE bits are enabled. It does not mention a dependency to SCR's TE/RE bit. To avoid sending DMA requests before a channel is configured, we have to seperate the fsl-ssi behavior into offline and online configuration. SoCs before imx51 have to use offline configuration while later SoC versions can use online reconfiguration. This series adds support for online configuration and cleans up the configuration register code especially in fsl_ssi_trigger. At the end we have a seperation between enable/disable logic and the actual configuration register values used to enable/disable TX/RX. v4 is another rebase onto latest topic/fsl branch. It includes the DAI master mode changes from Nicolin. @Nicolin: Could you please test this series with your hardware to verify that your patch still works? I tested this series on mx53. The series is based on Mark Brown's topic/fsl branch. Regards, Markus Changes in v4: - Support for irqstats on all imx SoCs - Fix offline_config and use switch statement - Introduce ssi_private->irq_stats bool Changes in v3: - Rebase on latest topic/fsl - Added patch to cleanup probe function error handling - Fix interrupt stats for imx53 - Replace imx50-ssi by imx51-ssi Markus Pargmann (9): ASoC: fsl-ssi: Fix probe error handling ASoC: fsl-ssi: Move sysfs stats to debugfs ASoC: fsl-ssi: Add imx51-ssi and of_device_id matching ASoC: fsl-ssi: Fix interrupt stats for imx ASoC: fsl-ssi: Add offline_config flag ASoC: fsl-ssi: Add configuration helper functions ASoC: fsl-ssi: Move RX/TX configuration to seperate functions ASoC: fsl-ssi: Drop ac97 specific trigger function ARM: DTS: imx5* imx6*, use imx51-ssi arch/arm/boot/dts/imx53.dtsi | 10 +- arch/arm/boot/dts/imx6qdl.dtsi | 12 +- arch/arm/boot/dts/imx6sl.dtsi | 12 +- sound/soc/fsl/fsl_ssi.c | 567 ++++++++++++++++++++++++++++------------- 4 files changed, 411 insertions(+), 190 deletions(-) -- 1.8.5.1

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[alsa-devel] [PATCH] ASoC: core: Fix possible NULL pointer dereference of pcm->config
by Xiubo Li 16 Jan '14

16 Jan '14

Since the soc generic dmaengine pcm driver allows using the defualt settings, so the pcm->config maybe NULL. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 560a778..5bace12 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -176,17 +176,20 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); struct snd_dmaengine_dai_dma_data *dma_data; + dma_filter_fn fn = NULL; dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) return pcm->chan[0]; - if (pcm->config->compat_request_channel) + if (pcm->config && pcm->config->compat_request_channel) return pcm->config->compat_request_channel(rtd, substream); - return snd_dmaengine_pcm_request_channel(pcm->config->compat_filter_fn, - dma_data->filter_data); + if (pcm->config) + fn = pcm->config->compat_filter_fn; + + return snd_dmaengine_pcm_request_channel(fn, dma_data->filter_data); } static bool dmaengine_pcm_can_report_residue(struct dma_chan *chan) -- 1.8.4

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[alsa-devel] ASoC fixes for v3.13
by Mark Brown 16 Jan '14

16 Jan '14

The following changes since commit 7e22e91102c6b9df7c4ae2168910e19d2bb14cd6: Linux 3.13-rc8 (2014-01-12 17:04:18 +0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc8 for you to fetch changes up to c5281227c90578c7b7beffca56e8c3a68d4607de: Merge remote-tracking branches 'asoc/fix/adau1701', 'asoc/fix/fsl' and 'asoc/fix/tlv320aic32x4' into asoc-linus (2014-01-15 23:17:15 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.13 A few small fixes in drivers, nothing too remarkable here but all good to have - mainly these are fixes for things that were introduced in the last merge window but only just got useful testing. ---------------------------------------------------------------- Lars-Peter Clausen (1): ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constant Mark Brown (1): Merge remote-tracking branches 'asoc/fix/adau1701', 'asoc/fix/fsl' and 'asoc/fix/tlv320aic32x4' into asoc-linus Markus Pargmann (2): ASoC: codec: tlv320aic32x4: Fix regmap range config ASoC: tlv320aic32x4: Fix regmap range_min Nicolin Chen (1): ASoC: imx-spdif: Use snd-soc-dummy CODEC driver to link card sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 4 +- sound/soc/fsl/Kconfig | 1 - sound/soc/fsl/imx-spdif.c | 84 ++++++++++------------------------------ 4 files changed, 24 insertions(+), 67 deletions(-)

2 4

[alsa-devel] [GIT PULL] ASoC updates for v3.13
by Mark Brown 16 Jan '14

16 Jan '14

The following changes since commit 7e22e91102c6b9df7c4ae2168910e19d2bb14cd6: Linux 3.13-rc8 (2014-01-12 17:04:18 +0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc8-2 for you to fetch changes up to c6affc0dba45bb2074ff56b21386d22eeaf682f8: Merge remote-tracking branches 'asoc/fix/adau1701' and 'asoc/fix/tlv320aic32x4' into asoc-linus (2014-01-16 14:02:52 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.13 A few small fixes in drivers, nothing too remarkable here but all good to have - mainly these are fixes for things that were introduced in the last merge window but only just got useful testing. ---------------------------------------------------------------- Lars-Peter Clausen (1): ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constant Mark Brown (1): Merge remote-tracking branches 'asoc/fix/adau1701' and 'asoc/fix/tlv320aic32x4' into asoc-linus Markus Pargmann (2): ASoC: codec: tlv320aic32x4: Fix regmap range config ASoC: tlv320aic32x4: Fix regmap range_min sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-)

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[alsa-devel] [GIT PULL] ASoC updates for v3.14
by Mark Brown 16 Jan '14

16 Jan '14

The following changes since commit cdef2e5f358db5fd27eec4c72eb295c232287813: sound: oss: remove last sleep_on users (2014-01-14 16:12:07 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-2 for you to fetch changes up to 701caa51a2ce74182d39380ca11defeb163d98c1: Merge remote-tracking branches 'asoc/topic/adsp', 'asoc/topic/atmel', 'asoc/topic/bcm2835', 'asoc/topic/docs', 'asoc/topic/fsl', 'asoc/topic/generic', 'asoc/topic/kirkwood', 'asoc/topic/mc13783', 'asoc/topic/mxs', 'asoc/topic/nuc900', 'asoc/topic/sai', 'asoc/topic/sh', 'asoc/topic/ssm2602', 'asoc/topic/tlv320aic3x', 'asoc/topic/twl4030', 'asoc/topic/ux500', 'asoc/topic/width' and 'asoc/topic/x86' into for-tiwai (2014-01-16 12:44:01 +0000) ---------------------------------------------------------------- ASoC: More updates for v3.14 A few more updates for v3.14 since the last set, highlights include: - Lots of DMA updates from Lars-Peter - Improvements to the constraints handling code from Lars-Peter - A very helpful conversion of the TWL4030 driver to regmap from Peter - A new driver for the Freescale ESAI controller from Nicolin Chen - Conversion of some of the drivers to use params_width() ---------------------------------------------------------------- Alexander Shiyan (5): ASoC: fsl_ssi: Fix printing return code on clk error ASoC: mc13783: Use module_platform_driver_probe() ASoC: mc13783: Use core error messages if registration fails ASoC: mc13783: Drop fixed ADC & DAC ports usage ASoC: mc13783: trivial: Cleanup module Andreas Pretzsch (1): ASoC: ssm2602: add 16kHz sampling rate support Arun Shamanna Lakshmi (1): ASoC: dapm: Change prototype of soc_widget_read Charles Keepax (5): ASoC: wm_adsp: Factor out ADSP2 boot proceedure ASoC: wm_adsp: Start DSP booting earlier in the DAPM process ASoC: wm_adsp: Mark wm_adsp2_boot_work as static mfd: wm5110: Add registers for headphone short circuit control ASoC: wm5110: Add controls for headphone short circuit protection Fabio Estevam (1): ASoC: fsl: fsl_ssi: Use '%ld' to print 'long int' Jean-Francois Moine (1): ASoC: pcm: Fix lack of platform bespoke_trigger() call Lars-Peter Clausen (24): ASoC: atmel: Don't set unused struct snd_pcm_hardware fields ASoC: dpcm: Add helper function for initializing runtime pcm ASoC: dpcm: Allow PCMs to omit the set of supported formats ASoC: fsl: Don't set unused struct snd_pcm_hardware fields ASoC: mxs: Don't set unused struct snd_pcm_hardware fields ASoC: mxs: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constant ASoC: intel: Don't set unused struct snd_pcm_hardware fields ASoC: kirkwood: Don't set unused struct snd_pcm_hardware fields ASoC: nuc900: Don't set unused struct snd_pcm_hardware fields ASoC: sh: Don't set unused struct snd_pcm_hardware fields ASoC: ux500: Don't set unused struct snd_pcm_hardware fields ASoC: Allow PCMs to restrict the supported formats ASoC: pcm: Properly initialize hw->rate_max ASoC: fsl: Don't mix SNDRV_PCM_RATE_CONTINUOUS with specific rates ASoC: s6000: Don't mix SNDRV_PCM_RATE_CONTINUOUS with specific rates ALSA: Add helper function for intersecting two rate masks ASoC: pcm: Use snd_pcm_rate_mask_intersect() helper dma: Indicate residue granularity in dma_slave_caps dma: pl330: Set residue_granularity ASoC: generic-dmaengine-pcm: Check NO_RESIDUE flag at runtime ASoC: generic-dmaengine-pcm: Check DMA residue granularity ASoC: axi-{spdif,i2s}: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag ASoC: samsung: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag Lee Jones (10): ARM: ux500: Don't use enums for MSP IDs - for easy DT conversion ASoC: ux500: Provide better checking for Device Tree and/or Platform Data ASoC: Ux500: Match platform by device node when booting Device Tree ASoC: ux500_pcm: Stop pretending that we support varying address widths ASoC: ux500_pcm: Expect different saved DMA data when obtaining from DAI store ASoC: ux500_pcm: Extend Device Tree support to deal with DMA data ASoC: ux500: Store DMA data in the DAI differently in the pdata and DT case ASoC: ux500_pcm: Take out pointless dev_dbg() call ASoC: ux500_pcm: Differentiate between pdata and DT initialisation ASoC: ux500: Dramatically reduce the size of the DAI driver data struct Liam Girdwood (4): ASoC: docs: Update the Overview document ASoC: dpcm: Explicitly set BE DAI link supported stream directions ASoC: utils: Add internal call to determine if DAI is dummy. ASoC: sapm: Automatically connect DAI link widgets in DAPM graph. Mark Brown (30): Merge tag 'ib-asoc-3.14.2' of git://git.linaro.org/people/ljones/mfd into asoc-tlw4030 ASoC: ad1836: Use params_width() rather than explicit memory format ASoC: ad193x: Use params_width() rather than memory format ASoC: adau1373: Use params_width() rather than memory format ASoC: adau1701: Use params_width() rather than memory format ASoC: adav80x: Use params_width() rather than memory format Merge tag 'v3.13-rc3' into asoc-arizona ASoC: alc5623: Use params_width() rather than memory format ASoC: alc5632: Use params_width() rather than memory format ASoC: cs42l51: Use params_width() rather than memory format ASoC: da7210: Use params_width() rather than memory format ASoC: da7213: Use params_width() rather than memory format ASoC: da732x: Use params_width() rather than memory format ASoC: da9055: Use params_width() rather than memory format ASoC: isabelle: Use params_width() rather than memory format ASoC: max98088: Use params_width() rather than memory format ASoC: max98090: Use params_width() rather than memory format ASoC: max98095: Use params_width() rather than memory format ASoC: max9850: Use params_width() rather than memory format Merge tag 'v3.13-rc3' into asoc-pcm Merge branch 'topic/axi' of git://git.kernel.org/.../broonie/sound into asoc-dma Merge branch 'topic/samsung' of git://git.kernel.org/.../broonie/sound into asoc-dma ASoC: core: Return -ENOTSUPP from set_sysclk() if no operation provided Merge remote-tracking branches 'asoc/fix/adau1701' and 'asoc/fix/tlv320aic32x4' into for-tiwai Merge remote-tracking branch 'asoc/topic/core' into for-tiwai Merge remote-tracking branch 'asoc/topic/dapm' into for-tiwai Merge remote-tracking branch 'asoc/topic/dma' into for-tiwai Merge remote-tracking branch 'asoc/topic/pcm' into for-tiwai Merge remote-tracking branch 'asoc/topic/arizona' into for-tiwai Merge remote-tracking branches 'asoc/topic/adsp', 'asoc/topic/atmel', 'asoc/topic/bcm2835', 'asoc/topic/docs', 'asoc/topic/fsl', 'asoc/topic/generic', 'asoc/topic/kirkwood', 'asoc/topic/mc13783', 'asoc/topic/mxs', 'asoc/topic/nuc900', 'asoc/topic/sai', 'asoc/topic/sh', 'asoc/topic/ssm2602', 'asoc/topic/tlv320aic3x', 'asoc/topic/twl4030', 'asoc/topic/ux500', 'asoc/topic/width' and 'asoc/topic/x86' into for-tiwai Markus Pargmann (13): ASoC: fsl-ssi: Fix probe error handling ASoC: fsl-ssi: Move sysfs stats to debugfs ASoC: fsl-ssi: Add imx51-ssi and of_device_id matching ASoC: fsl-ssi: Fix interrupt stats for imx ASoC: fsl-ssi: Add offline_config flag ASoC: fsl-ssi: Add configuration helper functions ASoC: fsl-ssi: Move RX/TX configuration to seperate functions ASoC: fsl-ssi: Drop ac97 specific trigger function ASoC: fsl-ssi doc: Add list of supported compatibles ASoC: fsl-ssi: Fix stats compile warning ASoC: codec: tlv320aic32x4: Fix regmap range config ASoC: tlv320aic3x: Add tlv320aic32x4 as compatible ASoC: tlv320aic32x4: Fix regmap range_min Nicolin Chen (2): ASoC: fsl_ssi: Set default slot number for common cases ASoC: fsl_esai: Add ESAI CPU DAI driver Peter Ujfalusi (11): mfd: twl-core: Simplify IO wrapper functions by moving common code out mfd: twl-core: API to set the regcache bypass for a given regmap in twl mfd: twl-core: Enable regcache for audio registers ASoC: twl4030: Separate write condition checking from I/O function ASoC: twl4030: Remove check defaults functionality ASoC: twl4030: Remove reset registers functionality ASoC: twl4030: Introduce local ctl register cache ASoC: twl4030: Remove local reg cache ASoC: twl4030: Parameter alignment fixes (for code consistency) ASoC: twl4030: Move the ctl cache update local to twl4030_write() function ASoC: twl4030: Pass the twl4030_priv directly to twl4030_can_write_to_chip() Takashi Iwai (1): ASoC: bcm: Remove obsoleted Kconfig dependency Wei Yongjun (2): ASoC: ux500: Fix sparse non static symbol warning ASoC: fsl-ssi: Add missing clk_disable_unprepare() on error in fsl_ssi_probe() Xiubo Li (8): ASoC: simple-card: Fix the sysclk selection. ASoC: simple-card: fix the DAPM routes map parsing ASoC: simple-card: keep the property's name the same pattern ASoC: fsl-sai: Clean up the code ASoC: simple-card: fix a bug where cinfo will be NULL before using it ASoC: simple-card: fix the cinfo error check ASoC: simple-card: use snd_soc_card_set/get_drvdata ASoC: simple-card: fix one bug to writing to the platform data .../devicetree/bindings/sound/fsl,esai.txt | 50 ++ .../devicetree/bindings/sound/fsl,ssi.txt | 7 +- .../devicetree/bindings/sound/simple-card.txt | 2 +- .../devicetree/bindings/sound/tlv320aic3x.txt | 1 + Documentation/sound/alsa/soc/overview.txt | 27 +- arch/arm/mach-ux500/board-mop500-audio.c | 8 +- drivers/dma/pl330.c | 1 + drivers/mfd/twl-core.c | 190 ++++- drivers/mfd/wm5110-tables.c | 6 + include/linux/dmaengine.h | 28 + include/linux/i2c/twl.h | 5 +- include/linux/mfd/arizona/registers.h | 27 + include/linux/platform_data/asoc-ux500-msp.h | 9 +- include/sound/pcm.h | 2 + include/sound/soc-dai.h | 2 + include/sound/soc-dapm.h | 1 + include/sound/soc.h | 4 + sound/core/pcm_misc.c | 39 + sound/soc/adi/axi-i2s.c | 3 +- sound/soc/adi/axi-spdif.c | 3 +- sound/soc/atmel/atmel-pcm-dma.c | 1 - sound/soc/atmel/atmel-pcm-pdc.c | 1 - sound/soc/bcm/Kconfig | 1 - sound/soc/codecs/ad1836.c | 10 +- sound/soc/codecs/ad193x.c | 10 +- sound/soc/codecs/adau1373.c | 10 +- sound/soc/codecs/adau1701.c | 36 +- sound/soc/codecs/adav80x.c | 30 +- sound/soc/codecs/alc5623.c | 10 +- sound/soc/codecs/alc5632.c | 8 +- sound/soc/codecs/arizona.h | 17 +- sound/soc/codecs/cs42l51.c | 14 +- sound/soc/codecs/da7210.c | 10 +- sound/soc/codecs/da7213.c | 10 +- sound/soc/codecs/da732x.c | 10 +- sound/soc/codecs/da9055.c | 10 +- sound/soc/codecs/isabelle.c | 6 +- sound/soc/codecs/max98088.c | 6 +- sound/soc/codecs/max98090.c | 4 +- sound/soc/codecs/max98095.c | 6 +- sound/soc/codecs/max9850.c | 8 +- sound/soc/codecs/mc13783.c | 34 +- sound/soc/codecs/ssm2602.c | 14 +- sound/soc/codecs/tlv320aic32x4.c | 4 +- sound/soc/codecs/twl4030.c | 376 ++++------ sound/soc/codecs/wm5110.c | 7 + sound/soc/codecs/wm_adsp.c | 199 +++-- sound/soc/codecs/wm_adsp.h | 12 +- sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_dma.c | 7 - sound/soc/fsl/fsl_esai.c | 815 +++++++++++++++++++++ sound/soc/fsl/fsl_esai.h | 354 +++++++++ sound/soc/fsl/fsl_sai.c | 7 +- sound/soc/fsl/fsl_ssi.c | 586 ++++++++++----- sound/soc/fsl/imx-pcm-dma.c | 3 - sound/soc/fsl/imx-pcm-fiq.c | 3 - sound/soc/fsl/mpc5200_dma.c | 4 - sound/soc/fsl/mpc5200_psc_i2s.c | 3 +- sound/soc/generic/simple-card.c | 70 +- sound/soc/intel/sst_platform.c | 10 - sound/soc/intel/sst_platform.h | 4 - sound/soc/kirkwood/kirkwood-dma.c | 16 - sound/soc/mxs/mxs-pcm.c | 6 - sound/soc/nuc900/nuc900-pcm.c | 3 - sound/soc/s6000/s6000-i2s.c | 3 +- sound/soc/samsung/dmaengine.c | 1 - sound/soc/sh/dma-sh7760.c | 17 - sound/soc/sh/fsi.c | 6 - sound/soc/sh/rcar/core.c | 6 - sound/soc/soc-core.c | 5 +- sound/soc/soc-dapm.c | 75 +- sound/soc/soc-generic-dmaengine-pcm.c | 66 +- sound/soc/soc-pcm.c | 58 +- sound/soc/soc-utils.c | 7 + sound/soc/ux500/mop500.c | 2 + sound/soc/ux500/ux500_msp_dai.c | 146 ++-- sound/soc/ux500/ux500_msp_i2s.c | 56 +- sound/soc/ux500/ux500_msp_i2s.h | 2 +- sound/soc/ux500/ux500_pcm.c | 65 +- 80 files changed, 2656 insertions(+), 1034 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,esai.txt create mode 100644 sound/soc/fsl/fsl_esai.c create mode 100644 sound/soc/fsl/fsl_esai.h

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[alsa-devel] different sample rate in alsa.conf file, ALSA DMIX mode
by Pavel Botev 16 Jan '14

16 Jan '14

Hi ALSA support team, Can we set more sample rate for DMIX mode in the alsa.conf file than the default one (defaults.pcm.dmix.rate 48000)? Can we use one alsa.conf file, with many sample rates (8KHz, 11.025KHz, 16KHz, 22.05KHz, 32KHz, 44.1KHz, 48KHz, 64KHz, 88.2KHz and 96KHz), not with just one default sample rate? The problem is that when we use one sample rate in alsa.conf and other in the ALSA application, we have resampling executed which is very CPU intensive. Thus we search a way to use alsa.conf file with many sample rates, thus avoiding resampling. If not possible, can we change the algorithm of resampling, thus have low CPU usage when resampling is activated? Thanks for your time. Best regards, Pavel

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[alsa-devel] [PATCH] ALSA: hda - add headset mic detect quirks for some Dell machines
by Hui Wang 16 Jan '14

16 Jan '14

When we plug a 3-ring headset on some Dell machines, the headset mic can't be detected, after apply this patch, the headset mic can work well on all those machines. On the machine with the Subsytem ID 0x10280610, if we use ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, the headset mic can be detected and work well, but the sound can't be outputed via headphone anymore, use ALC269_FIXUP_DELL3_MIC_NO_PRESENCE can fix this problem. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson <david.henningsson(a)canonical.com> Tested-by: David Chen <david.chen(a)canonical.com> Tested-by: Cyrus Lien <cyrus.lien(a)canonical.com> Tested-by: Shawn Wang <shawn.wang(a)canonical.com> Tested-by: Chih-Hsyuan Ho <chih.ho(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a1c16457..f9b22fb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4253,7 +4253,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x060f, "Dell", ALC269_FIXUP_DELL3_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL3_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), @@ -4264,6 +4265,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0640, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0651, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0652, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0653, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0658, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0662, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), @@ -5065,6 +5071,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_AUTO_MUTE), + SND_PCI_QUIRK(0x1028, 0x064e, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), -- 1.8.1.2

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[alsa-devel] [3.13-rc8] AM335x BeagleBone Buffer XRUN
by Jack Mitchell 16 Jan '14

16 Jan '14

I am currently trying to debug a buffer overrun when capturing with the BeagleBone Black and Audio Cape attached on 3.13-rc8, this is pretty much the same setup as the am335x-evm. The following is the traceback captured using the XRUN_BUFFER_DEBUG kernel config option. I think an interrupt is taking too long to return. Example Traceback 1 (Writing to uSD card) > [ 802.578728] ALSA sound/core/pcm_lib.c:177 XRUN: pcmC0D0c:0 > CPU: 0 PID: 0 Comm: swapper/0 Not tainted 3.13.0-rc8-00005-ga6da83f-dirty #12 > [ 802.593139] [<c00160e8>] (unwind_backtrace+0x0/0x12c) from [<c0012b08>] (show_stack+0x10/0x14) > [ 802.602174] [<c0012b08>] (show_stack+0x10/0x14) from [<c058c80c>] (dump_stack+0x7c/0x94) > [ 802.610659] [<c058c80c>] (dump_stack+0x7c/0x94) from [<c04862e8>] (xrun+0x80/0xb0) > [ 802.618596] [<c04862e8>] (xrun+0x80/0xb0) from [<c0486e1c>] (snd_pcm_update_state+0xf8/0x100) > [ 802.627531] [<c0486e1c>] (snd_pcm_update_state+0xf8/0x100) from [<c0487048>] (snd_pcm_update_hw_ptr0+0x224/0x82c) > [ 802.638281] [<c0487048>] (snd_pcm_update_hw_ptr0+0x224/0x82c) from [<c0487700>] (snd_pcm_period_elapsed+0xb0/0xf4) > [ 802.649131] [<c0487700>] (snd_pcm_period_elapsed+0xb0/0xf4) from [<c0021458>] (dma_irq_handler+0x130/0x1f4) > [ 802.659345] [<c0021458>] (dma_irq_handler+0x130/0x1f4) from [<c008e504>] (handle_irq_event_percpu+0x5c/0x21c) > [ 802.669735] [<c008e504>] (handle_irq_event_percpu+0x5c/0x21c) from [<c008e700>] (handle_irq_event+0x3c/0x5c) > [ 802.680035] [<c008e700>] (handle_irq_event+0x3c/0x5c) from [<c0091038>] (handle_level_irq+0xac/0x118) > [ 802.689697] [<c0091038>] (handle_level_irq+0xac/0x118) from [<c008dc8c>] (generic_handle_irq+0x28/0x3c) > [ 802.699542] [<c008dc8c>] (generic_handle_irq+0x28/0x3c) from [<c000f968>] (handle_IRQ+0x4c/0xb4) > [ 802.708750] [<c000f968>] (handle_IRQ+0x4c/0xb4) from [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) > [ 802.718237] [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) from [<c0592a64>] (__irq_svc+0x44/0x5c) > [ 802.727618] Exception stack(0xc088df58 to 0xc088dfa0) > [ 802.732912] df40: 00000001 00000001 > [ 802.741479] df60: 00000000 c0897cb0 c088c000 c088c000 00000000 c0933bd4 c08949a0 c0894880 > [ 802.750047] df80: c0894930 c059f4a8 c088c020 c088dfa0 c0082ef8 c000fd00 20070013 ffffffff > [ 802.758620] [<c0592a64>] (__irq_svc+0x44/0x5c) from [<c000fd00>] (arch_cpu_idle+0x20/0x3c) > [ 802.767283] [<c000fd00>] (arch_cpu_idle+0x20/0x3c) from [<c008db84>] (cpu_startup_entry+0x120/0x1f0) > [ 802.776858] [<c008db84>] (cpu_startup_entry+0x120/0x1f0) from [<c0826a9c>] (start_kernel+0x304/0x364) Example Traceback 2 (Writing to tmpfs) > [ 1310.025256] ALSA sound/core/pcm_lib.c:177 XRUN: pcmC0D0c:0 > [ 1310.031046] CPU: 0 PID: 998 Comm: avconv Not tainted 3.13.0-rc8-00005-ga6da83f-dirty #12 > [ 1310.039580] [<c00160e8>] (unwind_backtrace+0x0/0x12c) from [<c0012b08>] (show_stack+0x10/0x14) > [ 1310.048615] [<c0012b08>] (show_stack+0x10/0x14) from [<c058c80c>] (dump_stack+0x7c/0x94) > [ 1310.057102] [<c058c80c>] (dump_stack+0x7c/0x94) from [<c04862e8>] (xrun+0x80/0xb0) > 1310.065039] [<c04862e8>] (xrun+0x80/0xb0) from [<c0486e1c>] (snd_pcm_update_state+0xf8/0x100) > [ 1310.073974] [<c0486e1c>] (snd_pcm_update_state+0xf8/0x100) from [<c0487048>] (snd_pcm_update_hw_ptr0+0x224/0x82c) > [ 1310.084724] [<c0487048>] (snd_pcm_update_hw_ptr0+0x224/0x82c) from [<c0487700>] (snd_pcm_period_elapsed+0xb0/0xf4) > [ 1310.095574] [<c0487700>] (snd_pcm_period_elapsed+0xb0/0xf4) from [<c0021458>] (dma_irq_handler+0x130/0x1f4) > [ 1310.105788] [<c0021458>] (dma_irq_handler+0x130/0x1f4) from [<c008e504>] (handle_irq_event_percpu+0x5c/0x21c) > [ 1310.116178] [<c008e504>] (handle_irq_event_percpu+0x5c/0x21c) from [<c008e700>] (handle_irq_event+0x3c/0x5c) > [ 1310.126477] [<c008e700>] (handle_irq_event+0x3c/0x5c) from [<c0091038>] (handle_level_irq+0xac/0x118) > [ 1310.136138] [<c0091038>] (handle_level_irq+0xac/0x118) from [<c008dc8c>] (generic_handle_irq+0x28/0x3c) > [ 1310.145982] [<c008dc8c>] (generic_handle_irq+0x28/0x3c) from [<c000f968>] (handle_IRQ+0x4c/0xb4) > [ 1310.155190] [<c000f968>] (handle_IRQ+0x4c/0xb4) from [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) > [ 1310.164677] [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) from [<c0592a64>] (__irq_svc+0x44/0x5c) > [ 1310.174056] Exception stack(0xde0f7ee0 to 0xde0f7f28) > [ 1310.179354] 7ee0: 00000001 00000000 00000000 de3f8a80 00000000 00000002 c0047968 00000054 > [ 1310.187920] 7f00: 00000000 b5440c64 c0933670 beb92fb8 de0f6028 de0f7f28 c0082eb0 c0047348 > [ 1310.196480] 7f20: 60000113 ffffffff > [ 1310.200153] [<c0592a64>] (__irq_svc+0x44/0x5c) from [<c0047348>] (__do_softirq+0xa8/0x2f4) > [ 1310.208817] [<c0047348>] (__do_softirq+0xa8/0x2f4) from [<c0047968>] (irq_exit+0xac/0x104) > [ 1310.217479] [<c0047968>] (irq_exit+0xac/0x104) from [<c000f970>] (handle_IRQ+0x54/0xb4) > [ 1310.225868] [<c000f970>] (handle_IRQ+0x54/0xb4) from [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) > [ 1310.235347] [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) from [<c0592c3c>] (__irq_usr+0x3c/0x60) > [ 1310.244724] Exception stack(0xde0f7fb0 to 0xde0f7ff8) > [ 1310.250019] 7fa0: beb94588 beb92ab8 beb9301c beb93038 > [ 1310.258586] 7fc0: 00088ca0 00000140 00000160 b5440f18 00000080 b5440c64 beb92ab8 beb92fb8 > [ 1310.267150] 7fe0: 4b000001 beb9267c b5f4d6a0 b5f4c8c8 80000010 ffffffff I have done some tests and capturing to a uSD card is the easiest way to trigger XRUNS, where they occur often. Capturing to the eMMC is a little bit better. Capturing directly to tmpfs is nearly perfect, apart from very rare XRUN occurrences (10 minutes average maybe). This makes sense as EDMA is used to perform writes to the uSD and eMMC devices, so if they're being hogged with writes, then there is no time to transfer out the sound buffer and we get an overrun. So, does anybody have ideas where to start debugging this further? I have tried to increase the buffer size from 32K up to the supposed max of 128K but I can't find where to do it... if anyone could shed any light I would most grateful, do you think increasing the buffer would solve this, as it would give more time to trigger the irq and get the data out before overrun? I have tried multiple userspace programs to do the capture and the best so far is arecord initialized like this: nice -n -20 arecord -D hw:0,0 -f cd -v jack-test.wav > root@beaglebone:~# nice -n -20 arecord -D hw:0,0 -f dat -v jack-test.wav > Recording WAVE 'jack-test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo > DEBUG: buffer_time = 682667 > DEBUG: max_buffer_size = 32768 > Hardware PCM card 0 'AM335x-EVM' device 0 subdevice 0 > Its setup is: > stream : CAPTURE > access : RW_INTERLEAVED > format : S16_LE > subformat : STD > channels : 2 > rate : 48000 > exact rate : 48000 (48000/1) > msbits : 16 > buffer_size : 32768 > period_size : 2048 > period_time : 42666 > tstamp_mode : NONE > period_step : 1 > avail_min : 2048 > period_event : 0 > start_threshold : 1 > stop_threshold : 32768 > silence_threshold: 0 > silence_size : 0 > boundary : 1073741824 > appl_ptr : 0 > hw_ptr : 0 > overrun!!! (at least 221.598 ms long) > Status: > state : XRUN > trigger_time: 1389534120.970142840 > tstamp : 1389534121.191651631 > delay : 0 > avail : 32768 > avail_max : 32768 If you need any more information or debugging output please let me know. Regards, -- Jack Mitchell (jack(a)embed.me.uk) Embedded Systems Engineer http://www.embed.me.uk --

1 1

Re: [alsa-devel] [pulseaudio-discuss] Need help with audio gain on an Intel HDA equipt ASUS M2N-SLI Deluxe
by Raymond Yau 16 Jan '14

16 Jan '14

2014/1/16 Axel Braun <axel.braun(a)gmx.de> > Am Donnerstag, 16. Januar 2014, 09:03:25 schrieb Raymond Yau: > > Post the output of > > > > pactl list > > > > pulseaudio verbose log > > I jump in here as well, same problem on a ThinkPad T520, sound volume is > way > too low - watching movie via loudspeakers is fairly impossible. > > Any hint you could give me? > post the output of alsa-info.sh , there is a bug in alsa-driver which treat dock headphone and headphone to support surround 4.0 Profile: input:analog-stereo: Analog Stereo Input (sinks: 0, sources: 1, priority: 60, available: yes) output:analog-stereo: Analog Stereo Output (sinks: 1, sources: 0, priority: 6000, available: yes) output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (sinks: 1, sources: 1, priority: 6060, available: yes) output:analog-surround-40: Analog Surround 4.0 Output (sinks: 1, sources: 0, priority: 700, available: yes) output:analog-surround-40+input:analog-stereo: Analog Surround 4.0 Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 760, available: yes)

1 0

[alsa-devel] [PATCH] ASoC: simple-card: Add snd_card's name parsing from DT node support
by Xiubo Li 16 Jan '14

16 Jan '14

If the DT is used and the CPU DAI device has only one DAI, the card name will be like : ALSA device list: 0: 40031000.sai-sgtl5000 And this name maybe a little ugly to some customers, so here the card name parsing from DT node is supported. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card | 0 sound/soc/generic/simple-card. | 0 sound/soc/generic/simple-card.c | 7 ++++++- 3 files changed, 6 insertions(+), 1 deletion(-) create mode 100644 sound/soc/generic/simple-card create mode 100644 sound/soc/generic/simple-card. diff --git a/sound/soc/generic/simple-card b/sound/soc/generic/simple-card new file mode 100644 index 0000000..e69de29 diff --git a/sound/soc/generic/simple-card. b/sound/soc/generic/simple-card. new file mode 100644 index 0000000..e69de29 diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 0e2fbdd..b40b57c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -143,6 +143,10 @@ static int asoc_simple_card_parse_of(struct device_node *node, return ret; } + /* parsing the card name from DT */ + snd_soc_of_parse_card_name(&info->snd_card, + "simple-audio-card,name"); + /* CPU sub-node */ ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,cpu"); @@ -257,7 +261,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) /* * init snd_soc_card */ - cinfo->snd_card.name = cinfo->card; + if (!cinfo->snd_card.name) + cinfo->snd_card.name = cinfo->card; cinfo->snd_card.owner = THIS_MODULE; cinfo->snd_card.dai_link = &cinfo->snd_link; cinfo->snd_card.num_links = 1; -- 1.8.4

3 2

[alsa-devel] [PATCHv2] ASoC: simple-card: Add snd_card's name parsing from DT node support
by Xiubo Li 16 Jan '14

16 Jan '14

If the DT is used and the CPU DAI device has only one DAI, the card name will be like : ALSA device list: 0: 40031000.sai-sgtl5000 And this name maybe a little ugly to some customers, so here the card name parsing from DT node is supported. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Resend this patch. sound/soc/generic/simple-card.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d437c80..711d6a1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -152,6 +152,10 @@ static int asoc_simple_card_parse_of(struct device_node *node, return ret; } + /* parsing the card name from DT */ + snd_soc_of_parse_card_name(&info->snd_card, + "simple-audio-card,name"); + /* CPU sub-node */ ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,cpu"); @@ -266,7 +270,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) /* * init snd_soc_card */ - cinfo->snd_card.name = cinfo->card; + if (!cinfo->snd_card.name) + cinfo->snd_card.name = cinfo->card; cinfo->snd_card.owner = THIS_MODULE; cinfo->snd_card.dai_link = &cinfo->snd_link; cinfo->snd_card.num_links = 1; -- 1.8.4

1 0

[alsa-devel] Problems with simple-card
by Jean-Francois Moine 16 Jan '14

16 Jan '14

Xiubo, I did a mistake in the v1 of my 'ASoC: simple-card: simplify code': I did not initialize the pointer to the asoc_simple_card_dai_init() function when DT. Then, I fixed that, and the simple card does not work for me. First, without any 'format' in the DT, I get a fmt for each CPU / CODEC DAI: SND_SOC_DAIFMT_CBS_CFS is always set. Well, some code is executed for nothing, but this is not critical. The main problem is about sysclk: when there is no clock in the DT, you get the clock of the CPU or CODEC DAI. In my system, two clocks are declared in the controller CPU DAI, but there is no .set_sysclk pointer. So, snd_soc_dai_set_sysclk() returns -EINVAL and the card is not created. As I don't know why you need these fmt and sysclk, may you have a look at these problems? -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

4 4

[alsa-devel] [PATCH] ASoC: simple-card: simplify the daifmt code
by Xiubo Li 16 Jan '14

16 Jan '14

In the asoc_simple_card_parse_of() will parse the device node's CPU/CODEC DAI commone fmts, and then in asoc_simple_card_sub_parse_of() will parse the CPU/CODEC DAI's sub-node fmts, so we can combine the info->daifmt and info->set.fmt in asoc_simple_card_sub_parse_of() not while just before set_fmt(). Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index b6014fe..0e2fbdd 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -16,15 +16,12 @@ #include <sound/simple_card.h> static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, - struct asoc_simple_dai *set, - unsigned int daifmt) + struct asoc_simple_dai *set) { int ret; - daifmt |= set->fmt; - - if (daifmt) { - ret = snd_soc_dai_set_fmt(dai, daifmt); + if (set->fmt) { + ret = snd_soc_dai_set_fmt(dai, set->fmt); if (ret && ret != -ENOTSUPP) { dev_err(dai->dev, "simple-card: set_fmt error\n"); return ret; @@ -48,14 +45,13 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; - unsigned int daifmt = info->daifmt; int ret; - ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt); + ret = __asoc_simple_card_dai_init(codec, &info->codec_dai); if (ret < 0) return ret; - ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt); + ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai); if (ret < 0) return ret; @@ -64,6 +60,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) static int asoc_simple_card_sub_parse_of(struct device_node *np, + unsigned int daifmt, struct asoc_simple_dai *dai, struct device_node **node) { @@ -89,6 +86,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, * and specific "format" if it has */ dai->fmt = snd_soc_of_parse_daifmt(np, NULL); + dai->fmt |= daifmt; /* * dai->sysclk come from @@ -149,7 +147,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) - ret = asoc_simple_card_sub_parse_of(np, + ret = asoc_simple_card_sub_parse_of(np, info->daifmt, &info->cpu_dai, of_cpu); if (ret < 0) @@ -159,7 +157,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) - ret = asoc_simple_card_sub_parse_of(np, + ret = asoc_simple_card_sub_parse_of(np, info->daifmt, &info->codec_dai, of_codec); if (ret < 0) -- 1.8.4

1 0

[alsa-devel] [PATCH] ASoC: simple-card: fix __asoc_simple_card_dai_init
by Xiubo Li 16 Jan '14

16 Jan '14

If the CPU/CODEC DAI set_sysclk() is not support, the -ENOTSUPP will returnd. Here do the check like set_fmt(). Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2a1b1b5..b6014fe 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -19,22 +19,27 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set, unsigned int daifmt) { - int ret = 0; + int ret; daifmt |= set->fmt; - if (daifmt) + if (daifmt) { ret = snd_soc_dai_set_fmt(dai, daifmt); - - if (ret == -ENOTSUPP) { - dev_dbg(dai->dev, "ASoC: set_fmt is not supported\n"); - ret = 0; + if (ret && ret != -ENOTSUPP) { + dev_err(dai->dev, "simple-card: set_fmt error\n"); + return ret; + } } - if (!ret && set->sysclk) + if (set->sysclk) { ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); + if (ret && ret != -ENOTSUPP) { + dev_err(dai->dev, "simple-card: set_sysclk error\n"); + return ret; + } + } - return ret; + return 0; } static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) -- 1.8.4

1 0

[alsa-devel] [PATCH 0/3] ASoC: tlv320aic32x4 bugfixes
by Markus Pargmann 16 Jan '14

16 Jan '14

Hi, Another round of tlv320aic32x4 bugfixes. I will send some more non-bugfix patches for this codec next week including the DT compatibles we talked about. Regards, Markus Markus Pargmann (3): ASoC: tlv320aic32x4: Fix regmap range_min ASoC: tlv320aic32x4: Fix mono playback ASoC: tlv320aic32x4: Fix MICPGA input configuration sound/soc/codecs/tlv320aic32x4.c | 30 ++++++++++++++++++------------ sound/soc/codecs/tlv320aic32x4.h | 3 +++ 2 files changed, 21 insertions(+), 12 deletions(-) -- 1.8.5.2

2 4

[alsa-devel] [PATCH] ASoC: core: Return -ENOTSUPP from set_sysclk() if no operation provided
by Mark Brown 15 Jan '14

15 Jan '14

From: Mark Brown <broonie(a)linaro.org> Make it easier for generic code to work with set_sysclk() by distinguishing between the operation not being supported and an error as is done for other operations like set_dai_fmt() Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a128f0cd752..fe1df50805a3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3485,7 +3485,7 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, return dai->codec->driver->set_sysclk(dai->codec, clk_id, 0, freq, dir); else - return -EINVAL; + return -ENOTSUPP; } EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); @@ -3506,7 +3506,7 @@ int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, return codec->driver->set_sysclk(codec, clk_id, source, freq, dir); else - return -EINVAL; + return -ENOTSUPP; } EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk); -- 1.8.5.2

1 0

Re: [alsa-devel] Surround sound not working on ALC887 based motherboard
by Raymond Yau 15 Jan '14

15 Jan '14

2014/1/15 Olivier Crête <olivier.crete(a)ocrete.ca> > Hi, > > > > > I have a Asus P5G41T-M LX Plus motherboard which is ICH7 + > > > Realtek > > > ALC887-VD, alsa only lets me access the first 2 channels > > > (stereo), but > > > not the 6 other ones. It works fine in Windows. > > > > > > Trying to run this after killing pulseaudio fails: > > > > > > $ aplay -D surround40:0 -f S16_LE -r 44100 -c 4 /dev/zero -v > > > Playing raw data '/dev/zero' : Signed 16 bit Little Endian, > > > Rate 44100 Hz, Channels 4 > > > aplay: set_params:1239: Channels count non available > > > > > > > > > Output from alsa-info: > > > > http://www.alsa-project.org/db/?f=fe378d28f0dbb5a427b005d20866a762fd37d31a > > > > > > > > post output of pactl list or pulseaudio verbose log > > > > http://www.ocrete.ca/surround/pactl-list.txt > > http://www.ocrete.ca/surround/pulseaudio-vv.txt > > > > > you have to change channel mode from 2ch to 6ch to retask the input > > > jacks as output if your motherboard only have three audio jacks at > > > rear panel > > > > How would one do that ? Pulseaudio only offers Stereo profiles. > amixer -c0 set "Channel Mode" "6ch" > > > do you mean pulseaudio cannot probe the surroun40 or surround51 > > profile ? > > Yes, but I can't use it with aplay even without pulseaudio running. > > > Simple mixer control 'Channel Mode',0 > > Capabilities: enum > > Items: '2ch' '4ch' '6ch' > > Item0: '2ch' > > > For some reason didn't get your last email, but I saw I saw it on the > archive.. Thanks for your solution, it worked! Definitely something > PulseAudio should expose better. do it really work since rear mic become active port even when it had been retasked as output ? Ports: analog-input-front-mic: Front Microphone (priority: 8500, not available) analog-input-rear-mic: Rear Microphone (priority: 8200, available) analog-input-linein: Line In (priority: 8100, available) Active Port: analog-input-rear-mic

2 1

[alsa-devel] [PATCH] Change the prototype of soc_widget_read
by Arun Shamanna Lakshmi 15 Jan '14

15 Jan '14

soc_widget_read API returns the register data and it is possible that a register can contain 0xffffffff at any point of time. In such cases snd_soc_widget_write is not called after the read operation inside snd_soc_update_bits_locked API. Thus, change the prototype of soc_widget_read to return only the error code and pass the register data through the pointer argument. Arun Shamanna Lakshmi (1): ASoC: dapm: Change prototype of soc_widget_read sound/soc/soc-dapm.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) -- 1.7.9.5

2 2

[alsa-devel] [PATCH 0/2] ASoC: codec tlv320aic32x4 fixes
by Markus Pargmann 15 Jan '14

15 Jan '14

Hi, Here are two fixes for the tlv320aic32x4 codec. The first patch fixes a regmap bug introduced in 3.13. The second is not as important, it adds the codec to the compatible list in the binding documentation. Regards, Markus Markus Pargmann (2): ASoC: codec: tlv320aic32x4: Fix regmap range config ASoC: codec doc, tlv320aic3x: Add tlv320aic32x4 as compatible Documentation/devicetree/bindings/sound/tlv320aic3x.txt | 1 + sound/soc/codecs/tlv320aic32x4.c | 2 +- 2 files changed, 2 insertions(+), 1 deletion(-) -- 1.8.5.2

2 9

[alsa-devel] [PATCH v3 0/6] dma: Indicate residue granularity in dma_slave_caps
by Lars-Peter Clausen 14 Jan '14

14 Jan '14

New in v3: * Expanded the kernel doc documentation for the dma_residue_granularity struct a bit New in v2: * Dropped byte level granularity as discussed during v1 review * Added support for auto-detecting the NO_RESIDUE flag on the compat path in the generic ASoC dmaengine PCM driver * Updated a few ASoC drivers to take advantage of the new API Updated original cover text from v1 below: This patch series introduces a new field to the dma_slave_caps struct which exposes the granularity with which the reported residue is updated, updates the pl330 DMA driver to set this field and lets the generic dmaengine PCM audio driver check that field. The reason for introducing the field is that for audio we need to know the granularity with which the reported residue is updated. The uncertainty of the ALSA period pointer depends on the granularity of the reported residue. E.g. if residue is only updated once every period this means that the position in the buffer from/to which the hardware is currently reading/writing could be anywhere between the reported PCM pointer and the reported PCM pointer plus one period size (plus some extra margin for interrupt latency, etc). Certain algorithms (e.g. pulseaudio's timer based scheduling) only work correctly if the uncertainty of the PCM pointer is low enough. Exposing this information from the DMA driver makes it possible for applications to switch to alternative modes of operation which also work with a high granularity. The patch series introduces four levels of granularity, which I think will cover most DMA cores. I'd like some feedback though if anybody can think of a DMA controller for which none of the levels would be correct. The four levels are: * DESCRIPTOR: The DMA channel is only able to tell whether a descriptor has been completed or not, which means residue reporting is not supported by this channel. The residue field of the dma_tx_state field will always be 0. * SEGMENT: The DMA channel updates the residue field after each successfully completed segment of the transfer (For cyclic transfers this is after each period). This is typically implemented by having the hardware generate an interrupt after each transferred segment and then the driver updates the outstanding residue by the size of the segment. Another possibility is if the hardware supports SG and the segment descriptor has a field which gets set by the hardware after the segment has been completed. The driver then counts the number of segments which do not have the flag set to compute the residue. * BURST: The DMA channel updates the residue field after each transferred burst. This is typically only supported if the hardware has a progress register of some sort (E.g. a register with the current read/write address or a register with the amount of bursts/beats/bytes that have been transferred or still need to be transferred). - Lars Lars-Peter Clausen (6): dma: Indicate residue granularity in dma_slave_caps dma: pl330: Set residue_granularity ASoC: generic-dmaengine-pcm: Check NO_RESIDUE flag at runtime ASoC: generic-dmaengine-pcm: Check DMA residue granularity ASoC: axi-{spdif,i2s}: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag ASoC: samsung: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag drivers/dma/pl330.c | 1 + include/linux/dmaengine.h | 28 +++++++++++++++ sound/soc/adi/axi-i2s.c | 3 +- sound/soc/adi/axi-spdif.c | 3 +- sound/soc/samsung/dmaengine.c | 1 - sound/soc/soc-generic-dmaengine-pcm.c | 66 ++++++++++++++++++++++------------- 6 files changed, 73 insertions(+), 29 deletions(-) -- 1.8.0

2 7

[alsa-devel] [PATCH resend] ASoC: simple-card: fix one bug to writing to the platform data
by Xiubo Li 14 Jan '14

14 Jan '14

It's a bug that writing to the platform data directly, for it should be constant. So just copy it before writing. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Resend this patch. sound/soc/generic/simple-card.c | 40 +++++++++++++++++++++------------------- 1 file changed, 21 insertions(+), 19 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 5528dd6..53395f5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -9,9 +9,10 @@ * published by the Free Software Foundation. */ #include <linux/clk.h> +#include <linux/module.h> #include <linux/of.h> #include <linux/platform_device.h> -#include <linux/module.h> +#include <linux/string.h> #include <sound/simple_card.h> static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, @@ -190,36 +191,37 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct device_node *of_cpu, *of_codec, *of_platform; struct device *dev = &pdev->dev; + int ret; cinfo = NULL; of_cpu = NULL; of_codec = NULL; of_platform = NULL; + + cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); + if (!cinfo) + return -ENOMEM; + if (np && of_device_is_available(np)) { - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); - if (cinfo) { - int ret; - cinfo->snd_card.dev = &pdev->dev; - ret = asoc_simple_card_parse_of(np, cinfo, dev, - &of_cpu, - &of_codec, - &of_platform); - if (ret < 0) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "parse error %d\n", ret); - return ret; - } - } else { - return -ENOMEM; + cinfo->snd_card.dev = dev; + + ret = asoc_simple_card_parse_of(np, cinfo, dev, + &of_cpu, + &of_codec, + &of_platform); + if (ret < 0) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "parse error %d\n", ret); + return ret; } } else { - cinfo = pdev->dev.platform_data; - if (!cinfo) { + if (!dev->platform_data) { dev_err(dev, "no info for asoc-simple-card\n"); return -EINVAL; } - cinfo->snd_card.dev = &pdev->dev; + memcpy(cinfo, dev->platform_data, sizeof(*cinfo)); + cinfo->snd_card.dev = dev; } if (!cinfo->name || -- 1.8.4

2 1

[alsa-devel] [PATCH v2 1/5] ASoC: pcm: Properly initialize hw->rate_max
by Lars-Peter Clausen 14 Jan '14

14 Jan '14

If none of the components (CODEC or CPU DAI) sets a maximum sample rate we'll end up with the rate_max field of the runtime hardware set to 0. (Note that it is still possible for the components to constrain the supported sample rates using other methods, e.g. setting a list constraint) If rate_max is 0 this means that the sound card doesn't support any rates at all, which is not the desired result. So initialize rate_max to UINT_MAX. For symmetry reasons also set rate_min to 0. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- No changes in v2 --- sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 313adba..b6e0c32 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -252,6 +252,9 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) hw->rates |= codec_stream->rates; + hw->rate_min = 0; + hw->rate_max = UINT_MAX; + snd_pcm_limit_hw_rates(runtime); hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); -- 1.8.0

3 7

[alsa-devel] [PATCH] ASoC: simple-card: simplify code
by Jean-Francois Moine 14 Jan '14

14 Jan '14

This patch - removes the fields of the platform data which are of no use to the non-DT platform callers, - uses a new private structure to handle all the sound card information, - simplifies the code and make easier a possible multi-DAI links extension. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- Xiubo, I also removed 'of_device_is_available' which seems really useless: the module is not probed when the DT status is not "okay". --- include/sound/simple_card.h | 4 -- sound/soc/generic/simple-card.c | 178 ++++++++++----------- 2 files changed, 88 insertions(+), 94 deletions(-) diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 6c74527..e1ac996 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -29,10 +29,6 @@ struct asoc_simple_card_info { unsigned int daifmt; struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; - - /* used in simple-card.c */ - struct snd_soc_dai_link snd_link; - struct snd_soc_card snd_card; }; #endif /* __SIMPLE_CARD_H */ diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 5528dd6..580900e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -14,6 +14,13 @@ #include <linux/module.h> #include <sound/simple_card.h> +struct simple_card_data { + struct snd_soc_card snd_card; + unsigned int daifmt; + struct asoc_simple_dai cpu_dai; + struct asoc_simple_dai codec_dai; +}; + static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set, unsigned int daifmt) @@ -38,18 +45,18 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *info = + struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; - unsigned int daifmt = info->daifmt; + unsigned int daifmt = priv->daifmt; int ret; - ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt); + ret = __asoc_simple_card_dai_init(codec, &priv->codec_dai, daifmt); if (ret < 0) return ret; - ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt); + ret = __asoc_simple_card_dai_init(cpu, &priv->cpu_dai, daifmt); if (ret < 0) return ret; @@ -59,7 +66,8 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) static int asoc_simple_card_sub_parse_of(struct device_node *np, struct asoc_simple_dai *dai, - struct device_node **node) + struct device_node **node, + const char **name) { struct clk *clk; int ret; @@ -73,7 +81,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, return -ENODEV; /* get dai->name */ - ret = snd_soc_of_get_dai_name(np, &dai->name); + ret = snd_soc_of_get_dai_name(np, name); if (ret < 0) goto parse_error; @@ -117,23 +125,21 @@ parse_error: } static int asoc_simple_card_parse_of(struct device_node *node, - struct asoc_simple_card_info *info, - struct device *dev, - struct device_node **of_cpu, - struct device_node **of_codec, - struct device_node **of_platform) + struct simple_card_data *priv, + struct device *dev) { + struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; struct device_node *np; char *name; int ret; /* get CPU/CODEC common format via simple-audio-card,format */ - info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & + priv->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); /* DAPM routes */ if (of_property_read_bool(node, "simple-audio-card,routing")) { - ret = snd_soc_of_parse_audio_routing(&info->snd_card, + ret = snd_soc_of_parse_audio_routing(&priv->snd_card, "simple-audio-card,routing"); if (ret) return ret; @@ -144,8 +150,10 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) ret = asoc_simple_card_sub_parse_of(np, - &info->cpu_dai, - of_cpu); + &priv->cpu_dai, + (struct device_node **) + &dai_link->cpu_of_node, + &dai_link->cpu_dai_name); if (ret < 0) return ret; @@ -154,109 +162,99 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) ret = asoc_simple_card_sub_parse_of(np, - &info->codec_dai, - of_codec); + &priv->codec_dai, + (struct device_node **) + &dai_link->codec_of_node, + &dai_link->codec_dai_name); if (ret < 0) return ret; /* card name is created from CPU/CODEC dai name */ name = devm_kzalloc(dev, - strlen(info->cpu_dai.name) + - strlen(info->codec_dai.name) + 2, + strlen(dai_link->cpu_dai_name) + + strlen(dai_link->codec_dai_name) + 2, GFP_KERNEL); - sprintf(name, "%s-%s", info->cpu_dai.name, info->codec_dai.name); - info->name = info->card = name; + sprintf(name, "%s-%s", dai_link->cpu_dai_name, dai_link->codec_dai_name); + priv->snd_card.name = name; + dai_link->name = dai_link->stream_name = name; /* simple-card assumes platform == cpu */ - *of_platform = *of_cpu; + dai_link->platform_of_node = dai_link->cpu_of_node; - dev_dbg(dev, "card-name : %s\n", info->card); - dev_dbg(dev, "platform : %04x\n", info->daifmt); + dev_dbg(dev, "card-name : %s\n", name); + dev_dbg(dev, "platform : %04x\n", priv->daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", - info->cpu_dai.name, - info->cpu_dai.fmt, - info->cpu_dai.sysclk); + dai_link->cpu_dai_name, + priv->cpu_dai.fmt, + priv->cpu_dai.sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", - info->codec_dai.name, - info->codec_dai.fmt, - info->codec_dai.sysclk); + dai_link->codec_dai_name, + priv->codec_dai.fmt, + priv->codec_dai.sysclk); return 0; } static int asoc_simple_card_probe(struct platform_device *pdev) { + struct simple_card_data *priv; + struct snd_soc_dai_link *dai_link; struct asoc_simple_card_info *cinfo; struct device_node *np = pdev->dev.of_node; - struct device_node *of_cpu, *of_codec, *of_platform; struct device *dev = &pdev->dev; - cinfo = NULL; - of_cpu = NULL; - of_codec = NULL; - of_platform = NULL; - if (np && of_device_is_available(np)) { - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); - if (cinfo) { - int ret; - cinfo->snd_card.dev = &pdev->dev; - ret = asoc_simple_card_parse_of(np, cinfo, dev, - &of_cpu, - &of_codec, - &of_platform); - if (ret < 0) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "parse error %d\n", ret); - return ret; - } - } else { - return -ENOMEM; - } - } else { - cinfo = pdev->dev.platform_data; - if (!cinfo) { - dev_err(dev, "no info for asoc-simple-card\n"); + priv = devm_kzalloc(dev, + sizeof(*priv) + sizeof(*dai_link), + GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->snd_card.owner = THIS_MODULE; + priv->snd_card.dev = &pdev->dev; + dai_link = (struct snd_soc_dai_link *) (priv + 1); + priv->snd_card.dai_link = dai_link; + priv->snd_card.num_links = 1; + + cinfo = pdev->dev.platform_data; + if (cinfo) { + if (!cinfo->name || + !cinfo->card || + !cinfo->codec_dai.name || + !cinfo->codec || + !cinfo->platform || + !cinfo->cpu_dai.name) { + dev_err(dev, "insufficient asoc_simple_card_info settings\n"); return -EINVAL; } - cinfo->snd_card.dev = &pdev->dev; - } - - if (!cinfo->name || - !cinfo->card || - !cinfo->codec_dai.name || - !(cinfo->codec || of_codec) || - !(cinfo->platform || of_platform) || - !(cinfo->cpu_dai.name || of_cpu)) { - dev_err(dev, "insufficient asoc_simple_card_info settings\n"); + dai_link->name = cinfo->name; + dai_link->stream_name = cinfo->name; + dai_link->cpu_dai_name = cinfo->cpu_dai.name; + dai_link->platform_name = cinfo->platform; + dai_link->codec_name = cinfo->codec; + dai_link->codec_dai_name = cinfo->codec_dai.name; + memcpy(&priv->cpu_dai, &cinfo->cpu_dai, + sizeof(priv->cpu_dai)); + memcpy(&priv->codec_dai, &cinfo->codec_dai, + sizeof(priv->codec_dai)); + dai_link->init = asoc_simple_card_dai_init; + } else if (np) { + int ret; + + ret = asoc_simple_card_parse_of(np, priv, dev); + if (ret < 0) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "parse error %d\n", ret); + return ret; + } + } else { + dev_err(dev, "no info for asoc-simple-card\n"); return -EINVAL; } - /* - * init snd_soc_dai_link - */ - cinfo->snd_link.name = cinfo->name; - cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; - cinfo->snd_link.platform_name = cinfo->platform; - cinfo->snd_link.codec_name = cinfo->codec; - cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; - cinfo->snd_link.cpu_of_node = of_cpu; - cinfo->snd_link.codec_of_node = of_codec; - cinfo->snd_link.platform_of_node = of_platform; - cinfo->snd_link.init = asoc_simple_card_dai_init; - - /* - * init snd_soc_card - */ - cinfo->snd_card.name = cinfo->card; - cinfo->snd_card.owner = THIS_MODULE; - cinfo->snd_card.dai_link = &cinfo->snd_link; - cinfo->snd_card.num_links = 1; - - snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); + snd_soc_card_set_drvdata(&priv->snd_card, priv); - return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); + return devm_snd_soc_register_card(&pdev->dev, &priv->snd_card); } static const struct of_device_id asoc_simple_of_match[] = { -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 3

[alsa-devel] [PATCH, RFC 00/30] sleep_on removal
by Arnd Bergmann 14 Jan '14

14 Jan '14

The functions sleep_on, sleep_on_timeout, interruptible_sleep_on and interruptible_sleep_on_timeout have been deprecated for as long as I can remember, and a number of people have contributed patches in the past to remove them from various drivers. This has recently popped up again and I decided to spend some of my time on tracking down the last users and kill it for good. The work is somewhat related to the BKL removal I did a couple of years ago, since most of these drivers were using the BKL in a way that actually made the sleep_on function safe to call. As the BKL was converted into mutexes, the semantics changed slightly and typically opened up a race between checking a condition and going to sleep. I don't have any of the hardware that I'm sending the patches for, so it would be great if someone could test them before they get applied. Otherwise please review very carefully. I'm definitely happy for any patches to go into maintainer trees right away. Obviously the final patch cannot go in until everything else gets merged first and I suspect there will be a series of patches for maintainerless drivers that will go along with it. Arnd Arnd Bergmann (30): ataflop: fix sleep_on races scsi: atari_scsi: fix sleep_on race DAC960: remove sleep_on usage swim3: fix interruptible_sleep_on race [media] omap_vout: avoid sleep_on race [media] usbvision: remove bogus sleep_on_timeout [media] radio-cadet: avoid interruptible_sleep_on race [media] arv: fix sleep_on race staging: serqt_usb2: don't use sleep_on staging: gdm72xx: fix interruptible_sleep_on race staging: panel: fix interruptible_sleep_on race parport: fix interruptible_sleep_on race cris: sync_serial: remove interruptible_sleep_on tty/amiserial: avoid interruptible_sleep_on usbserial: stop using interruptible_sleep_on tty: synclink: avoid sleep_on race atm: nicstar: remove interruptible_sleep_on_timeout atm: firestream: fix interruptible_sleep_on race isdn: pcbit: fix interruptible_sleep_on race isdn: hisax/elsa: fix sleep_on race in elsa FSM isdn: divert, hysdn: fix interruptible_sleep_on race isdn: fix multiple sleep_on races oss: msnd_pinnacle: avoid interruptible_sleep_on_timeout oss: midibuf: fix sleep_on races oss: vwsnd: avoid interruptible_sleep_on oss: dmasound: kill SLEEP() macro to avoid race oss: remove last sleep_on users sgi-xp: open-code interruptible_sleep_on_timeout char: nwbutton: open-code interruptible_sleep_on sched: remove sleep_on() and friends Documentation/DocBook/kernel-hacking.tmpl | 10 ------ arch/cris/arch-v10/drivers/sync_serial.c | 4 ++- arch/cris/arch-v32/drivers/sync_serial.c | 4 ++- drivers/atm/firestream.c | 4 +-- drivers/atm/nicstar.c | 13 ++++---- drivers/block/DAC960.c | 34 +++++++++---------- drivers/block/ataflop.c | 16 ++++----- drivers/block/swim3.c | 18 ++++++---- drivers/char/nwbutton.c | 5 ++- drivers/char/pcmcia/synclink_cs.c | 4 +-- drivers/isdn/divert/divert_procfs.c | 7 ++-- drivers/isdn/hisax/elsa.c | 9 +++-- drivers/isdn/hisax/elsa_ser.c | 3 +- drivers/isdn/hysdn/hysdn_proclog.c | 7 ++-- drivers/isdn/i4l/isdn_common.c | 13 +++++--- drivers/isdn/pcbit/drv.c | 6 ++-- drivers/media/platform/arv.c | 4 +-- drivers/media/platform/omap/omap_vout_vrfb.c | 3 +- drivers/media/radio/radio-cadet.c | 12 +++++-- drivers/media/usb/usbvision/usbvision.h | 4 +-- drivers/misc/sgi-xp/xpc_channel.c | 5 ++- drivers/parport/share.c | 3 +- drivers/scsi/atari_scsi.c | 12 +++++-- drivers/staging/gdm72xx/gdm_usb.c | 5 +-- drivers/staging/panel/panel.c | 4 +-- drivers/staging/serqt_usb2/serqt_usb2.c | 17 ++++------ drivers/tty/amiserial.c | 26 ++++++++++----- drivers/tty/synclink.c | 4 +-- drivers/tty/synclink_gt.c | 4 +-- drivers/tty/synclinkmp.c | 4 +-- drivers/usb/serial/ch341.c | 29 +++++++++++----- drivers/usb/serial/cypress_m8.c | 49 ++++++++++++++++++---------- drivers/usb/serial/f81232.c | 29 +++++++++++----- drivers/usb/serial/pl2303.c | 29 +++++++++++----- include/linux/wait.h | 11 ------- kernel/sched/core.c | 46 -------------------------- sound/oss/dmabuf.c | 14 ++++---- sound/oss/dmasound/dmasound.h | 1 - sound/oss/dmasound/dmasound_core.c | 28 +++++++++++----- sound/oss/midibuf.c | 18 +++++----- sound/oss/msnd_pinnacle.c | 31 ++++++++++-------- sound/oss/sequencer.c | 16 ++++----- sound/oss/sleep.h | 18 ++++++++++ sound/oss/swarm_cs4297a.c | 14 ++++---- sound/oss/vwsnd.c | 14 +++++--- 45 files changed, 334 insertions(+), 277 deletions(-) create mode 100644 sound/oss/sleep.h -- 1.8.3.2 Cc: akpm(a)osdl.org Cc: chas(a)cmf.nrl.navy.mil Cc: davem(a)davemloft.net Cc: geert(a)linux-m68k.org Cc: gregkh(a)linuxfoundation.org Cc: hverkuil(a)xs4all.nl Cc: mingo(a)kernel.org Cc: JBottomley(a)parallels.com Cc: perex(a)perex.cz Cc: axboe(a)kernel.dk Cc: jslaby(a)suse.cz Cc: isdn(a)linux-pingi.de Cc: m.chehab(a)samsung.com Cc: peterz(a)infradead.org Cc: robinmholt(a)gmail.com Cc: tiwai(a)suse.de Cc: alsa-devel(a)alsa-project.org Cc: devel(a)driverdev.osuosl.org Cc: linux-atm-general(a)lists.sourceforge.net Cc: linux-cris-kernel(a)axis.com Cc: linux-media(a)vger.kernel.org Cc: linux-scsi(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: netdev(a)vger.kernel.org

2 6

[alsa-devel] [ISSUE] crackles and clicks
by Eric F 14 Jan '14

14 Jan '14

Please never do the following things on public mailing lists: 1. Take the mailing list address out of Cc. ALWAYS(!) use "reply to all" 2. Top-post. Please quote your reposonse properly, as everyone else does 3. Send HTML-Email. Please use plain-text only. As for your problem, please refer to the the reply I posted to you earlier mail. I'm really having trouble understanding your problem. Please follow the three basic rules above when replying :) I use Gmail, haven't seen a reply to all option... I'm not used to this, pplease excuse me... HERE IS WHAT I SAID : Hello, There is no distortion ! only crackles and clicks. When using it with Jack, no xruns at all. But for testing purpose, I have disabled Jack, mpd, I use only aplay. Dunno if there is a way for buffer stuff with it. Gonna buy a powered USB Hub. Thanks !

1 0

[alsa-devel] [RFC][PATCH] ASoC: simple-card: Add asoc_simple_card_data for the simple card driver data
by Xiubo Li 14 Jan '14

14 Jan '14

Whether the dt is used or not, almost all the simple card information for the DAI link and sound card are initialized in the simple card driver. And for the platform caller, the snd_link and snd_card are of no use, so move them from struct asoc_simple_card_info to struct asoc_simple_card_data. And now only one DAI link is supported for simple card. Suggested-by: Jean-Francois Moine <moinejf(a)free.fr> Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- include/sound/simple_card.h | 4 -- sound/soc/generic/simple-card.c | 115 +++++++++++++++++++++------------------- 2 files changed, 60 insertions(+), 59 deletions(-) diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 6c74527..e1ac996 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -29,10 +29,6 @@ struct asoc_simple_card_info { unsigned int daifmt; struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; - - /* used in simple-card.c */ - struct snd_soc_dai_link snd_link; - struct snd_soc_card snd_card; }; #endif /* __SIMPLE_CARD_H */ diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2a1b1b5..eab8f92 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -15,6 +15,12 @@ #include <linux/string.h> #include <sound/simple_card.h> +struct asoc_simple_card_data { + struct asoc_simple_card_info info; + struct snd_soc_dai_link snd_link; + struct snd_soc_card snd_card; +}; + static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set, unsigned int daifmt) @@ -39,18 +45,18 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *info = + struct asoc_simple_card_data *sdata = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; - unsigned int daifmt = info->daifmt; + unsigned int daifmt = sdata->info.daifmt; int ret; - ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt); + ret = __asoc_simple_card_dai_init(codec, &sdata->info.codec_dai, daifmt); if (ret < 0) return ret; - ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt); + ret = __asoc_simple_card_dai_init(cpu, &sdata->info.cpu_dai, daifmt); if (ret < 0) return ret; @@ -118,23 +124,23 @@ parse_error: } static int asoc_simple_card_parse_of(struct device_node *node, - struct asoc_simple_card_info *info, - struct device *dev, + struct asoc_simple_card_data *sdata, struct device_node **of_cpu, struct device_node **of_codec, struct device_node **of_platform) { + struct device *dev = sdata->snd_card.dev; struct device_node *np; char *name; int ret; /* get CPU/CODEC common format via simple-audio-card,format */ - info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & - (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); + sdata->info.daifmt = snd_soc_of_parse_daifmt(node, + "simple-audio-card,") & SND_SOC_DAIFMT_FORMAT_MASK; /* DAPM routes */ if (of_property_read_bool(node, "simple-audio-card,routing")) { - ret = snd_soc_of_parse_audio_routing(&info->snd_card, + ret = snd_soc_of_parse_audio_routing(&sdata->snd_card, "simple-audio-card,routing"); if (ret) return ret; @@ -145,7 +151,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) ret = asoc_simple_card_sub_parse_of(np, - &info->cpu_dai, + &sdata->info.cpu_dai, of_cpu); if (ret < 0) return ret; @@ -155,60 +161,60 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) ret = asoc_simple_card_sub_parse_of(np, - &info->codec_dai, + &sdata->info.codec_dai, of_codec); if (ret < 0) return ret; - if (!info->cpu_dai.name || !info->codec_dai.name) + if (!sdata->info.cpu_dai.name || !sdata->info.codec_dai.name) return -EINVAL; /* card name is created from CPU/CODEC dai name */ name = devm_kzalloc(dev, - strlen(info->cpu_dai.name) + - strlen(info->codec_dai.name) + 2, + strlen(sdata->info.cpu_dai.name) + + strlen(sdata->info.codec_dai.name) + 2, GFP_KERNEL); - sprintf(name, "%s-%s", info->cpu_dai.name, info->codec_dai.name); - info->name = info->card = name; + sprintf(name, "%s-%s", sdata->info.cpu_dai.name, + sdata->info.codec_dai.name); + sdata->info.name = sdata->info.card = name; /* simple-card assumes platform == cpu */ *of_platform = *of_cpu; - dev_dbg(dev, "card-name : %s\n", info->card); - dev_dbg(dev, "platform : %04x\n", info->daifmt); + dev_dbg(dev, "card-name : %s\n", sdata->info.card); + dev_dbg(dev, "platform : %04x\n", sdata->info.daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", - info->cpu_dai.name, - info->cpu_dai.fmt, - info->cpu_dai.sysclk); + sdata->info.cpu_dai.name, + sdata->info.cpu_dai.fmt, + sdata->info.cpu_dai.sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", - info->codec_dai.name, - info->codec_dai.fmt, - info->codec_dai.sysclk); + sdata->info.codec_dai.name, + sdata->info.codec_dai.fmt, + sdata->info.codec_dai.sysclk); return 0; } static int asoc_simple_card_probe(struct platform_device *pdev) { - struct asoc_simple_card_info *cinfo; + struct asoc_simple_card_data *sdata; struct device_node *np = pdev->dev.of_node; struct device_node *of_cpu, *of_codec, *of_platform; struct device *dev = &pdev->dev; int ret; - cinfo = NULL; of_cpu = NULL; of_codec = NULL; of_platform = NULL; - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); - if (!cinfo) + sdata = devm_kzalloc(dev, sizeof(*sdata), GFP_KERNEL); + if (!sdata) return -ENOMEM; - if (np && of_device_is_available(np)) { - cinfo->snd_card.dev = dev; + sdata->snd_card.dev = dev; - ret = asoc_simple_card_parse_of(np, cinfo, dev, + if (np && of_device_is_available(np)) { + ret = asoc_simple_card_parse_of(np, sdata, &of_cpu, &of_codec, &of_platform); @@ -223,16 +229,15 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return -EINVAL; } - memcpy(cinfo, dev->platform_data, sizeof(*cinfo)); - cinfo->snd_card.dev = dev; + memcpy(&sdata->info, dev->platform_data, sizeof(sdata->info)); } - if (!cinfo->name || - !cinfo->card || - !cinfo->codec_dai.name || - !(cinfo->codec || of_codec) || - !(cinfo->platform || of_platform) || - !(cinfo->cpu_dai.name || of_cpu)) { + if (!sdata->info.name || + !sdata->info.card || + !sdata->info.codec_dai.name || + !(sdata->info.codec || of_codec) || + !(sdata->info.platform || of_platform) || + !(sdata->info.cpu_dai.name || of_cpu)) { dev_err(dev, "insufficient asoc_simple_card_info settings\n"); return -EINVAL; } @@ -240,28 +245,28 @@ static int asoc_simple_card_probe(struct platform_device *pdev) /* * init snd_soc_dai_link */ - cinfo->snd_link.name = cinfo->name; - cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; - cinfo->snd_link.platform_name = cinfo->platform; - cinfo->snd_link.codec_name = cinfo->codec; - cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; - cinfo->snd_link.cpu_of_node = of_cpu; - cinfo->snd_link.codec_of_node = of_codec; - cinfo->snd_link.platform_of_node = of_platform; - cinfo->snd_link.init = asoc_simple_card_dai_init; + sdata->snd_link.name = sdata->info.name; + sdata->snd_link.stream_name = sdata->info.name; + sdata->snd_link.cpu_dai_name = sdata->info.cpu_dai.name; + sdata->snd_link.platform_name = sdata->info.platform; + sdata->snd_link.codec_name = sdata->info.codec; + sdata->snd_link.codec_dai_name = sdata->info.codec_dai.name; + sdata->snd_link.cpu_of_node = of_cpu; + sdata->snd_link.codec_of_node = of_codec; + sdata->snd_link.platform_of_node = of_platform; + sdata->snd_link.init = asoc_simple_card_dai_init; /* * init snd_soc_card */ - cinfo->snd_card.name = cinfo->card; - cinfo->snd_card.owner = THIS_MODULE; - cinfo->snd_card.dai_link = &cinfo->snd_link; - cinfo->snd_card.num_links = 1; + sdata->snd_card.name = sdata->info.card; + sdata->snd_card.owner = THIS_MODULE; + sdata->snd_card.dai_link = &sdata->snd_link; + sdata->snd_card.num_links = 1; - snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); + snd_soc_card_set_drvdata(&sdata->snd_card, sdata); - return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); + return devm_snd_soc_register_card(dev, &sdata->snd_card); } static const struct of_device_id asoc_simple_of_match[] = { -- 1.8.4

4 4

[alsa-devel] [PATCH] ALSA: hda - Fix endless vmaster hook call in thinkpad_helper.c
by Takashi Iwai 14 Jan '14

14 Jan '14

The new vmaster hook, update_tpacpi_mute_led(), calls the original vmaster hook, but I forgot to save the original hook function but keep calling the updated one, which of course results in a stupid endless loop. Fixed now. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/thinkpad_helper.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 8492b8a3ce47..5799fbc24c28 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -8,6 +8,7 @@ #include <linux/thinkpad_acpi.h> static int (*led_set_func)(int, bool); +static void (*old_vmaster_hook)(void *, int); static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, void **rv) @@ -30,11 +31,8 @@ static bool is_thinkpad(struct hda_codec *codec) static void update_tpacpi_mute_led(void *private_data, int enabled) { - struct hda_codec *codec = private_data; - struct hda_gen_spec *spec = codec->spec; - - if (spec->vmaster_mute.hook) - spec->vmaster_mute.hook(private_data, enabled); + if (old_vmaster_hook) + old_vmaster_hook(private_data, enabled); if (led_set_func) led_set_func(TPACPI_LED_MUTE, !enabled); @@ -70,6 +68,7 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, removefunc = true; if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { + old_vmaster_hook = spec->vmaster_mute.hook; spec->vmaster_mute.hook = update_tpacpi_mute_led; removefunc = false; } @@ -86,6 +85,7 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { symbol_put(tpacpi_led_set); led_set_func = NULL; + old_vmaster_hook = NULL; } } -- 1.8.5.2

1 0

[alsa-devel] [PATCH] ALSA: snd-usb: re-order some quirk entries
by Daniel Mack 14 Jan '14

14 Jan '14

No code change, just a cosmetic cleanup to keep entries ordered by the device ID within a block of unique vendor IDs. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/quirks-table.h | 170 +++++++++++++++++++++++------------------------ 1 file changed, 84 insertions(+), 86 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 3454780..f652b10 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -72,22 +72,21 @@ } }, -/* Creative/Toshiba Multimedia Center SB-0500 */ +/* Creative/E-Mu devices */ { - USB_DEVICE(0x041e, 0x3048), + USB_DEVICE(0x041e, 0x3010), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Toshiba", - .product_name = "SB-0500", + .vendor_name = "Creative Labs", + .product_name = "Sound Blaster MP3+", .ifnum = QUIRK_NO_INTERFACE } }, - -/* Creative/E-Mu devices */ +/* Creative/Toshiba Multimedia Center SB-0500 */ { - USB_DEVICE(0x041e, 0x3010), + USB_DEVICE(0x041e, 0x3048), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Creative Labs", - .product_name = "Sound Blaster MP3+", + .vendor_name = "Toshiba", + .product_name = "SB-0500", .ifnum = QUIRK_NO_INTERFACE } }, @@ -2521,17 +2520,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE(0x1235, 0x0018), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Novation", - .product_name = "Twitch", + USB_DEVICE(0x1235, 0x0010), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Saffire 6 USB", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { + .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 4, .iface = 0, @@ -2561,26 +2560,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x1235, 0x4661), + USB_DEVICE(0x1235, 0x0018), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Novation", - .product_name = "ReMOTE25", - .ifnum = 0, - .type = QUIRK_MIDI_NOVATION - } -}, -{ - USB_DEVICE(0x1235, 0x0010), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Focusrite", - .product_name = "Saffire 6 USB", + .product_name = "Twitch", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 4, .iface = 0, @@ -2609,6 +2599,66 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE_VENDOR_SPEC(0x1235, 0x4661), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Novation", + .product_name = "ReMOTE25", + .ifnum = 0, + .type = QUIRK_MIDI_NOVATION + } +}, +{ + /* + * Focusrite Scarlett 18i6 + * + * Avoid mixer creation, which otherwise fails because some of + * the interface descriptor subtypes for interface 0 are + * unknown. That should be fixed or worked-around but this at + * least allows the device to be used successfully with a DAW + * and an external mixer. See comments below about other + * ignored interfaces. + */ + USB_DEVICE(0x1235, 0x8004), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Scarlett 18i6", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = & (const struct snd_usb_audio_quirk[]) { + { + /* InterfaceSubClass 1 (Control Device) */ + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + /* InterfaceSubClass 1 (Control Device) */ + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + /* InterfaceSubClass 1 (Device Firmware Update) */ + .ifnum = 5, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, /* Access Music devices */ { @@ -2711,7 +2761,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7240), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7210), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2719,13 +2769,13 @@ YAMAHA_DEVICE(0x7010, "UB99"), .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", - .product_name = "HVR-850", + .product_name = "HVR-950Q", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_AUDIO_ALIGN_TRANSFER, } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7210), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7217), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2739,7 +2789,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7217), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x721b), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2753,7 +2803,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721b), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x721e), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2767,7 +2817,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721e), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x721f), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2781,7 +2831,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721f), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7240), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2789,7 +2839,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", + .product_name = "HVR-850", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_AUDIO_ALIGN_TRANSFER, } @@ -3094,58 +3144,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* - * Focusrite Scarlett 18i6 - * - * Avoid mixer creation, which otherwise fails because some of - * the interface descriptor subtypes for interface 0 are - * unknown. That should be fixed or worked-around but this at - * least allows the device to be used successfully with a DAW - * and an external mixer. See comments below about other - * ignored interfaces. - */ - USB_DEVICE(0x1235, 0x8004), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Focusrite", - .product_name = "Scarlett 18i6", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = & (const struct snd_usb_audio_quirk[]) { - { - /* InterfaceSubClass 1 (Control Device) */ - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - /* InterfaceSubClass 1 (Control Device) */ - .ifnum = 3, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 4, - .type = QUIRK_MIDI_STANDARD_INTERFACE - }, - { - /* InterfaceSubClass 1 (Device Firmware Update) */ - .ifnum = 5, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = -1 - } - } - } -}, - -{ - /* * Some USB MIDI devices don't have an audio control interface, * so we have to grab MIDI streaming interfaces here. */ -- 1.8.4.2

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Fix Creative VF0420 rate
by Pavel Hofman 14 Jan '14

14 Jan '14

>From db1dcaf0f3c2459b02b29c55639a9a6943cffec9 Mon Sep 17 00:00:00 2001 From: Pavel Hofman <pavel.hofman(a)ivitera.com> Date: Tue, 14 Jan 2014 13:42:57 +0100 Subject: [PATCH] ALSA: usb-audio: Fix Creative VF0420 rate Creative Live! Cam Vista IM (VF0420) reports rate of 16kHz while working at 8kHz. The patch adds its USB ID to the existing quirk. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- sound/usb/format.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 3525231..d244fd3 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -189,8 +189,10 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof chip->usb_id == USB_ID(0x0ccd, 0x00b1)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; - /* Creative VF0470 Live Cam reports 16 kHz instead of 8kHz */ - if (rate == 16000 && chip->usb_id == USB_ID(0x041e, 0x4068)) + /* Creative VF0420/VF0470 Live Cams report 16 kHz instead of 8kHz */ + if (rate == 16000 && + (chip->usb_id == USB_ID(0x041e, 0x4064) || + chip->usb_id == USB_ID(0x041e, 0x4068))) rate = 8000; fp->rate_table[fp->nr_rates] = rate; -- 1.7.1

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Add support for Focusrite Saffire 6 USB
by Eduard Gilmutdinov 14 Jan '14

14 Jan '14

Signed-off-by: Eduard Gilmutdinov <edgilmutdinov(a)gmail.com> --- sound/usb/quirks-table.h | 40 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 9 +++++---- 2 files changed, 45 insertions(+), 4 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f5f0595..3454780 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2569,6 +2569,46 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_NOVATION } }, +{ + USB_DEVICE(0x1235, 0x0010), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Saffire 6 USB", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = 1, + .type = QUIRK_MIDI_RAW_BYTES + }, + { + .ifnum = -1 + } + } + } +}, /* Access Music devices */ { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0df9ede..e9a59e9 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -662,8 +662,9 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) /* * Novation Twitch DJ controller + * Focusrite Novation Saffire 6 USB audio card */ -static int snd_usb_twitch_boot_quirk(struct usb_device *dev) +static int snd_usb_novation_boot_quirk(struct usb_device *dev) { /* preemptively set up the device because otherwise the * raw MIDI endpoints are not active */ @@ -972,9 +973,9 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, /* Digidesign Mbox 2 */ return snd_usb_mbox2_boot_quirk(dev); - case USB_ID(0x1235, 0x0018): - /* Focusrite Novation Twitch */ - return snd_usb_twitch_boot_quirk(dev); + case USB_ID(0x1235, 0x0010): /* Focusrite Novation Saffire 6 USB */ + case USB_ID(0x1235, 0x0018): /* Focusrite Novation Twitch */ + return snd_usb_novation_boot_quirk(dev); case USB_ID(0x133e, 0x0815): /* Access Music VirusTI Desktop */ -- 1.8.5.2

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[alsa-devel] [PATCHv2 0/6] One simple card use case
by Xiubo Li 14 Jan '14

14 Jan '14

This patch series is one use case of simple card driver based on dts. For some audio cards they need to add some off-CODEC widgets. For the audio card, if it needs off-CODEC widgets, some explict code for a simple audio is needed, if not, just ignore of this. This can make the audio card driver simpler and easier to debug by using the simple card driver and some explict code if needed.

3 8

[alsa-devel] [PATCH v2] ALSA: hda - automute via amp instead of pinctl on some AIO models
by Hui Wang 14 Jan '14

14 Jan '14

On some AIO (All In One) models with the codec alc668 (Vendor ID: 0x10ec0668) on it, when we plug a headphone into the jack, the system will switch the output to headphone and set the speaker to automute as well as change the speaker Pin-ctls from 0x40 to 0x00, this will bring loud noise to the headphone. I tried to disable the corresponding EAPD, but it did not help to eliminate the noise. According to Takashi's suggestion, we use amp operation to replace the pinctl modification for the automute, this really eliminate the noise. BugLink: https://bugs.launchpad.net/bugs/1268468 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 22 +++++++++++++++++++--- 1 file changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d35cf3d..a1c16457 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3580,6 +3580,15 @@ static void alc_fixup_headset_mode_alc255(struct hda_codec *codec, alc_fixup_headset_mode(codec, fix, action); } +static void alc_fixup_auto_mute_via_amp(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct alc_spec *spec = codec->spec; + spec->gen.auto_mute_via_amp = 1; + } +} + static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4839,6 +4848,7 @@ enum { ALC662_FIXUP_BASS_CHMAP, ALC662_FIXUP_BASS_1A, ALC662_FIXUP_BASS_1A_CHMAP, + ALC668_FIXUP_AUTO_MUTE, }; static const struct hda_fixup alc662_fixups[] = { @@ -4999,6 +5009,12 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC668_FIXUP_AUTO_MUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_auto_mute_via_amp, + .chained = true, + .chain_id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE + }, [ALC668_FIXUP_DELL_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5044,11 +5060,11 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_AUTO_MUTE), + SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: add snd_soc_of_parse_audio_simple_widgets for DeviceTree
by Xiubo Li 14 Jan '14

14 Jan '14

This patch adds snd_soc_of_parse_audio_simple_widgets() and supports below style of widgets name on DT. "wname-prefix[ individual name]" "wname-prefix" includes: "Mic", "Line", "Hp", "Spk"... For instance: simple-audio-widgets = "Mic Jack", "Line In Jack", "Hp Jack", "Spk Ext", "Line Out Jack"; Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 62 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 64 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 03ce45b..b2b5900 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1172,6 +1172,8 @@ void snd_soc_util_exit(void); int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname); +int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, + const char *propname); int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); unsigned int snd_soc_of_parse_daifmt(struct device_node *np, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a128f0..36cadd3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4417,6 +4417,68 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_card_name); +static struct snd_soc_dapm_widget simple_widgets[] = { + SND_SOC_DAPM_MIC("Mic", NULL), + SND_SOC_DAPM_LINE("Line", NULL), + SND_SOC_DAPM_HP("Hp", NULL), + SND_SOC_DAPM_SPK("Spk", NULL), +}; + +int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, + const char *propname) +{ + struct device_node *np = card->dev->of_node; + struct snd_soc_dapm_widget *widgets; + const char *wname; + int i, j, cnt, ret; + + cnt = of_property_count_strings(np, propname); + if (cnt <= 0) { + dev_err(card->dev, + "ASoC: Property '%s' does not exist or " + "length is zero\n", propname); + return -EINVAL; + } + + widgets = devm_kcalloc(card->dev, cnt, sizeof(*widgets), GFP_KERNEL); + if (!widgets) { + dev_err(card->dev, + "ASoC: Could not allocate DAPM widgets table\n"); + return -ENOMEM; + } + + for (i = 0; i < cnt; i++) { + ret = of_property_read_string_index(np, propname, i, &wname); + if (ret) { + dev_err(card->dev, + "ASoC: Property '%s' index %d could not be read: %d\n", + propname, i, ret); + return -EINVAL; + } + + for (j = 0; j < ARRAY_SIZE(simple_widgets); j++) { + if (!strncmp(wname, simple_widgets[j].name, + strlen(simple_widgets[j].name))) { + widgets[i] = simple_widgets[j]; + break; + } + } + + if (j >= ARRAY_SIZE(simple_widgets)) { + dev_err(card->dev, + "ASoC: DAPM widget : '%s' is not supported\n", + wname); + return -EINVAL; + } + } + + card->dapm_widgets = widgets; + card->num_dapm_widgets = cnt; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets); + int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname) { -- 1.8.4

1 0

[alsa-devel] Surround sound not working on ALC887 based motherboard
by Olivier Crête 14 Jan '14

14 Jan '14

Hello, I have a Asus P5G41T-M LX Plus motherboard which is ICH7 + Realtek ALC887-VD, alsa only lets me access the first 2 channels (stereo), but not the 6 other ones. It works fine in Windows. Trying to run this after killing pulseaudio fails: $ aplay -D surround40:0 -f S16_LE -r 44100 -c 4 /dev/zero -v Playing raw data '/dev/zero' : Signed 16 bit Little Endian, Rate 44100 Hz, Channels 4 aplay: set_params:1239: Channels count non available Output from alsa-info: http://www.alsa-project.org/db/?f=fe378d28f0dbb5a427b005d20866a762fd37d31a -- Olivier Crête olivier.crete(a)ocrete.ca

2 3

[alsa-devel] [PATCH] ALSA: hda - Prevent speaker from entering automute on some AIO models
by Hui Wang 14 Jan '14

14 Jan '14

On some AIO (All In One) models with the codec alc668 (Vendor ID: 0x10ec0668) on it, when we plug a headphone into the jack, the system will switch the output to headphone and set the speaker to automute as well as change the speaker Pin-ctls from 0x40 to 0x00, this will bring loud noise to the headphone. I tried to disable the corresponding EAPD, but it did not help to eliminate the noise. So far, the only way to eliminate the noise is preventing the speaker from entering automute. BugLink: https://bugs.launchpad.net/bugs/1268468 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- This problem has been reported on three Dell AIO models. sound/pci/hda/patch_realtek.c | 24 +++++++++++++++++++++--- 1 file changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6a705a8..27111d0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3589,6 +3589,17 @@ static void alc_fixup_headset_mode_alc255(struct hda_codec *codec, alc_fixup_headset_mode(codec, fix, action); } +static void alc_fixup_spk_automute_alc668(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct alc_spec *spec = codec->spec; + /* On some AIO (All In One) machines, setting the speaker + to audomute will bring the noise to the headphone. */ + spec->gen.suppress_auto_mute = 1; + } +} + static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4935,6 +4946,7 @@ enum { ALC662_FIXUP_BASS_CHMAP, ALC662_FIXUP_BASS_1A, ALC662_FIXUP_BASS_1A_CHMAP, + ALC668_FIXUP_AUTO_MUTE, }; static const struct hda_fixup alc662_fixups[] = { @@ -5095,6 +5107,12 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC668_FIXUP_AUTO_MUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_spk_automute_alc668, + .chained = true, + .chain_id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE + }, [ALC668_FIXUP_DELL_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5140,11 +5158,11 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_AUTO_MUTE), + SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), -- 1.8.1.2

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[alsa-devel] Possible regression of Mackerel/ASoC Simple Card in next-20130107
by Simon Horman 14 Jan '14

14 Jan '14

Hi Morimoto-san, Hi all, I noticed that next-20130107 does not boot on the Mackerel board does not boot using its defconfig. The boot log is as follows. I have not investigated any further as typically when I investigate problems in next they turn out to be known problems. Linux version 3.13.0-rc7-next-20140107-00005-gad2dc8a (horms(a)ayumi.isobedori.kobe.vergenet.net) (gcc version 4.7.2 (Debian 4.7.2-4) ) #771 Wed Jan 8 12:16:31 JST 2014 CPU: ARMv7 Processor [413fc081] revision 1 (ARMv7), cr=10c53c7d CPU: PIPT / VIPT nonaliasing data cache, VIPT aliasing instruction cache Machine model: Mackerel (AP4 EVM 2nd) Memory policy: Data cache writeback CPU: All CPU(s) started in SVC mode. bootconsole [early_ttySC0] enabled Built 1 zonelists in Zone order, mobility grouping off. Total pages: 60960 Kernel command line: console=tty0, console=ttySC0,115200 earlyprintk=sh-sci.0,115200 root=/dev/nfs nfsroot=,tcp,v3 ip=dhcp mem=240m rw PID hash table entries: 1024 (order: 0, 4096 bytes) Dentry cache hash table entries: 32768 (order: 5, 131072 bytes) Inode-cache hash table entries: 16384 (order: 4, 65536 bytes) Memory: 238412K/245760K available (3419K kernel code, 200K rwdata, 1072K rodata, 148K init, 174K bss, 7348K reserved) Virtual kernel memory layout: vector : 0xffff0000 - 0xffff1000 ( 4 kB) fixmap : 0xfff00000 - 0xfffe0000 ( 896 kB) vmalloc : 0xcf800000 - 0xff000000 ( 760 MB) lowmem : 0xc0000000 - 0xcf000000 ( 240 MB) modules : 0xbf000000 - 0xc0000000 ( 16 MB) .text : 0xc0008000 - 0xc046bf6c (4496 kB) .init : 0xc046c000 - 0xc04910d0 ( 149 kB) .data : 0xc0492000 - 0xc04c41a0 ( 201 kB) .bss : 0xc04c41a0 - 0xc04efb08 ( 175 kB) NR_IRQS:16 nr_irqs:16 16 intc: Registered controller 'sh7372-intca' with 108 IRQs intc: Registered controller 'sh7372-intca-irq-lo' with 16 IRQs intc: Registered controller 'sh7372-intca-irq-hi' with 16 IRQs intc: Registered controller 'sh7372-intcs' with 61 IRQs sched_clock: 32 bits at 128 Hz, resolution 7812500ns, wraps every 16777216000000000ns Console: colour dummy device 80x30 console [tty0] enabled sh_tmu.0: used for clock events sh_tmu.0: used for periodic clock events sh_tmu.1: used as clock source Calibrating delay loop... 805.59 BogoMIPS (lpj=3145728) pid_max: default: 32768 minimum: 301 Mount-cache hash table entries: 512 CPU: Testing write buffer coherency: ok Setting up static identity map for 0x4035d778 - 0x4035d7d0 VFP support v0.3: implementor 41 architecture 3 part 30 variant c rev 3 pinctrl core: initialized pinctrl subsystem regulator-dummy: no parameters NET: Registered protocol family 16 DMA: preallocated 256 KiB pool for atomic coherent allocations sh-pfc pfc-sh7372: sh7372_pfc handling gpio 0 -> 190 sh-pfc pfc-sh7372: sh7372_pfc support registered No ATAGs? bio: create slab <bio-0> at 0 fixed-1.8V: at 1800 mV fixed-3.3V: at 3300 mV fixed-dummy: no parameters SCSI subsystem initialized usbcore: registered new interface driver usbfs usbcore: registered new interface driver hub usbcore: registered new device driver usb i2c-sh_mobile i2c-sh_mobile.0: I2C adapter 0 with bus speed 100000 Hz (L/H=f0/ce) i2c-sh_mobile i2c-sh_mobile.1: I2C adapter 1 with bus speed 100000 Hz (L/H=f0/ce) sh_cmt sh_cmt.2: used for clock events sh_cmt sh_cmt.2: used as clock source sh_tmu sh_tmu.0: kept as earlytimer sh_tmu sh_tmu.1: kept as earlytimer Advanced Linux Sound Architecture Driver Initialized. Switched to clocksource sh_tmu.1 NET: Registered protocol family 2 TCP established hash table entries: 2048 (order: 1, 8192 bytes) TCP bind hash table entries: 2048 (order: 1, 8192 bytes) TCP: Hash tables configured (established 2048 bind 2048) TCP: reno registered UDP hash table entries: 256 (order: 0, 4096 bytes) UDP-Lite hash table entries: 256 (order: 0, 4096 bytes) NET: Registered protocol family 1 RPC: Registered named UNIX socket transport module. RPC: Registered udp transport module. RPC: Registered tcp transport module. RPC: Registered tcp NFSv4.1 backchannel transport module. NFS: Registering the id_resolver key type Key type id_resolver registered Key type id_legacy registered nfs4filelayout_init: NFSv4 File Layout Driver Registering... msgmni has been set to 465 io scheduler noop registered (default) sh-mobile-hdmi sh-mobile-hdmi: Detected HDMI controller 0x0:0x0 Console: switching to colour frame buffer device 100x30 sh_mobile_lcdc_fb sh_mobile_lcdc_fb.0: registered sh_mobile_lcdc_fb.0/mainlcd as 800x480 16bpp. sh_mobile_lcdc_fb sh_mobile_lcdc_fb.1: registered sh_mobile_lcdc_fb.1/mainlcd as 1280x720 16bpp. SuperH (H)SCI(F) driver initialized sh-sci.0: ttySC0 at MMIO 0xe6c40000 (irq = 80, base_baud = 0) is a scifa console [ttySC0] enabled console [ttySC0] enabled bootconsole [early_ttySC0] disabled bootconsole [early_ttySC0] disabled sh-sci.1: ttySC1 at MMIO 0xe6c50000 (irq = 81, base_baud = 0) is a scifa sh-sci.2: ttySC2 at MMIO 0xe6c60000 (irq = 82, base_baud = 0) is a scifa sh-sci.3: ttySC3 at MMIO 0xe6c70000 (irq = 83, base_baud = 0) is a scifa sh-sci.4: ttySC4 at MMIO 0xe6c80000 (irq = 89, base_baud = 0) is a scifa sh-sci.5: ttySC5 at MMIO 0xe6cb0000 (irq = 90, base_baud = 0) is a scifa sh-sci.6: ttySC6 at MMIO 0xe6c30000 (irq = 91, base_baud = 0) is a scifb physmap platform flash device: 08000000 at 20000000 physmap-flash.0: Found 1 x16 devices at 0x0 in 16-bit bank. Manufacturer ID 0x000089 Chip ID 0x0088b1 Intel/Sharp Extended Query Table at 0x010A Intel/Sharp Extended Query Table at 0x010A Intel/Sharp Extended Query Table at 0x010A Intel/Sharp Extended Query Table at 0x010A Intel/Sharp Extended Query Table at 0x010A Using buffer write method Using auto-unlock on power-up/resume cfi_cmdset_0001: Erase suspend on write enabled Creating 5 MTD partitions on "physmap-flash.0": 0x000000000000-0x000000080000 : "loader" 0x000000080000-0x000000100000 : "bootenv" 0x000000100000-0x000000900000 : "kernel_ro" 0x000000900000-0x000001100000 : "kernel" 0x000001100000-0x000008000000 : "data" smsc911x: Driver version 2008-10-21 smsc911x smsc911x (unregistered net_device): couldn't get clock -2 libphy: smsc911x-mdio: probed smsc911x smsc911x eth0: attached PHY driver [Generic PHY] (mii_bus:phy_addr=smsc911x-fffffff:01, irq=-1) smsc911x smsc911x eth0: MAC Address: 74:90:50:00:00:ac usbcore: registered new interface driver usb-storage renesas_usbhs renesas_usbhs.0: host probed renesas_usbhs renesas_usbhs.0: gadget probed renesas_usbhs renesas_usbhs.0: probed renesas_usbhs renesas_usbhs.1: host probed renesas_usbhs renesas_usbhs.1: gadget probed renesas_usbhs renesas_usbhs.1: probed renesas_usbhs renesas_usbhs.1: USB Host Controller renesas_usbhs renesas_usbhs.1: new USB bus registered, assigned bus number 1 mousedev: PS/2 mouse device common for all mice hub 1-0:1.0: USB hub found hub 1-0:1.0: 1 port detected sh_mobile_sdhi sh_mobile_sdhi.0: mmc0 base at 0xe6850000 clock rate 99 MHz sh_mobile_sdhi sh_mobile_sdhi.2: mmc1 base at 0xe6870000 clock rate 99 MHz sh_mmcif sh_mmcif.0: driver version 2010-04-28 usbcore: registered new interface driver usbhid usbhid: USB HID core driver Unable to handle kernel NULL pointer dereference at virtual address 0000008c pgd = c0004000 [0000008c] *pgd=00000000 Internal error: Oops: 805 [#1] ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper Not tainted 3.13.0-rc7-next-20140107-00005-gad2dc8a #771 task: c8041bc0 ti: c8042000 task.ti: c8042000 PC is at asoc_simple_card_probe+0x44/0x2a4 LR is at platform_drv_probe+0x20/0x50 pc : [<c02b1648>] lr : [<c01f33b0>] psr: 60000113 sp : c8043da0 ip : c8043dd0 fp : c8043dcc r10: c048a278 r9 : c8042000 r8 : 00000000 r7 : 00000000 r6 : c04a5a50 r5 : c04bf58c r4 : c04a5a40 r3 : c02b1604 r2 : 00000000 r1 : 00000000 r0 : c04a5a40 Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c5387d Table: 40004019 DAC: 00000015 Process swapper (pid: 1, stack limit = 0xc8042238) Stack: (0xc8043da0 to 0xc8044000) 3da0: 00000000 00000000 c04a5a50 c04bf58c 00000000 c04bf58c c0482c2c c048a278 3dc0: c8043de4 c8043dd0 c01f33b0 c02b1610 c04e4af0 c04a5a50 c8043e04 c8043de8 3de0: c01f1e28 c01f339c c04a5a50 c04bf58c c04a5a84 00000000 c8043e24 c8043e08 3e00: c01f1fd4 c01f1d20 c807f5bc 00000000 c04bf58c c01f1f40 c8043e4c c8043e28 3e20: c01f057c c01f1f4c c8030e4c c807f5b0 c839b6b4 c04bf58c c839b680 c04b6f30 3e40: c8043e5c c8043e50 c01f1934 c01f052c c8043e84 c8043e60 c01f1538 c01f191c 3e60: c0412b28 c8043e70 c04bf58c 00000006 c04c41c0 00000070 c8043e9c c8043e88 3e80: c01f265c c01f1468 c0491070 00000006 c8043eac c8043ea0 c01f3304 c01f25e8 3ea0: c8043ebc c8043eb0 c0482c44 c01f32c0 c8043f4c c8043ec0 c00094e8 c0482c38 3ec0: 00000264 c046ab4c 00000000 00000000 c8043efc c8043ee0 c8043f04 c8043ee8 3ee0: c8043f00 c8043ef0 c046c4d0 c06f0daf c0386f80 00000070 c8043f4c c8043f08 3f00: c00348bc c046c4dc c048a250 00000006 00000006 c046a44c 00000000 c043536c 3f20: c8043f4c c0491070 00000006 c04c41c0 00000070 c046c4d0 c048a26c c048a278 3f40: c8043f94 c8043f50 c046cb50 c0009400 00000006 00000006 c046c4d0 16a3cbde 3f60: a0080220 6244bd2d 21509030 00000000 c035765c 00000000 00000000 00000000 3f80: 00000000 00000000 c8043fac c8043f98 c035766c c046ca6c 00000000 00000000 3fa0: 00000000 c8043fb0 c000f478 c0357668 00000000 00000000 00000000 00000000 3fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 3fe0: 00000000 00000000 00000000 00000000 00000013 00000000 00000002 54f1c041 Backtrace: [<c02b1604>] (asoc_simple_card_probe) from [<c01f33b0>] (platform_drv_probe+0x20/0x50) r10:c048a278 r8:c0482c2c r7:c04bf58c r6:00000000 r5:c04bf58c r4:c04a5a50 [<c01f3390>] (platform_drv_probe) from [<c01f1e28>] (driver_probe_device+0x114/0x22c) r5:c04a5a50 r4:c04e4af0 [<c01f1d14>] (driver_probe_device) from [<c01f1fd4>] (__driver_attach+0x94/0x98) r7:00000000 r6:c04a5a84 r5:c04bf58c r4:c04a5a50 [<c01f1f40>] (__driver_attach) from [<c01f057c>] (bus_for_each_dev+0x5c/0x90) r6:c01f1f40 r5:c04bf58c r4:00000000 r3:c807f5bc [<c01f0520>] (bus_for_each_dev) from [<c01f1934>] (driver_attach+0x24/0x28) r6:c04b6f30 r5:c839b680 r4:c04bf58c [<c01f1910>] (driver_attach) from [<c01f1538>] (bus_add_driver+0xdc/0x1d8) [<c01f145c>] (bus_add_driver) from [<c01f265c>] (driver_register+0x80/0xfc) r7:00000070 r6:c04c41c0 r5:00000006 r4:c04bf58c [<c01f25dc>] (driver_register) from [<c01f3304>] (__platform_driver_register+0x50/0x64) r5:00000006 r4:c0491070 [<c01f32b4>] (__platform_driver_register) from [<c0482c44>] (asoc_simple_card_init+0x18/0x20) [<c0482c2c>] (asoc_simple_card_init) from [<c00094e8>] (do_one_initcall+0xf4/0x150) [<c00093f4>] (do_one_initcall) from [<c046cb50>] (kernel_init_freeable+0xf0/0x1b8) r10:c048a278 r9:c048a26c r8:c046c4d0 r7:00000070 r6:c04c41c0 r5:00000006 r4:c0491070 [<c046ca60>] (kernel_init_freeable) from [<c035766c>] (kernel_init+0x10/0x118) r10:00000000 r9:00000000 r8:00000000 r7:00000000 r6:00000000 r5:c035765c r4:00000000 [<c035765c>] (kernel_init) from [<c000f478>] (ret_from_fork+0x14/0x3c) r4:00000000 r3:00000000 Code: ebff5df2 e1500007 1a000031 e3a07000 (e587608c) ---[ end trace fa979cfc03b9a020 ]--- Kernel panic - not syncing: Attempted to kill init! exitcode=0x0000000b

4 3

[alsa-devel] On mixer controls naming
by R. Lemos 14 Jan '14

14 Jan '14

Hi, I'm trying to write a new driver for a virtual sound card. It isn't actually a sound card, but only a software solution to control volume and bass of an external device (using SPI and GPIO). So it only has a mixer (no pcm or capture subdevices). I could successfully configure two controls "Master Playback Switch" and "Master Playback Volume". Alsamixer shows them correctly under "Playback" view (and very nicely it joined them together to form a single composite control "Master") The other control I've named "Tone Control - Bass", as indicated in writing-an-alsa-driver.pdf. But alsamixer shows it under both "Playback" and "Capture" views. How can I restrict this latter control to just playback? -- []'s Rodrigo

2 4

[alsa-devel] [PATCH v3] ASoC: imx: Allow using sound with DT descriptions only
by Alexander Shiyan 13 Jan '14

13 Jan '14

This patch allow using generic sound cards/codecs based on DT description. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/Kconfig | 43 +++++++++++++++++-------------------------- 1 file changed, 17 insertions(+), 26 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 1c7b3b7..8249000 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -7,6 +7,7 @@ config SND_SOC_FSL_SSI config SND_SOC_FSL_SPDIF tristate + select REGMAP_MMIO config SND_SOC_FSL_UTILS tristate @@ -103,27 +104,32 @@ config SND_MPC52xx_SOC_EFIKA endif # SND_POWERPC_SOC -menuconfig SND_IMX_SOC - tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC || COMPILE_TEST - help - Say Y or M if you want to add support for codecs attached to - the i.MX CPUs. - -if SND_IMX_SOC - -config SND_SOC_IMX_SSI +config SND_SOC_IMX_AUDMUX tristate config SND_SOC_IMX_PCM_FIQ tristate select FIQ + default y if SND_SOC_IMX_SSI=y || SND_SOC_FSL_SSI=y config SND_SOC_IMX_PCM_DMA tristate select SND_SOC_GENERIC_DMAENGINE_PCM + default y if SND_SOC_IMX_SSI=y || SND_SOC_FSL_SSI=y -config SND_SOC_IMX_AUDMUX +menuconfig SND_IMX_SOC + tristate "SoC Audio for Freescale i.MX CPUs" + depends on ARCH_MXC || COMPILE_TEST + select SND_SOC_FSL_SSI if OF + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_PCM_DMA + help + Say Y or M if you want to add support for codecs attached to + the i.MX CPUs. + +if SND_IMX_SOC + +config SND_SOC_IMX_SSI tristate config SND_MXC_SOC_WM1133_EV1 @@ -131,7 +137,6 @@ config SND_MXC_SOC_WM1133_EV1 depends on MACH_MX31ADS_WM1133_EV1 select SND_SOC_WM8350 select SND_SOC_IMX_PCM_FIQ - select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help Enable support for audio on the i.MX31ADS with the WM1133-EV1 @@ -141,8 +146,6 @@ config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" depends on MACH_IMX27_VISSTRIM_M10 && I2C select SND_SOC_TLV320AIC32X4 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help Say Y if you want to add support for SoC audio on Visstrim SM10 @@ -154,7 +157,6 @@ config SND_SOC_PHYCORE_AC97 select SND_SOC_AC97_BUS select SND_SOC_WM9712 select SND_SOC_IMX_PCM_FIQ - select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help Say Y if you want to add support for SoC audio on Phytec phyCORE @@ -169,7 +171,6 @@ config SND_SOC_EUKREA_TLV320 depends on I2C select SND_SOC_TLV320AIC23 select SND_SOC_IMX_PCM_FIQ - select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help Enable I2S based access to the TLV320AIC23B codec attached @@ -179,9 +180,6 @@ config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C select SND_SOC_WM8962 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX - select SND_SOC_FSL_SSI help Say Y if you want to add support for SoC audio on an i.MX board with a wm8962 codec. @@ -190,18 +188,13 @@ config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C select SND_SOC_SGTL5000 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX - select SND_SOC_FSL_SSI help Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" - select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF - select REGMAP_MMIO help SoC Audio support for i.MX boards with S/PDIF Say Y if you want to add support for SoC audio on an i.MX board with @@ -211,8 +204,6 @@ config SND_SOC_IMX_MC13783 tristate "SoC Audio support for I.MX boards with mc13783" depends on MFD_MC13XXX && ARM select SND_SOC_IMX_SSI - select SND_SOC_IMX_AUDMUX select SND_SOC_MC13783 - select SND_SOC_IMX_PCM_DMA endif # SND_IMX_SOC -- 1.8.3.2

3 5

[alsa-devel] [PATCH] ASoC: simple-card: use snd_soc_card_set/get_drvdata
by Xiubo Li 13 Jan '14

13 Jan '14

Remove asoc_simple_get_card_info macro and use snd_soc_card_set_drvdata and snd_soc_card_get_drvdata instead. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index dbd93cc..c025b6d 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -14,9 +14,6 @@ #include <linux/module.h> #include <sound/simple_card.h> -#define asoc_simple_get_card_info(p) \ - container_of(p->dai_link, struct asoc_simple_card_info, snd_link) - static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set, unsigned int daifmt) @@ -41,7 +38,8 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *info = asoc_simple_get_card_info(rtd); + struct asoc_simple_card_info *info = + snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; unsigned int daifmt = info->daifmt; @@ -268,6 +266,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo->snd_card.dai_link = &cinfo->snd_link; cinfo->snd_card.num_links = 1; + snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); + return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); } -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: fsl-sai: Add device tree nodes and its availiable check
by Xiubo Li 13 Jan '14

13 Jan '14

SAIs that are not connected on the board should have a status = "disabled" property in their device tree nodes. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index cdd3fa8..6b2ab91 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -404,6 +404,9 @@ static int fsl_sai_probe(struct platform_device *pdev) struct resource *res; int ret; + if (!np || !of_device_is_available(np)) + return -ENODEV; + sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) return -ENOMEM; -- 1.8.4

3 3

[alsa-devel] [PATCH] ALSA: hda - Apply codec power_filter to FG nodes
by Takashi Iwai 13 Jan '14

13 Jan '14

Apply the codec->power_filter to the FG nodes in general for reducing hackish set_power_state ops override in patch_sigmatel.c. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 12 +++++++++--- sound/pci/hda/hda_generic.c | 5 +++-- sound/pci/hda/hda_generic.h | 3 +++ sound/pci/hda/patch_sigmatel.c | 36 ++++++++++++------------------------ 4 files changed, 27 insertions(+), 29 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ab2a991d916f..ec4536c8d8d4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3927,6 +3927,8 @@ unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state) { + if (nid == codec->afg || nid == codec->mfg) + return power_state; if (power_state == AC_PWRST_D3 && get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_PIN && (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)) { @@ -3965,9 +3967,13 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, codec->patch_ops.set_power_state(codec, fg, power_state); else { - snd_hda_codec_read(codec, fg, flags, - AC_VERB_SET_POWER_STATE, - power_state); + state = power_state; + if (codec->power_filter) + state = codec->power_filter(codec, fg, state); + if (state == power_state || power_state != AC_PWRST_D3) + snd_hda_codec_read(codec, fg, flags, + AC_VERB_SET_POWER_STATE, + state); snd_hda_codec_set_power_to_all(codec, fg, power_state); } state = hda_sync_power_state(codec, fg, power_state); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 7513089c8985..8321a97d5c05 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4312,11 +4312,11 @@ static int check_auto_mic_availability(struct hda_codec *codec) } /* power_filter hook; make inactive widgets into power down */ -static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, +unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state) { - if (power_state != AC_PWRST_D0) + if (power_state != AC_PWRST_D0 || nid == codec->afg) return power_state; if (get_wcaps_type(get_wcaps(codec, nid)) >= AC_WID_POWER) return power_state; @@ -4324,6 +4324,7 @@ static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, return power_state; return AC_PWRST_D3; } +EXPORT_SYMBOL_GPL(snd_hda_gen_path_power_filter); /* mute all aamix inputs initially; parse up to the first leaves */ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 0929a06df812..07f767231c9f 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -335,5 +335,8 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec); #ifdef CONFIG_PM int snd_hda_gen_check_power_status(struct hda_codec *codec, hda_nid_t nid); #endif +unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, + hda_nid_t nid, + unsigned int power_state); #endif /* __SOUND_HDA_GENERIC_H */ diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 088a5afbd1b9..6998cf29b9bc 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -368,6 +368,17 @@ static int stac_vrefout_set(struct hda_codec *codec, return 1; } +/* prevent codec AFG to D3 state when vref-out pin is used for mute LED */ +/* this hook is set in stac_setup_gpio() */ +static unsigned int stac_vref_led_power_filter(struct hda_codec *codec, + hda_nid_t nid, + unsigned int power_state) +{ + if (nid == codec->afg && power_state == AC_PWRST_D3) + return AC_PWRST_D1; + return snd_hda_gen_path_power_filter(codec, nid, power_state); +} + /* update mute-LED accoring to the master switch */ static void stac_update_led_status(struct hda_codec *codec, int enabled) { @@ -4260,30 +4271,8 @@ static int stac_suspend(struct hda_codec *codec) stac_shutup(codec); return 0; } - -static void stac_set_power_state(struct hda_codec *codec, hda_nid_t fg, - unsigned int power_state) -{ - unsigned int afg_power_state = power_state; - struct sigmatel_spec *spec = codec->spec; - - if (power_state == AC_PWRST_D3) { - if (spec->vref_mute_led_nid) { - /* with vref-out pin used for mute led control - * codec AFG is prevented from D3 state - */ - afg_power_state = AC_PWRST_D1; - } - /* this delay seems necessary to avoid click noise at power-down */ - msleep(100); - } - snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, - afg_power_state); - snd_hda_codec_set_power_to_all(codec, fg, power_state); -} #else #define stac_suspend NULL -#define stac_set_power_state NULL #endif /* CONFIG_PM */ static const struct hda_codec_ops stac_patch_ops = { @@ -4466,8 +4455,7 @@ static void stac_setup_gpio(struct hda_codec *codec) spec->gpio_dir |= spec->gpio_led; spec->gpio_data |= spec->gpio_led; } else { - codec->patch_ops.set_power_state = - stac_set_power_state; + codec->power_filter = stac_vref_led_power_filter; } } -- 1.8.5.2

1 0

[alsa-devel] [PATCH 1/1] PATCH - alsa-plugins: Modify jack client name
by Corfu, ValentinX 13 Jan '14

13 Jan '14

The current jack client name contains the process id of the application providing the audio samples. This leads to unpredictable jack client names which makes handling of the connections by a controlling application very hard. The modification now allows to specify a client name and the number of channels as parameters in the configuration file The given number of channels are then created but not connected. The implementation is backward compatible and simply adds a new configuration option to the plugin. -------------------------------------------------------------- Intel Shannon Limited Registered in Ireland Registered Office: Collinstown Industrial Park, Leixlip, County Kildare Registered Number: 308263 Business address: Dromore House, East Park, Shannon, Co. Clare This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

2 1

[alsa-devel] [PATCH] ALSA: hda - Don't set indep_hp flag for old AD codecs
by Takashi Iwai 13 Jan '14

13 Jan '14

Some old AD codecs don't like the independent HP handling, either it contains a single DAC (AD1981) or it mandates the mixer routing (AD1986A). This patch removes the indep_hp flag for such codecs. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=68081 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 819d4a98abaf..7a426ed491f2 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -185,7 +185,7 @@ static const struct hda_codec_ops ad198x_auto_patch_ops = { }; -static int ad198x_parse_auto_config(struct hda_codec *codec) +static int ad198x_parse_auto_config(struct hda_codec *codec, bool indep_hp) { struct ad198x_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->gen.autocfg; @@ -195,7 +195,7 @@ static int ad198x_parse_auto_config(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->no_sticky_stream = 1; - spec->gen.indep_hp = 1; + spec->gen.indep_hp = indep_hp; spec->gen.add_stereo_mix_input = 1; err = snd_hda_parse_pin_defcfg(codec, cfg, NULL, 0); @@ -379,7 +379,7 @@ static int patch_ad1986a(struct hda_codec *codec) ad1986a_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = ad198x_parse_auto_config(codec); + err = ad198x_parse_auto_config(codec, false); if (err < 0) { snd_hda_gen_free(codec); return err; @@ -481,7 +481,7 @@ static int patch_ad1983(struct hda_codec *codec) spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); - err = ad198x_parse_auto_config(codec); + err = ad198x_parse_auto_config(codec, false); if (err < 0) goto error; err = ad1983_add_spdif_mux_ctl(codec); @@ -568,7 +568,7 @@ static int patch_ad1981(struct hda_codec *codec) snd_hda_pick_fixup(codec, NULL, ad1981_fixup_tbl, ad1981_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = ad198x_parse_auto_config(codec); + err = ad198x_parse_auto_config(codec, false); if (err < 0) goto error; err = ad1983_add_spdif_mux_ctl(codec); @@ -894,7 +894,7 @@ static int patch_ad1988(struct hda_codec *codec) snd_hda_pick_fixup(codec, ad1988_fixup_models, NULL, ad1988_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = ad198x_parse_auto_config(codec); + err = ad198x_parse_auto_config(codec, true); if (err < 0) goto error; err = ad1988_add_spdif_mux_ctl(codec); @@ -1071,7 +1071,7 @@ static int patch_ad1884(struct hda_codec *codec) snd_hda_pick_fixup(codec, NULL, ad1884_fixup_tbl, ad1884_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = ad198x_parse_auto_config(codec); + err = ad198x_parse_auto_config(codec, true); if (err < 0) goto error; err = ad1983_add_spdif_mux_ctl(codec); @@ -1113,7 +1113,7 @@ static int patch_ad1882(struct hda_codec *codec) spec->gen.mixer_merge_nid = 0x21; spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); - err = ad198x_parse_auto_config(codec); + err = ad198x_parse_auto_config(codec, true); if (err < 0) goto error; err = ad1988_add_spdif_mux_ctl(codec); -- 1.8.5.2

1 0

Re: [alsa-devel] [PATCH] ASoC: simple-card: fix one bug to writing to the platform data
by Jean-Francois Moine 13 Jan '14

13 Jan '14

Xiubo Li wrote: > It's a bug that writing to the platform data directly, for it should > be constant. So just copy it before writing. > > Signed-off-by: Xiubo Li <Li.Xiubo at freescale.com> > --- > sound/soc/generic/simple-card.c | 40 +++++++++++++++++++++------------------- > 1 file changed, 21 insertions(+), 19 deletions(-) > > diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c > index 406e4ea..5b65324 100644 > --- a/sound/soc/generic/simple-card.c > +++ b/sound/soc/generic/simple-card.c [snip] > @@ -204,36 +205,37 @@ static int asoc_simple_card_probe(struct platform_device *pdev) > struct device_node *np = pdev->dev.of_node; > struct device_node *of_cpu, *of_codec, *of_platform; > struct device *dev = &pdev->dev; > + int ret; > > cinfo = NULL; > of_cpu = NULL; > of_codec = NULL; > of_platform = NULL; > + > + cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); > + if (!cinfo) > + return -ENOMEM; > + > if (np && of_device_is_available(np)) { > - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); > - if (cinfo) { > - int ret; > - cinfo->snd_card.dev = &pdev->dev; > - ret = asoc_simple_card_parse_of(np, cinfo, dev, > - &of_cpu, > - &of_codec, > - &of_platform); > - if (ret < 0) { > - if (ret != -EPROBE_DEFER) > - dev_err(dev, "parse error %d\n", ret); > - return ret; > - } > - } else { > - return -ENOMEM; > + cinfo->snd_card.dev = dev; > + > + ret = asoc_simple_card_parse_of(np, cinfo, dev, > + &of_cpu, > + &of_codec, > + &of_platform); > + if (ret < 0) { > + if (ret != -EPROBE_DEFER) > + dev_err(dev, "parse error %d\n", ret); > + return ret; > } > } else { > - cinfo = pdev->dev.platform_data; > - if (!cinfo) { > + if (!pdev->dev.platform_data) { > dev_err(dev, "no info for asoc-simple-card\n"); > return -EINVAL; > } > > - cinfo->snd_card.dev = &pdev->dev; > + memcpy(cinfo, pdev->dev.platform_data, sizeof(cinfo)); > + cinfo->snd_card.dev = dev; > } > > if (!cinfo->name || If the original cinfo is not used anymore, the use of its structure to handle the card information is not a good idea: - almost all cinfo information are in the struct snd_soc_card, - this cinfo structure cannot be extended to handle many DAI links, - it contains simple-card information which are of no use for the platform caller. So, I'd rather have seen: - the removal of 'snd_link' and 'snd_card' from the platform interface (struct asoc_simple_card_info), - the definition of a local struct simple_card_data containing the struct snd_soc_card and a pointer to an array of fmt/sysclk values (one per DAI link). -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

3 3

[alsa-devel] [PATCH] ASoC: wm8804: Allow control of master clock divider in PLL generation
by Daniel Matuschek 13 Jan '14

13 Jan '14

Signed-off-by: Daniel Matuschek <daniel(a)matuschek.net> After some discussions of the patch last week, here is a new version. Simply reducing the post_table did not work, as for some frequencies both settings (MCLKDIV=0 and MCLKDIV=1) are needed (e.g. 96 and 192kHz) WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using an additional mclk_div divider, is is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. The behavior has to be controlled by the sound card driver, because some sample frequency share the same setting. e.g. 192kHz and 96kHz use 24.576MHz master clock. The only difference is the MCLK divider. --- sound/soc/codecs/wm8804.c | 17 ++++++++++++++--- sound/soc/codecs/wm8804.h | 4 ++++ 2 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 1704b1e..4619bf8 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -63,6 +63,7 @@ struct wm8804_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; + int mclk_div; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -318,7 +319,7 @@ static struct { #define FIXED_PLL_SIZE ((1ULL << 22) * 10) static int pll_factors(struct pll_div *pll_div, unsigned int target, - unsigned int source) + unsigned int source, unsigned int mclk_div) { u64 Kpart; unsigned long int K, Ndiv, Nmod, tmp; @@ -330,7 +331,8 @@ static int pll_factors(struct pll_div *pll_div, unsigned int target, */ for (i = 0; i < ARRAY_SIZE(post_table); i++) { tmp = target * post_table[i].div; - if (tmp >= 90000000 && tmp <= 100000000) { + if ((tmp >= 90000000 && tmp <= 100000000) && + (mclk_div == post_table[i].mclkdiv)) { pll_div->freqmode = post_table[i].freqmode; pll_div->mclkdiv = post_table[i].mclkdiv; target *= post_table[i].div; @@ -387,8 +389,12 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, } else { int ret; struct pll_div pll_div; + struct wm8804_priv *wm8804; - ret = pll_factors(&pll_div, freq_out, freq_in); + wm8804 = snd_soc_codec_get_drvdata(codec); + + ret = pll_factors(&pll_div, freq_out, freq_in, + wm8804->mclk_div); if (ret) return ret; @@ -452,6 +458,7 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { struct snd_soc_codec *codec; + struct wm8804_priv *wm8804; codec = dai->codec; switch (div_id) { @@ -459,6 +466,10 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, snd_soc_update_bits(codec, WM8804_PLL5, 0x30, (div & 0x3) << 4); break; + case WM8804_MCLK_DIV: + wm8804 = snd_soc_codec_get_drvdata(codec); + wm8804->mclk_div = div; + break; default: dev_err(dai->dev, "Unknown clock divider: %d\n", div_id); return -EINVAL; diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index 8ec14f5..e72d4f4 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -57,5 +57,9 @@ #define WM8804_CLKOUT_SRC_OSCCLK 4 #define WM8804_CLKOUT_DIV 1 +#define WM8804_MCLK_DIV 2 + +#define WM8804_MCLKDIV_256FS 0 +#define WM8804_MCLKDIV_128FS 1 #endif /* _WM8804_H */ -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: dpcm: don't do hw_param when BE has done hw_param
by Nenghua Cao 13 Jan '14

13 Jan '14

From: Nenghua Cao <nhcao(a)marvell.com> It fixes the following case: Two FEs connects the same BE; FE1 & BE path has been opened and hw_paramed. At this momment, FE2 & BE path is being opened and hw_paramed. The BE dai will do hw_param again even if it has done hw_param. It is not reasonable. FE1------------>BE FE2-------------^ Signed-off-by: Nenghua Cao <nhcao(a)marvell.com> --- sound/soc/soc-pcm.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 891b9a9..ec07e37 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1339,7 +1339,6 @@ static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) continue; if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE)) continue; -- 1.7.0.4

4 15

[alsa-devel] [PATCH] alsa-restore.rules: refer to correct attr
by Dave Reisner 13 Jan '14

13 Jan '14

$attr{number} in the RUN rule is an empty expansion. This makes sense, because the path doesn't exist -- i.e., it refers to the path: /sys/devices/pci0000:00/foo/bar/sound/card0/controlC0/number Instead, refer to $attr{device/number}, which does exist. Signed-off-by: Dave Reisner <dreisner(a)archlinux.org> --- This seems to have been broken when it was first introduced a little over three years ago (de7c3eff0e). Please CC me on replies as I am not subscribed to the list. alsactl/90-alsa-restore.rules.in | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/alsactl/90-alsa-restore.rules.in b/alsactl/90-alsa-restore.rules.in index 88e12e0..c68119d 100644 --- a/alsactl/90-alsa-restore.rules.in +++ b/alsactl/90-alsa-restore.rules.in @@ -2,7 +2,7 @@ ACTION=="add", SUBSYSTEM=="sound", KERNEL=="controlC*", KERNELS!="card*", GOTO=" GOTO="alsa_restore_end" LABEL="alsa_restore_go" -TEST!="@daemonswitch@", RUN+="@sbindir@/alsactl restore $attr{number}" -TEST=="@daemonswitch@", RUN+="@sbindir@/alsactl nrestore $attr{number}" +TEST!="@daemonswitch@", RUN+="@sbindir@/alsactl restore $attr{device/number}" +TEST=="@daemonswitch@", RUN+="@sbindir@/alsactl nrestore $attr{device/number}" LABEL="alsa_restore_end" -- 1.8.5.2

3 2

[alsa-devel] [PATCH] ASoC: simple-card: remove snd_link and snd_card from struct asoc_xx_info
by Xiubo Li 13 Jan '14

13 Jan '14

Whether the dt is used or not, almost all the simple card information for the DAI link and sound card are initialized in the simple card driver. And for the platform caller, the snd_link and snd_card are of no use, so remove it from struct asoc_simple_card_info, and let them to be the simple card driver's global data. This is also fix one bug about writing to the platform data directly, for it should be constant. And now only one DAI link is supported for simple card. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- This patch maybe not very perfect, such as supporting many DAI links, etc. And there still need followed patches to fulfil these. include/sound/simple_card.h | 4 --- sound/soc/generic/simple-card.c | 54 +++++++++++++++++++++++------------------ 2 files changed, 30 insertions(+), 28 deletions(-) diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 6c74527..e1ac996 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -29,10 +29,6 @@ struct asoc_simple_card_info { unsigned int daifmt; struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; - - /* used in simple-card.c */ - struct snd_soc_dai_link snd_link; - struct snd_soc_card snd_card; }; #endif /* __SIMPLE_CARD_H */ diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index dbd93cc..94dc743 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -14,9 +14,6 @@ #include <linux/module.h> #include <sound/simple_card.h> -#define asoc_simple_get_card_info(p) \ - container_of(p->dai_link, struct asoc_simple_card_info, snd_link) - static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set, unsigned int daifmt) @@ -41,7 +38,8 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *info = asoc_simple_get_card_info(rtd); + struct asoc_simple_card_info *info = + snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; unsigned int daifmt = info->daifmt; @@ -58,6 +56,16 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static struct snd_soc_dai_link simple_dai_link = { + .init = asoc_simple_card_dai_init, +}; + +static struct snd_soc_card simple_snd_card = { + .owner = THIS_MODULE, + .dai_link = &simple_dai_link, + .num_links = 1, +}; + static int asoc_simple_card_sub_parse_of(struct device_node *np, struct asoc_simple_dai *dai, @@ -144,7 +152,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, /* DAPM routes */ if (of_property_read_bool(node, "simple-audio-card,routing")) { - ret = snd_soc_of_parse_audio_routing(&info->snd_card, + ret = snd_soc_of_parse_audio_routing(&simple_snd_card, "simple-audio-card,routing"); if (ret) return ret; @@ -213,7 +221,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); if (cinfo) { int ret; - cinfo->snd_card.dev = &pdev->dev; + simple_snd_card.dev = dev; ret = asoc_simple_card_parse_of(np, cinfo, dev, &of_cpu, &of_codec, @@ -233,7 +241,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return -EINVAL; } - cinfo->snd_card.dev = &pdev->dev; + simple_snd_card.dev = dev; } if (!cinfo->name || @@ -247,28 +255,26 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } /* - * init snd_soc_dai_link + * init simple_dai_link */ - cinfo->snd_link.name = cinfo->name; - cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; - cinfo->snd_link.platform_name = cinfo->platform; - cinfo->snd_link.codec_name = cinfo->codec; - cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; - cinfo->snd_link.cpu_of_node = of_cpu; - cinfo->snd_link.codec_of_node = of_codec; - cinfo->snd_link.platform_of_node = of_platform; - cinfo->snd_link.init = asoc_simple_card_dai_init; + simple_dai_link.name = cinfo->name; + simple_dai_link.stream_name = cinfo->name; + simple_dai_link.cpu_dai_name = cinfo->cpu_dai.name; + simple_dai_link.platform_name = cinfo->platform; + simple_dai_link.codec_name = cinfo->codec; + simple_dai_link.codec_dai_name = cinfo->codec_dai.name; + simple_dai_link.cpu_of_node = of_cpu; + simple_dai_link.codec_of_node = of_codec; + simple_dai_link.platform_of_node = of_platform; /* - * init snd_soc_card + * init simple_snd_card */ - cinfo->snd_card.name = cinfo->card; - cinfo->snd_card.owner = THIS_MODULE; - cinfo->snd_card.dai_link = &cinfo->snd_link; - cinfo->snd_card.num_links = 1; + simple_snd_card.name = cinfo->card; + + snd_soc_card_set_drvdata(&simple_snd_card, cinfo); - return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); + return devm_snd_soc_register_card(dev, &simple_snd_card); } static const struct of_device_id asoc_simple_of_match[] = { -- 1.8.4

3 2

[alsa-devel] How to make Realtek HD-A codec alc282 create "Input Source" control to select a input for recording?
by Lin, Mengdong 13 Jan '14

13 Jan '14

Hi, How to make the driver create a "input source" control for ALC282 with 3 Mic pins? My platform uses a Realtek HD-A codec ALC282 card, it has 3 Mic pins: Mic = 0x18 (MIC1) Internal Mic = 0x19 (MIC2) Internal Mic = 0x12 (DMIC) Recording on Mic1 (0x18) cannot work because the mixer 0x23 always mutes the input from 0x18. And there is no "input source" control to let the mixer select and unmute 0x18. I've tried to disable pin sense on pin 0x18 as a board fix-up like below, but "input source" control is still not created: caps = snd_hda_query_pin_caps(codec, 0x18); caps &= ~AC_PINCAP_PRES_DETECT; snd_hda_override_pin_caps(codec, 0x18, caps); snd_hda_codec_set_pincfg(codec, 0x18, 0x03a19120); ... set 'Jack Detect Override' Thanks Mengdong Here is the alsa-info: upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.61 !!################################ !!Script ran on: Thu Jan 9 06:09:23 UTC 2014 !!Linux Distribution !!------------------ Ubuntu Saucy Salamander (development branch) \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu Saucy Salamander (development branch)" NAME="Ubuntu" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu Saucy Salamander (development branch)" HOME_URL="http://www.ubuntu.com/" SUPPORT_URL="http://help.ubuntu.com/" BUG_REPORT_URL="http://bugs.launchpad.net/ubuntu/" !!DMI Information !!--------------- Manufacturer: Intel Corporation Product Name: Broadwell Client platform Product Version: 0.1 Firmware Version: BDW-E1R1.86C.0052.R01.1311260206 !!Kernel Information !!------------------ Kernel release: 3.13.0-rc4-drm-bdw+ Operating System: GNU/Linux Architecture: x86_64 Processor: x86_64 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k3.13.0-rc4-drm-bdw+ Library version: 1.0.27.2 Utilities version: 1.0.27.1 !!Loaded ALSA modules !!------------------- snd_hda_intel snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xd1030000 irq 61 1 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xd1034000 irq 59 !!PCI Soundcards installed in the system !!-------------------------------------- 00:03.0 Audio device: Intel Corporation Device 160c (rev 04) 00:1b.0 Audio device: Intel Corporation Device 9ca0 !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:03.0 0403: 8086:160c (rev 04) Subsystem: 8086:2010 -- 00:1b.0 0403: 8086:9ca0 Subsystem: 8086:7270 !!Modprobe options (Sound related) !!-------------------------------- snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 snd_usb_audio: index=-2 snd_usb_caiaq: index=-2 snd_usb_ua101: index=-2 snd_usb_us122l: index=-2 snd_usb_usx2y: index=-2 snd_cmipci: mpu_port=0x330 fm_port=0x388 snd_pcsp: index=-2 snd_usb_audio: index=-2 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 32,1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 32,1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Intel Broadwell HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862808 Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D3, actual=D3 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D3, actual=D3 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=3 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="ELD", index=0, device=3 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=01, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=7 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=1, device=0 Control: name="IEC958 Playback Pro Mask", index=1, device=0 Control: name="IEC958 Playback Default", index=1, device=0 Control: name="IEC958 Playback Switch", index=1, device=0 Control: name="ELD", index=0, device=7 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=02, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 3 0x02* 0x03 0x04 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=8 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=2, device=0 Control: name="IEC958 Playback Pro Mask", index=2, device=0 Control: name="IEC958 Playback Default", index=2, device=0 Control: name="IEC958 Playback Switch", index=2, device=0 Control: name="ELD", index=0, device=8 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=03, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono Codec: Realtek ALC282 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0282 Subsystem Id: 0x10ec0282 Revision Id: 0x100002 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=3, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC282 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x54 0x54] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC282 Analog", type="Audio", device=0 Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 Amp-In vals: [0x29 0x29] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 Amp-In vals: [0x97 0x97] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x22 Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Internal Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 5 0x18 0x19 0x1a 0x1b 0x1d Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x0d Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 Amp-In vals: [0x29 0x29] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Phantom Jack", index=1, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x90a60930: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00010014: OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x0c Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x17 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x0f Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x03 0x03] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x03a19820: [Jack] Mic at Ext Left Conn = 1/8, Color = Pink DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x19 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x90a7092f: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x2, Sequence = 0xf Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1a [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1d [Pin Complex] wcaps 0x400400: Mono Pincap 0x00000020: IN Pin Default 0x40189d2d: [N/A] Speaker at Ext N/A Conn = DIN, Color = Pink DefAssociation = 0x2, Sequence = 0xd Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1e [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=117 Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000001c: OUT HP Detect Pin Default 0x0321401f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c 0x0d* Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 6 0x18 0x19 0x1a 0x1b 0x1d 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 6 0x18 0x19 0x1a 0x1b 0x1d 0x0b --endcollapse--

2 5

[alsa-devel] [PATCH] ASoC: wm8804: Cleanup of post_table
by Daniel Matuschek 12 Jan '14

12 Jan '14

Signed-off-by: Daniel Matuschek <inof(a)crazy-audio.com> Existing post_table has duplicate settings for some PLL settings. Only the first setting was used with mclk_div=1 With the updated post_table, the driver uses mclk_div=0 whenever it is possible. This allows the WM8804 to work at 256xfs internal clock and only use 128xfs internal clock for high sample rates where 256xfs is not supported anymore. --- sound/soc/codecs/wm8804.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 1704b1e..f605b8b 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -307,9 +307,7 @@ static struct { unsigned int mclkdiv; } post_table[] = { { 2, 0, 0 }, - { 4, 0, 1 }, { 4, 1, 0 }, - { 8, 1, 1 }, { 8, 2, 0 }, { 16, 2, 1 }, { 12, 3, 0 }, -- 1.7.9.5

2 1

[alsa-devel] searching volunteer to test Hoontech DSP24 to check if the driver is realy working well
by YvanM 12 Jan '14

12 Jan '14

Hi everybody ! First please apologize if I'm in the wrong place. Secondly thanks for all the work you are all doing ! I have a Hoontech DSP24 (http://fr.audiofanzine.com/carte-son-rack/hoontech/DSP24-+-ADDA2000-DSP2000…), but I had problem to make it working. I don't use it so, if someone with knowledge wants to check if this soundcard is really recognized, I can send it to him. After that, feel free to play with it, improve the driver, or even sell it ! The problem I had : - spent 3 months to find on the Internet the following lines to had to /etc/modprobe.d/alsa-base to make the card recognized : stall snd-ice1712 /sbin/modprobe --ignore-install snd-ice1712 && { /sbin/modprobe -Qb snd-ice1712 ; } options snd-ice1712 model=dsp24 - impossible to have "normal" stereo : it seems that each logical canal (right and left) is redirected to both right and left speaker. I someone wants to have a try with it, do not hesitate to mail me. It is a pity that a so good soundcard can't be used anymore… And as I said, feel then free to do what you want with it. Thanks, Yvan

3 2

[alsa-devel] [PATCH] ALSA: Enable CONFIG_ZONE_DMA for smaller PCI DMA masks
by Takashi Iwai 11 Jan '14

11 Jan '14

The PCI devices with DMA masks smaller than 32bit should enable CONFIG_ZONE_DMA. Since the recent change of page allocator, page allocations via dma_alloc_coherent() with the limited DMA mask bits may fail more frequently, ended up with no available buffers, when CONFIG_ZONE_DMA isn't enabled. With CONFIG_ZONE_DMA, the system has much more chance to obtain such pages. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=68221 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/Kconfig | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 3a0504dfeb8a..8756c8e32922 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -25,6 +25,7 @@ config SND_ALS300 select SND_PCM select SND_AC97_CODEC select SND_OPL3_LIB + select ZONE_DMA help Say 'Y' or 'M' to include support for Avance Logic ALS300/ALS300+ @@ -49,6 +50,7 @@ config SND_ALI5451 tristate "ALi M5451 PCI Audio Controller" select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for the integrated AC97 sound device on motherboards using the ALi M5451 Audio Controller @@ -153,6 +155,7 @@ config SND_AZT3328 select SND_PCM select SND_RAWMIDI select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for Aztech AZF3328 (PCI168) soundcards. @@ -459,6 +462,7 @@ config SND_EMU10K1 select SND_HWDEP select SND_RAWMIDI select SND_AC97_CODEC + select ZONE_DMA help Say Y to include support for Sound Blaster PCI 512, Live!, Audigy and E-mu APS (partially supported) soundcards. @@ -474,6 +478,7 @@ config SND_EMU10K1X tristate "Emu10k1X (Dell OEM Version)" select SND_AC97_CODEC select SND_RAWMIDI + select ZONE_DMA help Say Y here to include support for the Dell OEM version of the Sound Blaster Live!. @@ -507,6 +512,7 @@ config SND_ES1938 select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on ESS Solo-1 (ES1938, ES1946, ES1969) chips. @@ -518,6 +524,7 @@ config SND_ES1968 tristate "ESS ES1968/1978 (Maestro-1/2/2E)" select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on ESS Maestro 1/2/2E chips. @@ -606,6 +613,7 @@ config SND_ICE1712 select SND_MPU401_UART select SND_AC97_CODEC select BITREVERSE + select ZONE_DMA help Say Y here to include support for soundcards based on the ICE1712 (Envy24) chip. @@ -693,6 +701,7 @@ config SND_LX6464ES config SND_MAESTRO3 tristate "ESS Allegro/Maestro3" select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on ESS Maestro 3 (Allegro) chips. @@ -789,6 +798,7 @@ config SND_SIS7019 tristate "SiS 7019 Audio Accelerator" depends on X86 && !X86_64 select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for the SiS 7019 Audio Accelerator. @@ -800,6 +810,7 @@ config SND_SONICVIBES select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on the S3 SonicVibes chip. @@ -811,6 +822,7 @@ config SND_TRIDENT tristate "Trident 4D-Wave DX/NX; SiS 7018" select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on Trident 4D-Wave DX/NX or SiS 7018 chips. -- 1.8.5.2

1 0

Re: [alsa-devel] [PATCH v2] ASoC: fsl_esai: Add ESAI CPU DAI driver
by Nicolin Chen 10 Jan '14

10 Jan '14

Resent this because of losing attached file. On Fri, Jan 10, 2014 at 10:32:52AM +0800, Nicolin Chen wrote: > On Thu, Jan 09, 2014 at 06:44:53PM +0000, Mark Brown wrote: > > On Thu, Jan 09, 2014 at 06:57:58PM +0800, Nicolin Chen wrote: > > > > > +/** > > > + * This function configures the ratio between MCLK (HCK) and BCLK (SCK) > > > + * (For DAI Master Mode only) > > > + * > > > + * Note: Machine driver should calculate the ratio to call this function. > > > + * Only effective after calling set_dai_sysclk() to set HCK direction. > > > + */ > > > +static int fsl_esai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) > > > > Why does the machine driver have to do this by hand, being able to > > override is fine but having sensible defaults is easier? Or does it > > actually do that and the comment just needs updating? > > The divider part of ESAI is pretty complicated due to caring about three > configure bits - ETO, ETI, HCKD. (I've attached a diagram to this mail.) > So setting sysclk() alone is not enough to take care all the configurations. > That's why I designed these two interfaces at the first place. And it should > be hard to find a default situation. > > But there is one approach to omit this calling for machine driver is to do > the set_bclk_ratio() at this CPU DAI driver. And this might be a good idea > because we can then separate the settings between PLAYBACK and CAPTURE, even > though we might then need to check the Master or Slave state to apply the > settings accordingly. > > I'll try this idea in v3 if you feel it's plausible. > > > > + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component, > > > + &fsl_esai_dai, 1); > > > + if (ret) { > > > + dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); > > > + return ret; > > > + } > > > + > > > + ret = imx_pcm_dma_init(pdev); > > > + if (ret) > > > + dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); > > > + > > > + /* Reset ESAI unit */ > > > + regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ERST); > > > + > > > + /* > > > + * We need to enable ESAI so as to access some of its registers. > > > + * Otherwise, we would fail to dump regmap from user space. > > > + */ > > > + regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ESAIEN); > > > > I would expect to see the hardware initialisation before we start > > registering with the core otherwise the core might start trying to run > > prior to the hardware being initialised. > > Will switch this initial code in v3 as your suggestion. > > Thank you, > Nicolin Chen

2 6

[alsa-devel] [PATCH 1/5] ASoC: pcm: Properly initialize hw->rate_max
by Lars-Peter Clausen 10 Jan '14

10 Jan '14

If none of the components (CODEC or CPU DAI) sets a maximum sample rate we'll end up with the rate_max field of the runtime hardware set to 0. (Note that it is still possible for the components to constrain the supported sample rates using other methods, e.g. setting a list constraint) If rate_max is 0 this means that the sound card doesn't support any rates at all, which is not the desired result. So initialize rate_max to UINT_MAX. For symmetry reasons also set rate_min to 0. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 313adba..b6e0c32 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -252,6 +252,9 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) hw->rates |= codec_stream->rates; + hw->rate_min = 0; + hw->rate_max = UINT_MAX; + snd_pcm_limit_hw_rates(runtime); hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); -- 1.8.0

2 6

[alsa-devel] [PATCH v2 28/28] drm/i2c: tda998x: add a CODEC interface
by Jean-Francois Moine 10 Jan '14

10 Jan '14

The TDA998x chips may get audio input from I2S or S/PDIF and this input is defined by the audio subsystem. In the other way, the audio subsystem may need to know if audio output may be done through the HDMI connector. This patch adds a CODEC interface for exchanges between the HDMI transmitter and the audio subsystem. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- drivers/gpu/drm/i2c/tda998x_drv.c | 70 ++++++++++++++++++++-- 1 file changed, 65 insertions(+), 5 deletions(-) diff --git a/drivers/gpu/drm/i2c/tda998x_drv.c b/drivers/gpu/drm/i2c/tda998x_drv.c index 67d7450..5a186e4 100644 --- a/drivers/gpu/drm/i2c/tda998x_drv.c +++ b/drivers/gpu/drm/i2c/tda998x_drv.c @@ -42,7 +42,8 @@ struct tda998x_priv { u8 vip_cntrl_1; u8 vip_cntrl_2; - u8 audio_type; /* audio type */ + u8 audio_type; /* audio type and 'active' flag */ +#define AUDIO_ACTIVE 0x80 u8 audio_frame[6]; u32 audio_port; @@ -50,6 +51,9 @@ struct tda998x_priv { wait_queue_head_t wq_edid; volatile int wq_edid_wait; struct drm_encoder *encoder; + + struct snd_soc_codec *codec; + void (*event)(struct snd_soc_codec *codec, int activate); }; #define to_tda998x_priv(x) ((struct tda998x_priv *)to_encoder_slave(x)->slave_priv) @@ -731,14 +735,14 @@ tda998x_configure_audio(struct tda998x_priv *priv, /* Set audio input source */ switch (priv->audio_type) { - case AFMT_SPDIF: + case AFMT_SPDIF | AUDIO_ACTIVE: reg_write(priv, REG_MUX_AP, MUX_AP_SELECT_SPDIF); clksel_aip = AIP_CLKSEL_AIP(SEL_AIP_SPDIF); clksel_fs = AIP_CLKSEL_FS(CTSREF_FS64SPDIF); cts_n = CTS_N_M(3) | CTS_N_K(3); break; - case AFMT_I2S: + case AFMT_I2S | AUDIO_ACTIVE: reg_write(priv, REG_MUX_AP, MUX_AP_SELECT_I2S); clksel_aip = AIP_CLKSEL_AIP(SEL_AIP_I2S); clksel_fs = AIP_CLKSEL_FS(CTSREF_ACLK); @@ -813,6 +817,58 @@ tda998x_configure_audio(struct tda998x_priv *priv, tda998x_write_aif(priv); } +/* tda998x codec interface */ +void tda998x_audio_register(struct i2c_client *client, + struct snd_soc_codec *codec, + void (*event)(struct snd_soc_codec *codec, int activate)) +{ + struct tda998x_priv *priv = i2c_get_clientdata(client); + + /* memorize the codec and the callback function */ + priv->codec = codec; + priv->event = event; +} +EXPORT_SYMBOL_GPL(tda998x_audio_register); + +void tda998x_audio_update(struct i2c_client *client, + int type, + u32 port) +{ + struct tda998x_priv *priv = i2c_get_clientdata(client); + + /* if the audio output is active, it may be a second start or a stop */ + if (type == 0 || (priv->audio_type & AUDIO_ACTIVE)) { + if (type == 0) { + priv->audio_type &= ~AUDIO_ACTIVE; /* stop */ + reg_write(priv, REG_ENA_AP, 0); + } + return; + } + + priv->audio_port = port; + + /* don't restart audio if same input type */ + if (type == priv->audio_type) { + priv->audio_type |= AUDIO_ACTIVE; + reg_write(priv, REG_ENA_AP, priv->audio_port); + return; + } + + priv->audio_type = type | AUDIO_ACTIVE; + + tda998x_configure_audio(priv, &priv->encoder->crtc->hwmode); +} +EXPORT_SYMBOL_GPL(tda998x_audio_update); + +/* send an audio event to the tda codec */ +static void send_audio_event(struct tda998x_priv *priv, + int event) +{ + if (!priv->event) + return; + priv->event(priv->codec, event); +} + /* DRM encoder functions */ /* this function is not called when no info->platform (DT support) */ @@ -840,7 +896,7 @@ tda998x_encoder_set_config(struct drm_encoder *encoder, void *params) if (p->audio_cfg) { priv->audio_port = p->audio_cfg; - priv->audio_type = p->audio_format; + priv->audio_type = p->audio_format | AUDIO_ACTIVE; } } @@ -1105,9 +1161,10 @@ tda998x_encoder_mode_set(struct drm_encoder *encoder, tda998x_write_avi(priv, mode); - if (priv->audio_type) + if (priv->audio_type & AUDIO_ACTIVE) tda998x_configure_audio(priv, mode); } + send_audio_event(priv, priv->is_hdmi_sink); /* must be last register set: */ reg_write(priv, REG_TBG_CNTRL_0, 0); @@ -1263,6 +1320,7 @@ tda998x_encoder_get_modes(struct drm_encoder *encoder, drm_mode_connector_update_edid_property(connector, edid); n = drm_add_edid_modes(connector, edid); priv->is_hdmi_sink = drm_detect_hdmi_monitor(edid); + send_audio_event(priv, priv->is_hdmi_sink); kfree(edid); } @@ -1349,6 +1407,8 @@ tda998x_encoder_init(struct i2c_client *client, if (!priv) return -ENOMEM; + i2c_set_clientdata(client, priv); + priv->vip_cntrl_0 = VIP_CNTRL_0_SWAP_A(2) | VIP_CNTRL_0_SWAP_B(3); priv->vip_cntrl_1 = VIP_CNTRL_1_SWAP_C(0) | VIP_CNTRL_1_SWAP_D(1); priv->vip_cntrl_2 = VIP_CNTRL_2_SWAP_E(4) | VIP_CNTRL_2_SWAP_F(5); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH 1/8] ASoC: alc5623: Use params_width() rather than memory format
by Mark Brown 10 Jan '14

10 Jan '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/alc5623.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 256c364193a5..d3036283482a 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -714,17 +714,17 @@ static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream, iface &= ~ALC5623_DAI_I2S_DL_MASK; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: iface |= ALC5623_DAI_I2S_DL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= ALC5623_DAI_I2S_DL_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= ALC5623_DAI_I2S_DL_24; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= ALC5623_DAI_I2S_DL_32; break; default: -- 1.8.5.2

4 12

[alsa-devel] [PATCH] ASoC: fsl: imx-wm8962: Grant hw_params() permission to reprogram FLL
by Nicolin Chen 10 Jan '14

10 Jan '14

From: Nicolin Chen <b42378(a)freescale.com> Previously, we couldn't use hw_params() to reclock FLL becuase there might be a race between two simmultaneous substreams so the FLL reprogramming would be changed and accordingly mulfunction. Also it wouldn't make sense for analogue bypass path feature of WM8962. So we adopted the DAPM way to control it. However, if we want to playback a different sample rate file, we need to wait for DAPM to change its bias_level and reconfigure FLL. After we introduced full symmetry protection in the soc-pcm, we don't need to worry about the race any more. And the instance by using hw_params() to reprogram FLL will allow us to support flexible use cases -- for example: 'aplay -Dhw:0 44k16bit.wav 48k24bit.wav 32k16bit.wav'. Thus this patch mainly adds FLL reprogramming code to hw_params() so as to enchance the sound card's capability. Meanwhile in order not to break the analogue bypass path feature, we make both set_bias_level() and hw_params() ways coexist by only reclocking FLL if requiring a different frequency and meanwhile not running any analogue bypass path in the background. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/imx-wm8962.c | 167 ++++++++++++++++++++++++++++++++------------- 1 file changed, 121 insertions(+), 46 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3fd76bc..9621909 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -17,6 +17,7 @@ #include <linux/of_platform.h> #include <linux/i2c.h> #include <linux/slab.h> +#include <linux/spinlock.h> #include <linux/clk.h> #include <sound/soc.h> #include <sound/pcm_params.h> @@ -35,6 +36,8 @@ struct imx_wm8962_data { char platform_name[DAI_NAME_SIZE]; struct clk *codec_clk; unsigned int clk_frequency; + unsigned int fll_frequency; + spinlock_t fll_lock; }; struct imx_priv { @@ -50,15 +53,125 @@ static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { }; static int sample_rate = 44100; +static unsigned int fll_frequency = 44100 * 256; static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; +static inline u32 imx_wm8962_calculate_fll(u32 rate, snd_pcm_format_t format) +{ + switch (format) { + case SNDRV_PCM_FORMAT_S24_LE: + return rate * 384; + default: + return rate * 256; + } +} + +static int imx_wm8962_enable_fll(struct snd_soc_dai *codec_dai) +{ + struct imx_priv *priv = &card_priv; + struct device *dev = &priv->pdev->dev; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + u32 ret = 0; + + spin_lock(&data->fll_lock); + + /* Protect FLL if it has not been disabled */ + if (data->fll_frequency) + goto err_unlock; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, + data->clk_frequency, fll_frequency); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + goto err_unlock; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, + fll_frequency, SND_SOC_CLOCK_IN); + if (ret) + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + + data->fll_frequency = fll_frequency; + +err_unlock: + spin_unlock(&data->fll_lock); + + return ret; +} + +static int imx_wm8962_disable_fll(struct snd_soc_dai *codec_dai) +{ + struct imx_priv *priv = &card_priv; + struct device *dev = &priv->pdev->dev; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + int ret = 0; + + spin_lock(&data->fll_lock); + + if (!data->fll_frequency) + goto err_unlock; + + /* Switch to MCLK as sysclk once so as to disable FLL */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + 0, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + goto err_unlock; + } + + /* Disable FLL so that we can reset its output freq later */ + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, 0, 0); + if (ret) + dev_err(dev, "failed to stop FLL: %d\n", ret); + + data->fll_frequency = 0; + +err_unlock: + spin_unlock(&data->fll_lock); + + return ret; +} + +static int imx_wm8962_reprogram_fll(struct snd_pcm_substream *substream) +{ + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + u32 mask, ret = 0; + + mask = WM8962_MIXINL_TO_HPMIXL_MASK | WM8962_MIXINR_TO_HPMIXL_MASK | + WM8962_IN4L_TO_HPMIXL_MASK | WM8962_IN4R_TO_HPMIXL_MASK; + ret |= snd_soc_read(codec, WM8962_HEADPHONE_MIXER_1) & mask; + ret |= snd_soc_read(codec, WM8962_HEADPHONE_MIXER_2) & mask; + ret |= snd_soc_read(codec, WM8962_SPEAKER_MIXER_1) & mask; + ret |= snd_soc_read(codec, WM8962_SPEAKER_MIXER_2) & mask; + + spin_lock(&data->fll_lock); + /* Do not reprogram FLL if not changed or running analogue bypass */ + if (fll_frequency == data->fll_frequency || ret) { + spin_unlock(&data->fll_lock); + return 0; + } + spin_unlock(&data->fll_lock); + + ret = imx_wm8962_disable_fll(codec_dai); + if (ret) + return ret; + + return imx_wm8962_enable_fll(codec_dai); +} + static int imx_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { sample_rate = params_rate(params); sample_format = params_format(params); + fll_frequency = imx_wm8962_calculate_fll(sample_rate, sample_format); - return 0; + return imx_wm8962_reprogram_fll(substream); } static struct snd_soc_ops imx_hifi_ops = { @@ -70,60 +183,19 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, enum snd_soc_bias_level level) { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - struct imx_priv *priv = &card_priv; - struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); - struct device *dev = &priv->pdev->dev; - unsigned int pll_out; - int ret; if (dapm->dev != codec_dai->dev) return 0; switch (level) { case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - if (sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = sample_rate * 384; - else - pll_out = sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, data->clk_frequency, - pll_out); - if (ret < 0) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_FLL, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - } + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + return imx_wm8962_enable_fll(codec_dai); break; case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_MCLK, data->clk_frequency, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, - "failed to switch away from FLL: %d\n", - ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - 0, 0, 0); - if (ret < 0) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - } + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) + return imx_wm8962_disable_fll(codec_dai); break; default: @@ -225,6 +297,9 @@ static int imx_wm8962_probe(struct platform_device *pdev) goto fail; } + data->fll_frequency = 0; + spin_lock_init(&data->fll_lock); + data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); -- 1.8.4

3 7

[alsa-devel] [PATCH v3] ASoC: fsl_esai: Add ESAI CPU DAI driver
by Nicolin Chen 10 Jan '14

10 Jan '14

This patch implements a device-tree-only CPU DAI driver for Freescale ESAI controller that supports: - 12 channels playback and 8 channels record. [ Some of the inner transmitters and receivers are sharing same group of pins. So the maxmium 12 output or 8 input channels are only valid if there is no pin conflict occurring to it. ] - Independent (asynchronous mode) or shared (synchronous mode) transmit and receive sections with separate or shared internal/external clocks and frame syncs, operating in Master or Slave mode. [ Current ALSA seems not to allow CPU DAI drivers to configure DAI format separately for PLAYBACK and CAPTURE. So this first version only supports the case that uses the same DAI format for both directions. ] - Various DAI formats: I2S, Left-Justified, Right-Justified, DSP-A and DSP-B. - Programmable word length (8, 16, 20 or 24bits) - Flexible selection between system clock or external oscillator as input clock source, programmable internal clock divider and frame sync generation. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v3: * Added monaural audio support by using Normal mode. And accordingly dropped the esai-network-mode property in DT -- We let driver handle it. * Separated divider settings for Tx and Rx so that we can drop set_clkdiv(). * Changed set_bclk_ratio() to set_bclk() and let the new one be called in CPU DAI driver to omit the calling in machine driver. * Moved the ESAI reset and starting before registering the driver. * Implemented full symmetry for synchronous mode. * Use correct maxburst size. v2: * Corrected multi-line comments. * Added missing spaces between number and operator. .../devicetree/bindings/sound/fsl,esai.txt | 50 ++ sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_esai.c | 815 +++++++++++++++++++++ sound/soc/fsl/fsl_esai.h | 354 +++++++++ 5 files changed, 1224 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,esai.txt create mode 100644 sound/soc/fsl/fsl_esai.c create mode 100644 sound/soc/fsl/fsl_esai.h diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt new file mode 100644 index 0000000..d7b99fa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt @@ -0,0 +1,50 @@ +Freescale Enhanced Serial Audio Interface (ESAI) Controller + +The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port +for serial communication with a variety of serial devices, including industry +standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and +other DSPs. It has up to six transmitters and four receivers. + +Required properties: + + - compatible : Compatible list, must contain "fsl,imx35-esai". + + - reg : Offset and length of the register set for the device. + + - interrupts : Contains the spdif interrupt. + + - dmas : Generic dma devicetree binding as described in + Documentation/devicetree/bindings/dma/dma.txt. + + - dma-names : Two dmas have to be defined, "tx" and "rx". + + - clocks: Contains an entry for each entry in clock-names. + + - clock-names : Includes the following entries: + "core" The core clock used to access registers + "extal" The esai baud clock for esai controller used to derive + HCK, SCK and FS. + "fsys" The system clock derived from ahb clock used to derive + HCK, SCK and FS. + + - fsl,fifo-depth: The number of elements in the transmit and receive FIFOs. + This number is the maximum allowed value for TFCR[TFWM] or RFCR[RFWM]. + + - fsl,esai-synchronous: This is a boolean property. If present, indicating + that ESAI would work in the synchronous mode, which means all the settings + for Receiving would be duplicated from Transmition related registers. + +Example: + +esai: esai@02024000 { + compatible = "fsl,imx35-esai"; + reg = <0x02024000 0x4000>; + interrupts = <0 51 0x04>; + clocks = <&clks 208>, <&clks 118>, <&clks 208>; + clock-names = "core", "extal", "fsys"; + dmas = <&sdma 23 21 0>, <&sdma 24 21 0>; + dma-names = "rx", "tx"; + fsl,fifo-depth = <128>; + fsl,esai-synchronous; + status = "disabled"; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 514c275..324988d 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -8,6 +8,9 @@ config SND_SOC_FSL_SSI config SND_SOC_FSL_SPDIF tristate +config SND_SOC_FSL_ESAI + tristate + config SND_SOC_FSL_UTILS tristate diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index aaccbee..b12ad4b 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -14,11 +14,13 @@ obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-objs := fsl_ssi.o snd-soc-fsl-spdif-objs := fsl_spdif.o +snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o +obj-$(CONFIG_SND_SOC_FSL_ESAI) += snd-soc-fsl-esai.o obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c new file mode 100644 index 0000000..d0c72ed --- /dev/null +++ b/sound/soc/fsl/fsl_esai.c @@ -0,0 +1,815 @@ +/* + * Freescale ESAI ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/dmaengine.h> +#include <linux/module.h> +#include <linux/of_irq.h> +#include <linux/of_platform.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> + +#include "fsl_esai.h" +#include "imx-pcm.h" + +#define FSL_ESAI_RATES SNDRV_PCM_RATE_8000_192000 +#define FSL_ESAI_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/** + * fsl_esai: ESAI private data + * + * @dma_params_rx: DMA parameters for receive channel + * @dma_params_tx: DMA parameters for transmit channel + * @pdev: platform device pointer + * @regmap: regmap handler + * @coreclk: clock source to access register + * @extalclk: esai clock source to derive HCK, SCK and FS + * @fsysclk: system clock source to derive HCK, SCK and FS + * @fifo_depth: depth of tx/rx FIFO + * @slot_width: width of each DAI slot + * @hck_rate: clock rate of desired HCKx clock + * @sck_div: if using PSR/PM dividers for SCKx clock + * @slave_mode: if fully using DAI slave mode + * @synchronous: if using tx/rx synchronous mode + * @name: driver name + */ +struct fsl_esai { + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; + struct platform_device *pdev; + struct regmap *regmap; + struct clk *coreclk; + struct clk *extalclk; + struct clk *fsysclk; + u32 fifo_depth; + u32 slot_width; + u32 hck_rate[2]; + bool sck_div[2]; + bool slave_mode; + bool synchronous; + char name[32]; +}; + +static irqreturn_t esai_isr(int irq, void *devid) +{ + struct fsl_esai *esai_priv = (struct fsl_esai *)devid; + struct platform_device *pdev = esai_priv->pdev; + u32 esr; + + regmap_read(esai_priv->regmap, REG_ESAI_ESR, &esr); + + if (esr & ESAI_ESR_TINIT_MASK) + dev_dbg(&pdev->dev, "isr: Transmition Initialized\n"); + + if (esr & ESAI_ESR_RFF_MASK) + dev_warn(&pdev->dev, "isr: Receiving overrun\n"); + + if (esr & ESAI_ESR_TFE_MASK) + dev_warn(&pdev->dev, "isr: Transmition underrun\n"); + + if (esr & ESAI_ESR_TLS_MASK) + dev_dbg(&pdev->dev, "isr: Just transmitted the last slot\n"); + + if (esr & ESAI_ESR_TDE_MASK) + dev_dbg(&pdev->dev, "isr: Transmition data exception\n"); + + if (esr & ESAI_ESR_TED_MASK) + dev_dbg(&pdev->dev, "isr: Transmitting even slots\n"); + + if (esr & ESAI_ESR_TD_MASK) + dev_dbg(&pdev->dev, "isr: Transmitting data\n"); + + if (esr & ESAI_ESR_RLS_MASK) + dev_dbg(&pdev->dev, "isr: Just received the last slot\n"); + + if (esr & ESAI_ESR_RDE_MASK) + dev_dbg(&pdev->dev, "isr: Receiving data exception\n"); + + if (esr & ESAI_ESR_RED_MASK) + dev_dbg(&pdev->dev, "isr: Receiving even slots\n"); + + if (esr & ESAI_ESR_RD_MASK) + dev_dbg(&pdev->dev, "isr: Receiving data\n"); + + return IRQ_HANDLED; +} + +/** + * This function is used to calculate the divisors of psr, pm, fp and it is + * supposed to be called in set_dai_sysclk() and set_bclk(). + * + * @ratio: desired overall ratio for the paticipating dividers + * @usefp: for HCK setting, there is no need to set fp divider + * @fp: bypass other dividers by setting fp directly if fp != 0 + * @tx: current setting is for playback or capture + */ +static int fsl_esai_divisor_cal(struct snd_soc_dai *dai, bool tx, u32 ratio, + bool usefp, u32 fp) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + u32 psr, pm = 999, maxfp, prod, sub, savesub, i, j; + + maxfp = usefp ? 16 : 1; + + if (usefp && fp) + goto out_fp; + + if (ratio > 2 * 8 * 256 * maxfp || ratio < 2) { + dev_err(dai->dev, "the ratio is out of range (2 ~ %d)\n", + 2 * 8 * 256 * maxfp); + return -EINVAL; + } else if (ratio % 2) { + dev_err(dai->dev, "the raio must be even if using upper divider\n"); + return -EINVAL; + } + + ratio /= 2; + + psr = ratio <= 256 * maxfp ? ESAI_xCCR_xPSR_BYPASS : ESAI_xCCR_xPSR_DIV8; + + /* Set the max fluctuation -- 0.1% of the max devisor */ + savesub = (psr ? 1 : 8) * 256 * maxfp / 1000; + + /* Find the best value for PM */ + for (i = 1; i <= 256; i++) { + for (j = 1; j <= maxfp; j++) { + /* PSR (1 or 8) * PM (1 ~ 256) * FP (1 ~ 16) */ + prod = (psr ? 1 : 8) * i * j; + + if (prod == ratio) + sub = 0; + else if (prod / ratio == 1) + sub = prod - ratio; + else if (ratio / prod == 1) + sub = ratio - prod; + else + continue; + + /* Calculate the fraction */ + sub = sub * 1000 / ratio; + if (sub < savesub) { + savesub = sub; + pm = i; + fp = j; + } + + /* We are lucky */ + if (savesub == 0) + goto out; + } + } + + if (pm == 999) { + dev_err(dai->dev, "failed to calculate proper divisors\n"); + return -EINVAL; + } + +out: + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx), + ESAI_xCCR_xPSR_MASK | ESAI_xCCR_xPM_MASK, + psr | ESAI_xCCR_xPM(pm)); + +out_fp: + /* Bypass fp if not being required */ + if (maxfp <= 1) + return 0; + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx), + ESAI_xCCR_xFP_MASK, ESAI_xCCR_xFP(fp)); + + return 0; +} + +/** + * This function mainly configures the clock frequency of MCLK (HCKT/HCKR) + * + * @Parameters: + * clk_id: The clock source of HCKT/HCKR + * (Input from outside; output from inside, FSYS or EXTAL) + * freq: The required clock rate of HCKT/HCKR + * dir: The clock direction of HCKT/HCKR + * + * Note: If the direction is input, we do not care about clk_id. + */ +static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + struct clk *clksrc = esai_priv->extalclk; + bool tx = clk_id <= ESAI_HCKT_EXTAL; + bool in = dir == SND_SOC_CLOCK_IN; + u32 ret, ratio, ecr = 0; + unsigned long clk_rate; + + /* sck_div can be only bypassed if ETO/ERO=0 and SNC_SOC_CLOCK_OUT */ + esai_priv->sck_div[tx] = true; + + /* Set the direction of HCKT/HCKR pins */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx), + ESAI_xCCR_xHCKD, in ? 0 : ESAI_xCCR_xHCKD); + + if (in) + goto out; + + switch (clk_id) { + case ESAI_HCKT_FSYS: + case ESAI_HCKR_FSYS: + clksrc = esai_priv->fsysclk; + break; + case ESAI_HCKT_EXTAL: + ecr |= ESAI_ECR_ETI; + case ESAI_HCKR_EXTAL: + ecr |= ESAI_ECR_ERI; + break; + default: + return -EINVAL; + } + + if (IS_ERR(clksrc)) { + dev_err(dai->dev, "no assigned %s clock\n", + clk_id % 2 ? "extal" : "fsys"); + return PTR_ERR(clksrc); + } + clk_rate = clk_get_rate(clksrc); + + ratio = clk_rate / freq; + if (ratio * freq > clk_rate) + ret = ratio * freq - clk_rate; + else if (ratio * freq < clk_rate) + ret = clk_rate - ratio * freq; + else + ret = 0; + + /* Block if clock source can not be divided into the required rate */ + if (ret != 0 && clk_rate / ret < 1000) { + dev_err(dai->dev, "failed to derive required HCK%c rate\n", + tx ? 'T' : 'R'); + return -EINVAL; + } + + if (ratio == 1) { + /* Bypass all the dividers if not being needed */ + ecr |= tx ? ESAI_ECR_ETO : ESAI_ECR_ERO; + goto out; + } + + ret = fsl_esai_divisor_cal(dai, tx, ratio, false, 0); + if (ret) + return ret; + + esai_priv->sck_div[tx] = false; + +out: + esai_priv->hck_rate[tx] = freq; + + regmap_update_bits(esai_priv->regmap, REG_ESAI_ECR, + tx ? ESAI_ECR_ETI | ESAI_ECR_ETO : + ESAI_ECR_ERI | ESAI_ECR_ERO, ecr); + + return 0; +} + +/** + * This function configures the related dividers according to the bclk rate + */ +static int fsl_esai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + u32 hck_rate = esai_priv->hck_rate[tx]; + u32 sub, ratio = hck_rate / freq; + + /* Don't apply for fully slave mode*/ + if (esai_priv->slave_mode) + return 0; + + if (ratio * freq > hck_rate) + sub = ratio * freq - hck_rate; + else if (ratio * freq < hck_rate) + sub = hck_rate - ratio * freq; + else + sub = 0; + + /* Block if clock source can not be divided into the required rate */ + if (sub != 0 && hck_rate / sub < 1000) { + dev_err(dai->dev, "failed to derive required SCK%c rate\n", + tx ? 'T' : 'R'); + return -EINVAL; + } + + if (esai_priv->sck_div[tx] && (ratio > 16 || ratio == 0)) { + dev_err(dai->dev, "the ratio is out of range (1 ~ 16)\n"); + return -EINVAL; + } + + return fsl_esai_divisor_cal(dai, tx, ratio, true, + esai_priv->sck_div[tx] ? 0 : ratio); +} + +static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, + u32 rx_mask, int slots, int slot_width) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, + ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, + ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(tx_mask)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, + ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, + ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(rx_mask)); + + esai_priv->slot_width = slot_width; + + return 0; +} + +static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + u32 xcr = 0, xccr = 0, mask; + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* Data on rising edge of bclk, frame low, 1clk before data */ + xcr |= ESAI_xCR_xFSR; + xccr |= ESAI_xCCR_xFSP | ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* Data on rising edge of bclk, frame high */ + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + /* Data on rising edge of bclk, frame high, right aligned */ + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Data on rising edge of bclk, frame high, 1clk before data */ + xcr |= ESAI_xCR_xFSL | ESAI_xCR_xFSR; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_DSP_B: + /* Data on rising edge of bclk, frame high */ + xcr |= ESAI_xCR_xFSL; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do for both normal cases */ + break; + case SND_SOC_DAIFMT_IB_NF: + /* Invert bit clock */ + xccr ^= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_NB_IF: + /* Invert frame clock */ + xccr ^= ESAI_xCCR_xFSP; + break; + case SND_SOC_DAIFMT_IB_IF: + /* Invert both clocks */ + xccr ^= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP; + break; + default: + return -EINVAL; + } + + esai_priv->slave_mode = false; + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + esai_priv->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBS_CFM: + xccr |= ESAI_xCCR_xCKD; + break; + case SND_SOC_DAIFMT_CBM_CFS: + xccr |= ESAI_xCCR_xFSD; + break; + case SND_SOC_DAIFMT_CBS_CFS: + xccr |= ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; + break; + default: + return -EINVAL; + } + + mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR; + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr); + + mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP | + ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA; + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr); + + return 0; +} + +static int fsl_esai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + /* + * Some platforms might use the same bit to gate all three or two of + * clocks, so keep all clocks open/close at the same time for safety + */ + clk_prepare_enable(esai_priv->coreclk); + if (!IS_ERR(esai_priv->extalclk)) + clk_prepare_enable(esai_priv->extalclk); + if (!IS_ERR(esai_priv->fsysclk)) + clk_prepare_enable(esai_priv->fsysclk); + + if (!dai->active) { + /* Reset Port C */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC, + ESAI_PRRC_PDC_MASK, ESAI_PRRC_PDC(ESAI_GPIO)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC, + ESAI_PCRC_PC_MASK, ESAI_PCRC_PC(ESAI_GPIO)); + + /* Set synchronous mode */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_SAICR, + ESAI_SAICR_SYNC, esai_priv->synchronous ? + ESAI_SAICR_SYNC : 0); + + /* Set a default slot number -- 2 */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + } + + return 0; +} + +static int fsl_esai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 width = snd_pcm_format_width(params_format(params)); + u32 channels = params_channels(params); + u32 bclk, mask, val, ret; + + bclk = params_rate(params) * esai_priv->slot_width * 2; + + ret = fsl_esai_set_bclk(dai, tx, bclk); + if (ret) + return ret; + + /* Use Normal mode to support monaural audio */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + ESAI_xCR_xMOD_MASK, params_channels(params) > 1 ? + ESAI_xCR_xMOD_NETWORK : 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFR_MASK, ESAI_xFCR_xFR); + + mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK | + (tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK); + val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth) | + (tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val); + + mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0); + val = ESAI_xCR_xSWS(esai_priv->slot_width, width) | (tx ? ESAI_xCR_PADC : 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val); + + return 0; +} + +static void fsl_esai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + if (!IS_ERR(esai_priv->fsysclk)) + clk_disable_unprepare(esai_priv->fsysclk); + if (!IS_ERR(esai_priv->extalclk)) + clk_disable_unprepare(esai_priv->extalclk); + clk_disable_unprepare(esai_priv->coreclk); +} + +static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u8 i, channels = substream->runtime->channels; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFEN_MASK, ESAI_xFCR_xFEN); + + /* Write initial words reqiured by ESAI as normal procedure */ + for (i = 0; tx && i < channels; i++) + regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, + tx ? ESAI_xCR_TE(channels) : ESAI_xCR_RE(channels)); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0); + + /* Disable and reset FIFO */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFR | ESAI_xFCR_xFEN, ESAI_xFCR_xFR); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFR, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops fsl_esai_dai_ops = { + .startup = fsl_esai_startup, + .shutdown = fsl_esai_shutdown, + .trigger = fsl_esai_trigger, + .hw_params = fsl_esai_hw_params, + .set_sysclk = fsl_esai_set_dai_sysclk, + .set_fmt = fsl_esai_set_dai_fmt, + .set_tdm_slot = fsl_esai_set_dai_tdm_slot, +}; + +static int fsl_esai_dai_probe(struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &esai_priv->dma_params_tx, + &esai_priv->dma_params_rx); + + return 0; +} + +static struct snd_soc_dai_driver fsl_esai_dai = { + .probe = fsl_esai_dai_probe, + .playback = { + .channels_min = 1, + .channels_max = 12, + .rates = FSL_ESAI_RATES, + .formats = FSL_ESAI_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = FSL_ESAI_RATES, + .formats = FSL_ESAI_FORMATS, + }, + .ops = &fsl_esai_dai_ops, +}; + +static const struct snd_soc_component_driver fsl_esai_component = { + .name = "fsl-esai", +}; + +static bool fsl_esai_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ESAI_ERDR: + case REG_ESAI_ECR: + case REG_ESAI_ESR: + case REG_ESAI_TFCR: + case REG_ESAI_TFSR: + case REG_ESAI_RFCR: + case REG_ESAI_RFSR: + case REG_ESAI_RX0: + case REG_ESAI_RX1: + case REG_ESAI_RX2: + case REG_ESAI_RX3: + case REG_ESAI_SAISR: + case REG_ESAI_SAICR: + case REG_ESAI_TCR: + case REG_ESAI_TCCR: + case REG_ESAI_RCR: + case REG_ESAI_RCCR: + case REG_ESAI_TSMA: + case REG_ESAI_TSMB: + case REG_ESAI_RSMA: + case REG_ESAI_RSMB: + case REG_ESAI_PRRC: + case REG_ESAI_PCRC: + return true; + default: + return false; + } +} + +static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ESAI_ETDR: + case REG_ESAI_ECR: + case REG_ESAI_TFCR: + case REG_ESAI_RFCR: + case REG_ESAI_TX0: + case REG_ESAI_TX1: + case REG_ESAI_TX2: + case REG_ESAI_TX3: + case REG_ESAI_TX4: + case REG_ESAI_TX5: + case REG_ESAI_TSR: + case REG_ESAI_SAICR: + case REG_ESAI_TCR: + case REG_ESAI_TCCR: + case REG_ESAI_RCR: + case REG_ESAI_RCCR: + case REG_ESAI_TSMA: + case REG_ESAI_TSMB: + case REG_ESAI_RSMA: + case REG_ESAI_RSMB: + case REG_ESAI_PRRC: + case REG_ESAI_PCRC: + return true; + default: + return false; + } +} + +static const struct regmap_config fsl_esai_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + + .max_register = REG_ESAI_PCRC, + .readable_reg = fsl_esai_readable_reg, + .writeable_reg = fsl_esai_writeable_reg, +}; + +static int fsl_esai_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct fsl_esai *esai_priv; + struct resource *res; + const uint32_t *iprop; + void __iomem *regs; + int irq, ret; + + esai_priv = devm_kzalloc(&pdev->dev, sizeof(*esai_priv), GFP_KERNEL); + if (!esai_priv) + return -ENOMEM; + + esai_priv->pdev = pdev; + strcpy(esai_priv->name, np->name); + + /* Get the addresses and IRQ */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + esai_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "core", regs, &fsl_esai_regmap_config); + if (IS_ERR(esai_priv->regmap)) { + dev_err(&pdev->dev, "failed to init regmap: %ld\n", + PTR_ERR(esai_priv->regmap)); + return PTR_ERR(esai_priv->regmap); + } + + esai_priv->coreclk = devm_clk_get(&pdev->dev, "core"); + if (IS_ERR(esai_priv->coreclk)) { + dev_err(&pdev->dev, "failed to get core clock: %ld\n", + PTR_ERR(esai_priv->coreclk)); + return PTR_ERR(esai_priv->coreclk); + } + + esai_priv->extalclk = devm_clk_get(&pdev->dev, "extal"); + if (IS_ERR(esai_priv->extalclk)) + dev_warn(&pdev->dev, "failed to get extal clock: %ld\n", + PTR_ERR(esai_priv->extalclk)); + + esai_priv->fsysclk = devm_clk_get(&pdev->dev, "fsys"); + if (IS_ERR(esai_priv->fsysclk)) + dev_warn(&pdev->dev, "failed to get fsys clock: %ld\n", + PTR_ERR(esai_priv->fsysclk)); + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + return irq; + } + + ret = devm_request_irq(&pdev->dev, irq, esai_isr, 0, + esai_priv->name, esai_priv); + if (ret) { + dev_err(&pdev->dev, "failed to claim irq %u\n", irq); + return ret; + } + + /* Set a default slot size */ + esai_priv->slot_width = 32; + + /* Set a default master/slave state */ + esai_priv->slave_mode = true; + + /* Determine the FIFO depth */ + iprop = of_get_property(np, "fsl,fifo-depth", NULL); + if (iprop) + esai_priv->fifo_depth = be32_to_cpup(iprop); + else + esai_priv->fifo_depth = 64; + + esai_priv->dma_params_tx.maxburst = 16; + esai_priv->dma_params_rx.maxburst = 16; + esai_priv->dma_params_tx.addr = res->start + REG_ESAI_ETDR; + esai_priv->dma_params_rx.addr = res->start + REG_ESAI_ERDR; + + esai_priv->synchronous = + of_property_read_bool(np, "fsl,esai-synchronous"); + + /* Implement full symmetry for synchronous mode */ + if (esai_priv->synchronous) { + fsl_esai_dai.symmetric_rates = 1; + fsl_esai_dai.symmetric_channels = 1; + fsl_esai_dai.symmetric_samplebits = 1; + } + + dev_set_drvdata(&pdev->dev, esai_priv); + + /* Reset ESAI unit */ + ret = regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ERST); + if (ret) { + dev_err(&pdev->dev, "failed to reset ESAI: %d\n", ret); + return ret; + } + + /* + * We need to enable ESAI so as to access some of its registers. + * Otherwise, we would fail to dump regmap from user space. + */ + ret = regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ESAIEN); + if (ret) { + dev_err(&pdev->dev, "failed to enable ESAI: %d\n", ret); + return ret; + } + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component, + &fsl_esai_dai, 1); + if (ret) { + dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); + return ret; + } + + ret = imx_pcm_dma_init(pdev); + if (ret) + dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); + + return ret; +} + +static const struct of_device_id fsl_esai_dt_ids[] = { + { .compatible = "fsl,imx35-esai", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids); + +static struct platform_driver fsl_esai_driver = { + .probe = fsl_esai_probe, + .driver = { + .name = "fsl-esai-dai", + .owner = THIS_MODULE, + .of_match_table = fsl_esai_dt_ids, + }, +}; + +module_platform_driver(fsl_esai_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale ESAI CPU DAI driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:fsl-esai-dai"); diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h new file mode 100644 index 0000000..9c9f957 --- /dev/null +++ b/sound/soc/fsl/fsl_esai.h @@ -0,0 +1,354 @@ +/* + * fsl_esai.h - ALSA ESAI interface for the Freescale i.MX SoC + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _FSL_ESAI_DAI_H +#define _FSL_ESAI_DAI_H + +/* ESAI Register Map */ +#define REG_ESAI_ETDR 0x00 +#define REG_ESAI_ERDR 0x04 +#define REG_ESAI_ECR 0x08 +#define REG_ESAI_ESR 0x0C +#define REG_ESAI_TFCR 0x10 +#define REG_ESAI_TFSR 0x14 +#define REG_ESAI_RFCR 0x18 +#define REG_ESAI_RFSR 0x1C +#define REG_ESAI_xFCR(tx) (tx ? REG_ESAI_TFCR : REG_ESAI_RFCR) +#define REG_ESAI_xFSR(tx) (tx ? REG_ESAI_TFSR : REG_ESAI_RFSR) +#define REG_ESAI_TX0 0x80 +#define REG_ESAI_TX1 0x84 +#define REG_ESAI_TX2 0x88 +#define REG_ESAI_TX3 0x8C +#define REG_ESAI_TX4 0x90 +#define REG_ESAI_TX5 0x94 +#define REG_ESAI_TSR 0x98 +#define REG_ESAI_RX0 0xA0 +#define REG_ESAI_RX1 0xA4 +#define REG_ESAI_RX2 0xA8 +#define REG_ESAI_RX3 0xAC +#define REG_ESAI_SAISR 0xCC +#define REG_ESAI_SAICR 0xD0 +#define REG_ESAI_TCR 0xD4 +#define REG_ESAI_TCCR 0xD8 +#define REG_ESAI_RCR 0xDC +#define REG_ESAI_RCCR 0xE0 +#define REG_ESAI_xCR(tx) (tx ? REG_ESAI_TCR : REG_ESAI_RCR) +#define REG_ESAI_xCCR(tx) (tx ? REG_ESAI_TCCR : REG_ESAI_RCCR) +#define REG_ESAI_TSMA 0xE4 +#define REG_ESAI_TSMB 0xE8 +#define REG_ESAI_RSMA 0xEC +#define REG_ESAI_RSMB 0xF0 +#define REG_ESAI_xSMA(tx) (tx ? REG_ESAI_TSMA : REG_ESAI_RSMA) +#define REG_ESAI_xSMB(tx) (tx ? REG_ESAI_TSMB : REG_ESAI_RSMB) +#define REG_ESAI_PRRC 0xF8 +#define REG_ESAI_PCRC 0xFC + +/* ESAI Control Register -- REG_ESAI_ECR 0x8 */ +#define ESAI_ECR_ETI_SHIFT 19 +#define ESAI_ECR_ETI_MASK (1 << ESAI_ECR_ETI_SHIFT) +#define ESAI_ECR_ETI (1 << ESAI_ECR_ETI_SHIFT) +#define ESAI_ECR_ETO_SHIFT 18 +#define ESAI_ECR_ETO_MASK (1 << ESAI_ECR_ETO_SHIFT) +#define ESAI_ECR_ETO (1 << ESAI_ECR_ETO_SHIFT) +#define ESAI_ECR_ERI_SHIFT 17 +#define ESAI_ECR_ERI_MASK (1 << ESAI_ECR_ERI_SHIFT) +#define ESAI_ECR_ERI (1 << ESAI_ECR_ERI_SHIFT) +#define ESAI_ECR_ERO_SHIFT 16 +#define ESAI_ECR_ERO_MASK (1 << ESAI_ECR_ERO_SHIFT) +#define ESAI_ECR_ERO (1 << ESAI_ECR_ERO_SHIFT) +#define ESAI_ECR_ERST_SHIFT 1 +#define ESAI_ECR_ERST_MASK (1 << ESAI_ECR_ERST_SHIFT) +#define ESAI_ECR_ERST (1 << ESAI_ECR_ERST_SHIFT) +#define ESAI_ECR_ESAIEN_SHIFT 0 +#define ESAI_ECR_ESAIEN_MASK (1 << ESAI_ECR_ESAIEN_SHIFT) +#define ESAI_ECR_ESAIEN (1 << ESAI_ECR_ESAIEN_SHIFT) + +/* ESAI Status Register -- REG_ESAI_ESR 0xC */ +#define ESAI_ESR_TINIT_SHIFT 10 +#define ESAI_ESR_TINIT_MASK (1 << ESAI_ESR_TINIT_SHIFT) +#define ESAI_ESR_TINIT (1 << ESAI_ESR_TINIT_SHIFT) +#define ESAI_ESR_RFF_SHIFT 9 +#define ESAI_ESR_RFF_MASK (1 << ESAI_ESR_RFF_SHIFT) +#define ESAI_ESR_RFF (1 << ESAI_ESR_RFF_SHIFT) +#define ESAI_ESR_TFE_SHIFT 8 +#define ESAI_ESR_TFE_MASK (1 << ESAI_ESR_TFE_SHIFT) +#define ESAI_ESR_TFE (1 << ESAI_ESR_TFE_SHIFT) +#define ESAI_ESR_TLS_SHIFT 7 +#define ESAI_ESR_TLS_MASK (1 << ESAI_ESR_TLS_SHIFT) +#define ESAI_ESR_TLS (1 << ESAI_ESR_TLS_SHIFT) +#define ESAI_ESR_TDE_SHIFT 6 +#define ESAI_ESR_TDE_MASK (1 << ESAI_ESR_TDE_SHIFT) +#define ESAI_ESR_TDE (1 << ESAI_ESR_TDE_SHIFT) +#define ESAI_ESR_TED_SHIFT 5 +#define ESAI_ESR_TED_MASK (1 << ESAI_ESR_TED_SHIFT) +#define ESAI_ESR_TED (1 << ESAI_ESR_TED_SHIFT) +#define ESAI_ESR_TD_SHIFT 4 +#define ESAI_ESR_TD_MASK (1 << ESAI_ESR_TD_SHIFT) +#define ESAI_ESR_TD (1 << ESAI_ESR_TD_SHIFT) +#define ESAI_ESR_RLS_SHIFT 3 +#define ESAI_ESR_RLS_MASK (1 << ESAI_ESR_RLS_SHIFT) +#define ESAI_ESR_RLS (1 << ESAI_ESR_RLS_SHIFT) +#define ESAI_ESR_RDE_SHIFT 2 +#define ESAI_ESR_RDE_MASK (1 << ESAI_ESR_RDE_SHIFT) +#define ESAI_ESR_RDE (1 << ESAI_ESR_RDE_SHIFT) +#define ESAI_ESR_RED_SHIFT 1 +#define ESAI_ESR_RED_MASK (1 << ESAI_ESR_RED_SHIFT) +#define ESAI_ESR_RED (1 << ESAI_ESR_RED_SHIFT) +#define ESAI_ESR_RD_SHIFT 0 +#define ESAI_ESR_RD_MASK (1 << ESAI_ESR_RD_SHIFT) +#define ESAI_ESR_RD (1 << ESAI_ESR_RD_SHIFT) + +/* + * Transmit FIFO Configuration Register -- REG_ESAI_TFCR 0x10 + * Receive FIFO Configuration Register -- REG_ESAI_RFCR 0x18 + */ +#define ESAI_xFCR_TIEN_SHIFT 19 +#define ESAI_xFCR_TIEN_MASK (1 << ESAI_xFCR_TIEN_SHIFT) +#define ESAI_xFCR_TIEN (1 << ESAI_xFCR_TIEN_SHIFT) +#define ESAI_xFCR_REXT_SHIFT 19 +#define ESAI_xFCR_REXT_MASK (1 << ESAI_xFCR_REXT_SHIFT) +#define ESAI_xFCR_REXT (1 << ESAI_xFCR_REXT_SHIFT) +#define ESAI_xFCR_xWA_SHIFT 16 +#define ESAI_xFCR_xWA_WIDTH 3 +#define ESAI_xFCR_xWA_MASK (((1 << ESAI_xFCR_xWA_WIDTH) - 1) << ESAI_xFCR_xWA_SHIFT) +#define ESAI_xFCR_xWA(v) (((8 - ((v) >> 2)) << ESAI_xFCR_xWA_SHIFT) & ESAI_xFCR_xWA_MASK) +#define ESAI_xFCR_xFWM_SHIFT 8 +#define ESAI_xFCR_xFWM_WIDTH 8 +#define ESAI_xFCR_xFWM_MASK (((1 << ESAI_xFCR_xFWM_WIDTH) - 1) << ESAI_xFCR_xFWM_SHIFT) +#define ESAI_xFCR_xFWM(v) ((((v) - 1) << ESAI_xFCR_xFWM_SHIFT) & ESAI_xFCR_xFWM_MASK) +#define ESAI_xFCR_xE_SHIFT 2 +#define ESAI_xFCR_TE_WIDTH 6 +#define ESAI_xFCR_RE_WIDTH 4 +#define ESAI_xFCR_TE_MASK (((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) +#define ESAI_xFCR_RE_MASK (((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) +#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_TE_MASK) +#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_RE_MASK) +#define ESAI_xFCR_xFR_SHIFT 1 +#define ESAI_xFCR_xFR_MASK (1 << ESAI_xFCR_xFR_SHIFT) +#define ESAI_xFCR_xFR (1 << ESAI_xFCR_xFR_SHIFT) +#define ESAI_xFCR_xFEN_SHIFT 0 +#define ESAI_xFCR_xFEN_MASK (1 << ESAI_xFCR_xFEN_SHIFT) +#define ESAI_xFCR_xFEN (1 << ESAI_xFCR_xFEN_SHIFT) + +/* + * Transmit FIFO Status Register -- REG_ESAI_TFSR 0x14 + * Receive FIFO Status Register --REG_ESAI_RFSR 0x1C + */ +#define ESAI_xFSR_NTFO_SHIFT 12 +#define ESAI_xFSR_NRFI_SHIFT 12 +#define ESAI_xFSR_NTFI_SHIFT 8 +#define ESAI_xFSR_NRFO_SHIFT 8 +#define ESAI_xFSR_NTFx_WIDTH 3 +#define ESAI_xFSR_NRFx_WIDTH 2 +#define ESAI_xFSR_NTFO_MASK (((1 << ESAI_xFSR_NTFx_WIDTH) - 1) << ESAI_xFSR_NTFO_SHIFT) +#define ESAI_xFSR_NTFI_MASK (((1 << ESAI_xFSR_NTFx_WIDTH) - 1) << ESAI_xFSR_NTFI_SHIFT) +#define ESAI_xFSR_NRFO_MASK (((1 << ESAI_xFSR_NRFx_WIDTH) - 1) << ESAI_xFSR_NRFO_SHIFT) +#define ESAI_xFSR_NRFI_MASK (((1 << ESAI_xFSR_NRFx_WIDTH) - 1) << ESAI_xFSR_NRFI_SHIFT) +#define ESAI_xFSR_xFCNT_SHIFT 0 +#define ESAI_xFSR_xFCNT_WIDTH 8 +#define ESAI_xFSR_xFCNT_MASK (((1 << ESAI_xFSR_xFCNT_WIDTH) - 1) << ESAI_xFSR_xFCNT_SHIFT) + +/* ESAI Transmit Slot Register -- REG_ESAI_TSR 0x98 */ +#define ESAI_TSR_SHIFT 0 +#define ESAI_TSR_WIDTH 24 +#define ESAI_TSR_MASK (((1 << ESAI_TSR_WIDTH) - 1) << ESAI_TSR_SHIFT) + +/* Serial Audio Interface Status Register -- REG_ESAI_SAISR 0xCC */ +#define ESAI_SAISR_TODFE_SHIFT 17 +#define ESAI_SAISR_TODFE_MASK (1 << ESAI_SAISR_TODFE_SHIFT) +#define ESAI_SAISR_TODFE (1 << ESAI_SAISR_TODFE_SHIFT) +#define ESAI_SAISR_TEDE_SHIFT 16 +#define ESAI_SAISR_TEDE_MASK (1 << ESAI_SAISR_TEDE_SHIFT) +#define ESAI_SAISR_TEDE (1 << ESAI_SAISR_TEDE_SHIFT) +#define ESAI_SAISR_TDE_SHIFT 15 +#define ESAI_SAISR_TDE_MASK (1 << ESAI_SAISR_TDE_SHIFT) +#define ESAI_SAISR_TDE (1 << ESAI_SAISR_TDE_SHIFT) +#define ESAI_SAISR_TUE_SHIFT 14 +#define ESAI_SAISR_TUE_MASK (1 << ESAI_SAISR_TUE_SHIFT) +#define ESAI_SAISR_TUE (1 << ESAI_SAISR_TUE_SHIFT) +#define ESAI_SAISR_TFS_SHIFT 13 +#define ESAI_SAISR_TFS_MASK (1 << ESAI_SAISR_TFS_SHIFT) +#define ESAI_SAISR_TFS (1 << ESAI_SAISR_TFS_SHIFT) +#define ESAI_SAISR_RODF_SHIFT 10 +#define ESAI_SAISR_RODF_MASK (1 << ESAI_SAISR_RODF_SHIFT) +#define ESAI_SAISR_RODF (1 << ESAI_SAISR_RODF_SHIFT) +#define ESAI_SAISR_REDF_SHIFT 9 +#define ESAI_SAISR_REDF_MASK (1 << ESAI_SAISR_REDF_SHIFT) +#define ESAI_SAISR_REDF (1 << ESAI_SAISR_REDF_SHIFT) +#define ESAI_SAISR_RDF_SHIFT 8 +#define ESAI_SAISR_RDF_MASK (1 << ESAI_SAISR_RDF_SHIFT) +#define ESAI_SAISR_RDF (1 << ESAI_SAISR_RDF_SHIFT) +#define ESAI_SAISR_ROE_SHIFT 7 +#define ESAI_SAISR_ROE_MASK (1 << ESAI_SAISR_ROE_SHIFT) +#define ESAI_SAISR_ROE (1 << ESAI_SAISR_ROE_SHIFT) +#define ESAI_SAISR_RFS_SHIFT 6 +#define ESAI_SAISR_RFS_MASK (1 << ESAI_SAISR_RFS_SHIFT) +#define ESAI_SAISR_RFS (1 << ESAI_SAISR_RFS_SHIFT) +#define ESAI_SAISR_IF2_SHIFT 2 +#define ESAI_SAISR_IF2_MASK (1 << ESAI_SAISR_IF2_SHIFT) +#define ESAI_SAISR_IF2 (1 << ESAI_SAISR_IF2_SHIFT) +#define ESAI_SAISR_IF1_SHIFT 1 +#define ESAI_SAISR_IF1_MASK (1 << ESAI_SAISR_IF1_SHIFT) +#define ESAI_SAISR_IF1 (1 << ESAI_SAISR_IF1_SHIFT) +#define ESAI_SAISR_IF0_SHIFT 0 +#define ESAI_SAISR_IF0_MASK (1 << ESAI_SAISR_IF0_SHIFT) +#define ESAI_SAISR_IF0 (1 << ESAI_SAISR_IF0_SHIFT) + +/* Serial Audio Interface Control Register -- REG_ESAI_SAICR 0xD0 */ +#define ESAI_SAICR_ALC_SHIFT 8 +#define ESAI_SAICR_ALC_MASK (1 << ESAI_SAICR_ALC_SHIFT) +#define ESAI_SAICR_ALC (1 << ESAI_SAICR_ALC_SHIFT) +#define ESAI_SAICR_TEBE_SHIFT 7 +#define ESAI_SAICR_TEBE_MASK (1 << ESAI_SAICR_TEBE_SHIFT) +#define ESAI_SAICR_TEBE (1 << ESAI_SAICR_TEBE_SHIFT) +#define ESAI_SAICR_SYNC_SHIFT 6 +#define ESAI_SAICR_SYNC_MASK (1 << ESAI_SAICR_SYNC_SHIFT) +#define ESAI_SAICR_SYNC (1 << ESAI_SAICR_SYNC_SHIFT) +#define ESAI_SAICR_OF2_SHIFT 2 +#define ESAI_SAICR_OF2_MASK (1 << ESAI_SAICR_OF2_SHIFT) +#define ESAI_SAICR_OF2 (1 << ESAI_SAICR_OF2_SHIFT) +#define ESAI_SAICR_OF1_SHIFT 1 +#define ESAI_SAICR_OF1_MASK (1 << ESAI_SAICR_OF1_SHIFT) +#define ESAI_SAICR_OF1 (1 << ESAI_SAICR_OF1_SHIFT) +#define ESAI_SAICR_OF0_SHIFT 0 +#define ESAI_SAICR_OF0_MASK (1 << ESAI_SAICR_OF0_SHIFT) +#define ESAI_SAICR_OF0 (1 << ESAI_SAICR_OF0_SHIFT) + +/* + * Transmit Control Register -- REG_ESAI_TCR 0xD4 + * Receive Control Register -- REG_ESAI_RCR 0xDC + */ +#define ESAI_xCR_xLIE_SHIFT 23 +#define ESAI_xCR_xLIE_MASK (1 << ESAI_xCR_xLIE_SHIFT) +#define ESAI_xCR_xLIE (1 << ESAI_xCR_xLIE_SHIFT) +#define ESAI_xCR_xIE_SHIFT 22 +#define ESAI_xCR_xIE_MASK (1 << ESAI_xCR_xIE_SHIFT) +#define ESAI_xCR_xIE (1 << ESAI_xCR_xIE_SHIFT) +#define ESAI_xCR_xEDIE_SHIFT 21 +#define ESAI_xCR_xEDIE_MASK (1 << ESAI_xCR_xEDIE_SHIFT) +#define ESAI_xCR_xEDIE (1 << ESAI_xCR_xEDIE_SHIFT) +#define ESAI_xCR_xEIE_SHIFT 20 +#define ESAI_xCR_xEIE_MASK (1 << ESAI_xCR_xEIE_SHIFT) +#define ESAI_xCR_xEIE (1 << ESAI_xCR_xEIE_SHIFT) +#define ESAI_xCR_xPR_SHIFT 19 +#define ESAI_xCR_xPR_MASK (1 << ESAI_xCR_xPR_SHIFT) +#define ESAI_xCR_xPR (1 << ESAI_xCR_xPR_SHIFT) +#define ESAI_xCR_PADC_SHIFT 17 +#define ESAI_xCR_PADC_MASK (1 << ESAI_xCR_PADC_SHIFT) +#define ESAI_xCR_PADC (1 << ESAI_xCR_PADC_SHIFT) +#define ESAI_xCR_xFSR_SHIFT 16 +#define ESAI_xCR_xFSR_MASK (1 << ESAI_xCR_xFSR_SHIFT) +#define ESAI_xCR_xFSR (1 << ESAI_xCR_xFSR_SHIFT) +#define ESAI_xCR_xFSL_SHIFT 15 +#define ESAI_xCR_xFSL_MASK (1 << ESAI_xCR_xFSL_SHIFT) +#define ESAI_xCR_xFSL (1 << ESAI_xCR_xFSL_SHIFT) +#define ESAI_xCR_xSWS_SHIFT 10 +#define ESAI_xCR_xSWS_WIDTH 5 +#define ESAI_xCR_xSWS_MASK (((1 << ESAI_xCR_xSWS_WIDTH) - 1) << ESAI_xCR_xSWS_SHIFT) +#define ESAI_xCR_xSWS(s, w) ((w < 24 ? (s - w + ((w - 8) >> 2)) : (s < 32 ? 0x1e : 0x1f)) << ESAI_xCR_xSWS_SHIFT) +#define ESAI_xCR_xMOD_SHIFT 8 +#define ESAI_xCR_xMOD_WIDTH 2 +#define ESAI_xCR_xMOD_MASK (((1 << ESAI_xCR_xMOD_WIDTH) - 1) << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xMOD_ONDEMAND (0x1 << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xMOD_NETWORK (0x1 << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xMOD_AC97 (0x3 << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xWA_SHIFT 7 +#define ESAI_xCR_xWA_MASK (1 << ESAI_xCR_xWA_SHIFT) +#define ESAI_xCR_xWA (1 << ESAI_xCR_xWA_SHIFT) +#define ESAI_xCR_xSHFD_SHIFT 6 +#define ESAI_xCR_xSHFD_MASK (1 << ESAI_xCR_xSHFD_SHIFT) +#define ESAI_xCR_xSHFD (1 << ESAI_xCR_xSHFD_SHIFT) +#define ESAI_xCR_xE_SHIFT 0 +#define ESAI_xCR_TE_WIDTH 6 +#define ESAI_xCR_RE_WIDTH 4 +#define ESAI_xCR_TE_MASK (((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) +#define ESAI_xCR_RE_MASK (((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) +#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_TE_MASK) +#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_RE_MASK) + +/* + * Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8 + * Receive Clock Control Register -- REG_ESAI_RCCR 0xE0 + */ +#define ESAI_xCCR_xHCKD_SHIFT 23 +#define ESAI_xCCR_xHCKD_MASK (1 << ESAI_xCCR_xHCKD_SHIFT) +#define ESAI_xCCR_xHCKD (1 << ESAI_xCCR_xHCKD_SHIFT) +#define ESAI_xCCR_xFSD_SHIFT 22 +#define ESAI_xCCR_xFSD_MASK (1 << ESAI_xCCR_xFSD_SHIFT) +#define ESAI_xCCR_xFSD (1 << ESAI_xCCR_xFSD_SHIFT) +#define ESAI_xCCR_xCKD_SHIFT 21 +#define ESAI_xCCR_xCKD_MASK (1 << ESAI_xCCR_xCKD_SHIFT) +#define ESAI_xCCR_xCKD (1 << ESAI_xCCR_xCKD_SHIFT) +#define ESAI_xCCR_xHCKP_SHIFT 20 +#define ESAI_xCCR_xHCKP_MASK (1 << ESAI_xCCR_xHCKP_SHIFT) +#define ESAI_xCCR_xHCKP (1 << ESAI_xCCR_xHCKP_SHIFT) +#define ESAI_xCCR_xFSP_SHIFT 19 +#define ESAI_xCCR_xFSP_MASK (1 << ESAI_xCCR_xFSP_SHIFT) +#define ESAI_xCCR_xFSP (1 << ESAI_xCCR_xFSP_SHIFT) +#define ESAI_xCCR_xCKP_SHIFT 18 +#define ESAI_xCCR_xCKP_MASK (1 << ESAI_xCCR_xCKP_SHIFT) +#define ESAI_xCCR_xCKP (1 << ESAI_xCCR_xCKP_SHIFT) +#define ESAI_xCCR_xFP_SHIFT 14 +#define ESAI_xCCR_xFP_WIDTH 4 +#define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT) +#define ESAI_xCCR_xFP(v) ((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK) +#define ESAI_xCCR_xDC_SHIFT 9 +#define ESAI_xCCR_xDC_WIDTH 4 +#define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT) +#define ESAI_xCCR_xDC(v) ((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK) +#define ESAI_xCCR_xPSR_SHIFT 8 +#define ESAI_xCCR_xPSR_MASK (1 << ESAI_xCCR_xPSR_SHIFT) +#define ESAI_xCCR_xPSR_BYPASS (1 << ESAI_xCCR_xPSR_SHIFT) +#define ESAI_xCCR_xPSR_DIV8 (0 << ESAI_xCCR_xPSR_SHIFT) +#define ESAI_xCCR_xPM_SHIFT 0 +#define ESAI_xCCR_xPM_WIDTH 8 +#define ESAI_xCCR_xPM_MASK (((1 << ESAI_xCCR_xPM_WIDTH) - 1) << ESAI_xCCR_xPM_SHIFT) +#define ESAI_xCCR_xPM(v) ((((v) - 1) << ESAI_xCCR_xPM_SHIFT) & ESAI_xCCR_xPM_MASK) + +/* Transmit Slot Mask Register A/B -- REG_ESAI_TSMA/B 0xE4 ~ 0xF0 */ +#define ESAI_xSMA_xS_SHIFT 0 +#define ESAI_xSMA_xS_WIDTH 16 +#define ESAI_xSMA_xS_MASK (((1 << ESAI_xSMA_xS_WIDTH) - 1) << ESAI_xSMA_xS_SHIFT) +#define ESAI_xSMA_xS(v) ((v) & ESAI_xSMA_xS_MASK) +#define ESAI_xSMB_xS_SHIFT 0 +#define ESAI_xSMB_xS_WIDTH 16 +#define ESAI_xSMB_xS_MASK (((1 << ESAI_xSMB_xS_WIDTH) - 1) << ESAI_xSMB_xS_SHIFT) +#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMA_xS_MASK) + +/* Port C Direction Register -- REG_ESAI_PRRC 0xF8 */ +#define ESAI_PRRC_PDC_SHIFT 0 +#define ESAI_PRRC_PDC_WIDTH 12 +#define ESAI_PRRC_PDC_MASK (((1 << ESAI_PRRC_PDC_WIDTH) - 1) << ESAI_PRRC_PDC_SHIFT) +#define ESAI_PRRC_PDC(v) ((v) & ESAI_PRRC_PDC_MASK) + +/* Port C Control Register -- REG_ESAI_PCRC 0xFC */ +#define ESAI_PCRC_PC_SHIFT 0 +#define ESAI_PCRC_PC_WIDTH 12 +#define ESAI_PCRC_PC_MASK (((1 << ESAI_PCRC_PC_WIDTH) - 1) << ESAI_PCRC_PC_SHIFT) +#define ESAI_PCRC_PC(v) ((v) & ESAI_PCRC_PC_MASK) + +#define ESAI_GPIO 0xfff + +/* ESAI clock source */ +#define ESAI_HCKT_FSYS 0 +#define ESAI_HCKT_EXTAL 1 +#define ESAI_HCKR_FSYS 2 +#define ESAI_HCKR_EXTAL 3 + +/* ESAI clock divider */ +#define ESAI_TX_DIV_PSR 0 +#define ESAI_TX_DIV_PM 1 +#define ESAI_TX_DIV_FP 2 +#define ESAI_RX_DIV_PSR 3 +#define ESAI_RX_DIV_PM 4 +#define ESAI_RX_DIV_FP 5 +#endif /* _FSL_ESAI_DAI_H */ -- 1.8.4

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[alsa-devel] [PATCH v2] ASoC: imx: Allow using sound with DT descriptions only
by Alexander Shiyan 10 Jan '14

10 Jan '14

This patch allow using generic sound cards/codecs based on DT description. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/Kconfig | 25 +++---------------------- sound/soc/fsl/Makefile | 5 ++--- sound/soc/fsl/imx-pcm.h | 7 ------- 3 files changed, 5 insertions(+), 32 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 1c7b3b7..cf0d5af 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -7,6 +7,7 @@ config SND_SOC_FSL_SSI config SND_SOC_FSL_SPDIF tristate + select REGMAP_MMIO config SND_SOC_FSL_UTILS tristate @@ -106,6 +107,8 @@ endif # SND_POWERPC_SOC menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC || COMPILE_TEST + select SND_SOC_FSL_SSI if OF + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the i.MX CPUs. @@ -119,19 +122,11 @@ config SND_SOC_IMX_PCM_FIQ tristate select FIQ -config SND_SOC_IMX_PCM_DMA - tristate - select SND_SOC_GENERIC_DMAENGINE_PCM - -config SND_SOC_IMX_AUDMUX - tristate - config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the i.MX31ADS with WM1133-EV1 fitted" depends on MACH_MX31ADS_WM1133_EV1 select SND_SOC_WM8350 select SND_SOC_IMX_PCM_FIQ - select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help Enable support for audio on the i.MX31ADS with the WM1133-EV1 @@ -141,8 +136,6 @@ config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" depends on MACH_IMX27_VISSTRIM_M10 && I2C select SND_SOC_TLV320AIC32X4 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help Say Y if you want to add support for SoC audio on Visstrim SM10 @@ -154,7 +147,6 @@ config SND_SOC_PHYCORE_AC97 select SND_SOC_AC97_BUS select SND_SOC_WM9712 select SND_SOC_IMX_PCM_FIQ - select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help Say Y if you want to add support for SoC audio on Phytec phyCORE @@ -169,7 +161,6 @@ config SND_SOC_EUKREA_TLV320 depends on I2C select SND_SOC_TLV320AIC23 select SND_SOC_IMX_PCM_FIQ - select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help Enable I2S based access to the TLV320AIC23B codec attached @@ -179,9 +170,6 @@ config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C select SND_SOC_WM8962 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX - select SND_SOC_FSL_SSI help Say Y if you want to add support for SoC audio on an i.MX board with a wm8962 codec. @@ -190,18 +178,13 @@ config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C select SND_SOC_SGTL5000 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX - select SND_SOC_FSL_SSI help Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" - select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF - select REGMAP_MMIO help SoC Audio support for i.MX boards with S/PDIF Say Y if you want to add support for SoC audio on an i.MX board with @@ -211,8 +194,6 @@ config SND_SOC_IMX_MC13783 tristate "SoC Audio support for I.MX boards with mc13783" depends on MFD_MC13XXX && ARM select SND_SOC_IMX_SSI - select SND_SOC_IMX_AUDMUX select SND_SOC_MC13783 - select SND_SOC_IMX_PCM_DMA endif # SND_IMX_SOC diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index aaccbee..5474a0c 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -34,11 +34,10 @@ obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o # i.MX Platform Support snd-soc-imx-ssi-objs := imx-ssi.o snd-soc-imx-audmux-objs := imx-audmux.o +obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o -obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o -obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o -obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm-dma.o +obj-$(CONFIG_SND_IMX_SOC) += imx-pcm-dma.o snd-soc-imx-audmux.o # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index c79cb27..a1364cb 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -38,14 +38,7 @@ struct imx_pcm_fiq_params { struct snd_dmaengine_dai_dma_data *dma_params_tx; }; -#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA) int imx_pcm_dma_init(struct platform_device *pdev); -#else -static inline int imx_pcm_dma_init(struct platform_device *pdev) -{ - return -ENODEV; -} -#endif #if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_FIQ) int imx_pcm_fiq_init(struct platform_device *pdev, -- 1.8.3.2

2 2

[alsa-devel] [PATCH] ALSA: PCM: Warn when buffer preallocation fails
by Takashi Iwai 10 Jan '14

10 Jan '14

The failures of buffer preallocations at driver initializations aren't critical but it's still helpful to inform, so that user can know that something doesn't work as expected. For example, the recent page allocator change triggered regressions, but developers didn't notice until recently because the driver didn't complain. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_memory.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 01f8eafebda6..54debc07f5cb 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -51,6 +51,7 @@ static const size_t snd_minimum_buffer = 16384; static int preallocate_pcm_pages(struct snd_pcm_substream *substream, size_t size) { struct snd_dma_buffer *dmab = &substream->dma_buffer; + size_t orig_size = size; int err; do { @@ -63,6 +64,10 @@ static int preallocate_pcm_pages(struct snd_pcm_substream *substream, size_t siz size >>= 1; } while (size >= snd_minimum_buffer); dmab->bytes = 0; /* tell error */ + pr_warn("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %zu\n", + substream->pcm->card->number, substream->pcm->device, + substream->stream ? 'c' : 'p', substream->number, + substream->pcm->name, orig_size); return 0; } -- 1.8.5.2

1 0

[alsa-devel] [PATCH v2 1/2] mfd: wm5110: Add registers for headphone short circuit control
by Charles Keepax 10 Jan '14

10 Jan '14

Add the registers necessary to enable/disable the headphone short circuit protection. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5110-tables.c | 6 ++++++ include/linux/mfd/arizona/registers.h | 27 +++++++++++++++++++++++++++ 2 files changed, 33 insertions(+), 0 deletions(-) Changes since v1: - Add switch to the control names in the second patch in the series. Thanks, Charles diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 24ec464..30f8163 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -637,6 +637,9 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000491, 0x0000 }, /* R1169 - PDM SPK1 CTRL 2 */ { 0x00000492, 0x0069 }, /* R1170 - PDM SPK2 CTRL 1 */ { 0x00000493, 0x0000 }, /* R1171 - PDM SPK2 CTRL 2 */ + { 0x000004A0, 0x3480 }, /* R1184 - HP1 Short Circuit Ctrl */ + { 0x000004A1, 0x3480 }, /* R1185 - HP2 Short Circuit Ctrl */ + { 0x000004A2, 0x3480 }, /* R1186 - HP3 Short Circuit Ctrl */ { 0x00000500, 0x000C }, /* R1280 - AIF1 BCLK Ctrl */ { 0x00000501, 0x0008 }, /* R1281 - AIF1 Tx Pin Ctrl */ { 0x00000502, 0x0000 }, /* R1282 - AIF1 Rx Pin Ctrl */ @@ -1633,6 +1636,9 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_PDM_SPK1_CTRL_2: case ARIZONA_PDM_SPK2_CTRL_1: case ARIZONA_PDM_SPK2_CTRL_2: + case ARIZONA_HP1_SHORT_CIRCUIT_CTRL: + case ARIZONA_HP2_SHORT_CIRCUIT_CTRL: + case ARIZONA_HP3_SHORT_CIRCUIT_CTRL: case ARIZONA_AIF1_BCLK_CTRL: case ARIZONA_AIF1_TX_PIN_CTRL: case ARIZONA_AIF1_RX_PIN_CTRL: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index b40960d..2d3ac50 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -226,6 +226,9 @@ #define ARIZONA_PDM_SPK1_CTRL_2 0x491 #define ARIZONA_PDM_SPK2_CTRL_1 0x492 #define ARIZONA_PDM_SPK2_CTRL_2 0x493 +#define ARIZONA_HP1_SHORT_CIRCUIT_CTRL 0x4A0 +#define ARIZONA_HP2_SHORT_CIRCUIT_CTRL 0x4A1 +#define ARIZONA_HP3_SHORT_CIRCUIT_CTRL 0x4A2 #define ARIZONA_SPK_CTRL_2 0x4B5 #define ARIZONA_SPK_CTRL_3 0x4B6 #define ARIZONA_DAC_COMP_1 0x4DC @@ -3297,6 +3300,30 @@ #define ARIZONA_SPK2_FMT_WIDTH 1 /* SPK2_FMT */ /* + * R1184 (0x4A0) - HP1 Short Circuit Ctrl + */ +#define ARIZONA_HP1_SC_ENA 0x1000 /* HP1_SC_ENA */ +#define ARIZONA_HP1_SC_ENA_MASK 0x1000 /* HP1_SC_ENA */ +#define ARIZONA_HP1_SC_ENA_SHIFT 12 /* HP1_SC_ENA */ +#define ARIZONA_HP1_SC_ENA_WIDTH 1 /* HP1_SC_ENA */ + +/* + * R1185 (0x4A1) - HP2 Short Circuit Ctrl + */ +#define ARIZONA_HP2_SC_ENA 0x1000 /* HP2_SC_ENA */ +#define ARIZONA_HP2_SC_ENA_MASK 0x1000 /* HP2_SC_ENA */ +#define ARIZONA_HP2_SC_ENA_SHIFT 12 /* HP2_SC_ENA */ +#define ARIZONA_HP2_SC_ENA_WIDTH 1 /* HP2_SC_ENA */ + +/* + * R1186 (0x4A2) - HP3 Short Circuit Ctrl + */ +#define ARIZONA_HP3_SC_ENA 0x1000 /* HP3_SC_ENA */ +#define ARIZONA_HP3_SC_ENA_MASK 0x1000 /* HP3_SC_ENA */ +#define ARIZONA_HP3_SC_ENA_SHIFT 12 /* HP3_SC_ENA */ +#define ARIZONA_HP3_SC_ENA_WIDTH 1 /* HP3_SC_ENA */ + +/* * R1244 (0x4DC) - DAC comp 1 */ #define ARIZONA_OUT_COMP_COEFF_MASK 0xFFFF /* OUT_COMP_COEFF - [15:0] */ -- 1.7.2.5

3 3

[alsa-devel] [PATCH v2 0/6] dma: Indicate residue granularity in dma_slave_caps
by Lars-Peter Clausen 10 Jan '14

10 Jan '14

Here is an updated version of the DMA residue granularity patchset. As discuessed last time it would be best if this could go through ASoC tree. It needs some feedback from the DMA folks first though. New in v2: * Dropped byte level granularity as discussed during v1 review * Added support for auto-detecting the NO_RESIDUE flag on the compat path in the generic ASoC dmaengine PCM driver * Updated a few ASoC drivers to take advantage of the new API Updated original cover text from v1 below: This patch series introduces a new field to the dma_slave_caps struct which exposes the granularity with which the reported residue is updated, updates the pl330 DMA driver to set this field and lets the generic dmaengine PCM audio driver check that field. The reason for introducing the field is that for audio we need to know the granularity with which the reported residue is updated. The uncertainty of the ALSA period pointer depends on the granularity of the reported residue. E.g. if residue is only updated once every period this means that the position in the buffer from/to which the hardware is currently reading/writing could be anywhere between the reported PCM pointer and the reported PCM pointer plus one period size (plus some extra margin for interrupt latency, etc). Certain algorithms (e.g. pulseaudio's timer based scheduling) only work correctly if the uncertainty of the PCM pointer is low enough. Exposing this information from the DMA driver makes it possible for applications to switch to alternative modes of operation which also work with a high granularity. The patch series introduces four levels of granularity, which I think will cover most DMA cores. I'd like some feedback though if anybody can think of a DMA controller for which none of the levels would be correct. The four levels are: * DESCRIPTOR: The DMA channel is only able to tell whether a descriptor has been completed or not, which means residue reporting is not supported by this channel. The residue field of the dma_tx_state field will always be 0. * SEGMENT: The DMA channel updates the residue field after each successfully completed segment of the transfer (For cyclic transfers this is after each period). This is typically implemented by having the hardware generate an interrupt after each transferred segment and then the driver updates the outstanding residue by the size of the segment. Another possibility is if the hardware supports SG and the segment descriptor has a field which gets set by the hardware after the segment has been completed. The driver then counts the number of segments which do not have the flag set to compute the residue. * BURST: The DMA channel updates the residue field after each transferred burst. This is typically only supported if the hardware has a progress register of some sort (E.g. a register with the current read/write address or a register with the amount of bursts/beats/bytes that have been transferred or still need to be transferred). - Lars Lars-Peter Clausen (6): dma: Indicate residue granularity in dma_slave_caps dma: pl330: Set residue_granularity ASoC: generic-dmaengine-pcm: Check NO_RESIDUE flag at runtime ASoC: generic-dmaengine-pcm: Check DMA residue granularity ASoC: axi-{spdif,i2s}: Drop NO_RESIDUE flag ASoC: samsung: Remove NO_RESIDUE flag drivers/dma/pl330.c | 1 + include/linux/dmaengine.h | 15 ++++++++ sound/soc/adi/axi-i2s.c | 3 +- sound/soc/adi/axi-spdif.c | 3 +- sound/soc/samsung/dmaengine.c | 1 - sound/soc/soc-generic-dmaengine-pcm.c | 66 ++++++++++++++++++++++------------- 6 files changed, 60 insertions(+), 29 deletions(-) -- 1.8.0

3 10

[alsa-devel] [PATCH] a52: switch to AV_CODEC_ID identifiers
by Anton Khirnov 10 Jan '14

10 Jan '14

Fixes build with latest libavcodec versions. --- a52/pcm_a52.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/a52/pcm_a52.c b/a52/pcm_a52.c index 359608f..b467ec8 100644 --- a/a52/pcm_a52.c +++ b/a52/pcm_a52.c @@ -58,6 +58,9 @@ #endif #endif +#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(54, 25, 0) +#define AV_CODEC_ID_AC3 CODEC_ID_AC3 +#endif struct a52_ctx { snd_pcm_ioplug_t io; @@ -916,7 +919,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(a52) if (rec->codec == NULL) rec->codec = avcodec_find_encoder_by_name("ac3"); if (rec->codec == NULL) - rec->codec = avcodec_find_encoder(CODEC_ID_AC3); + rec->codec = avcodec_find_encoder(AV_CODEC_ID_AC3); if (rec->codec == NULL) { SNDERR("Cannot find codec engine"); err = -EINVAL; -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ALSA: ctxfi - Work around emu20k1 glitch to prevent buffered sound data loss
by Sarah Bessmer 10 Jan '14

10 Jan '14

Occasionally, on playback stream ringbuffer wraparound, the EMU20K1 hardware will momentarily return 0 instead of the proper current(loop) address. This patch handles that case, fixing the problem of playback position corruption and subsequent loss of buffered sound data, that occurs with some common buffering layout patterns(e.g. multiple simultaneous output streams with differently-sized or non-power-of-2-sized buffers). An alternate means of fixing the problem would be to read the ca register continuously, until two sequential reads return the same value; however, that would be a more invasive change, has performance implications, and isn't necessary unless there are also issues with the value not being updated atomically in regards to individual bits or something similar(which I have not encountered through light testing). I have no EMU20K2 hardware to confirm if the issue is present there, but even if it's not, this change shouldn't break anything that's not already broken. Signed-off-by: Sarah Bessmer <aotos(a)fastmail.fm> Cc: Takashi Iwai <tiwai(a)suse.de> --- --- linux-3.10.25/sound/pci/ctxfi/ctatc.c.orig 2014-01-09 10:12:00.000000000 -0800 +++ linux-3.10.25/sound/pci/ctxfi/ctatc.c 2014-01-09 10:32:42.000000000 -0800 @@ -433,10 +433,15 @@ atc_pcm_playback_position(struct ct_atc if (!src) return 0; position = src->ops->get_ca(src); + if(position < apcm->vm_block->addr) { + snd_printdd("ctxfi: bad ca - ca=0x%08x, vba=0x%08x, vbs=0x%08x\n", position, apcm->vm_block->addr, apcm->vm_block->size); + position = apcm->vm_block->addr; + } + size = apcm->vm_block->size; max_cisz = src->multi * src->rsc.msr; max_cisz = 128 * (max_cisz < 8 ? max_cisz : 8); return (position + size - max_cisz - apcm->vm_block->addr) % size;

2 1

[alsa-devel] [PATCH] ASoC: simple-card: fix one bug to writing to the platform data
by Xiubo Li 10 Jan '14

10 Jan '14

It's a bug that writing to the platform data directly, for it should be constant. So just copy it before writing. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 40 +++++++++++++++++++++------------------- 1 file changed, 21 insertions(+), 19 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 406e4ea..5b65324 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -9,9 +9,10 @@ * published by the Free Software Foundation. */ #include <linux/clk.h> +#include <linux/module.h> #include <linux/of.h> #include <linux/platform_device.h> -#include <linux/module.h> +#include <linux/string.h> #include <sound/simple_card.h> #define asoc_simple_get_card_info(p) \ @@ -204,36 +205,37 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct device_node *of_cpu, *of_codec, *of_platform; struct device *dev = &pdev->dev; + int ret; cinfo = NULL; of_cpu = NULL; of_codec = NULL; of_platform = NULL; + + cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); + if (!cinfo) + return -ENOMEM; + if (np && of_device_is_available(np)) { - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); - if (cinfo) { - int ret; - cinfo->snd_card.dev = &pdev->dev; - ret = asoc_simple_card_parse_of(np, cinfo, dev, - &of_cpu, - &of_codec, - &of_platform); - if (ret < 0) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "parse error %d\n", ret); - return ret; - } - } else { - return -ENOMEM; + cinfo->snd_card.dev = dev; + + ret = asoc_simple_card_parse_of(np, cinfo, dev, + &of_cpu, + &of_codec, + &of_platform); + if (ret < 0) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "parse error %d\n", ret); + return ret; } } else { - cinfo = pdev->dev.platform_data; - if (!cinfo) { + if (!pdev->dev.platform_data) { dev_err(dev, "no info for asoc-simple-card\n"); return -EINVAL; } - cinfo->snd_card.dev = &pdev->dev; + memcpy(cinfo, pdev->dev.platform_data, sizeof(cinfo)); + cinfo->snd_card.dev = dev; } if (!cinfo->name || -- 1.8.4

1 0

[alsa-devel] [PATCH v2] ASoC: fsl_esai: Add ESAI CPU DAI driver
by Nicolin Chen 09 Jan '14

09 Jan '14

This patch implements a device-tree-only CPU DAI driver for Freescale ESAI controller that supports: - 12 channels playback and 8 channels record. [ Some of the inner transmitters and receivers are sharing same group of pins. So the maxmium 12 output or 8 input channels are only valid if there is no pin conflict occurring to it. ] - Independent (asynchronous mode) or shared (synchronous mode) transmit and receive sections with separate or shared internal/external clocks and frame syncs, operating in Master or Slave mode. [ Current ALSA seems not to allow CPU DAI drivers to set DAI format and clks separately for PLAYBACK and CAPTURE. So this first version only supports the case that uses the same DAI format for both directions. ] - Various DAI formats: I2S, Left-Justified, Right-Justified, DSP-A and DSP-B. - Programmable word length (8, 16, 20 or 24bits) - Flexible selection between system clock or external oscillator as input clock source, programmable internal clock divider and frame sync generation. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v2: * Correct multi-line comments. * Add missing spaces between number and operator. .../devicetree/bindings/sound/fsl,esai.txt | 54 ++ sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_esai.c | 815 +++++++++++++++++++++ sound/soc/fsl/fsl_esai.h | 352 +++++++++ 5 files changed, 1226 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,esai.txt create mode 100644 sound/soc/fsl/fsl_esai.c create mode 100644 sound/soc/fsl/fsl_esai.h diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt new file mode 100644 index 0000000..4372833 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt @@ -0,0 +1,54 @@ +Freescale Enhanced Serial Audio Interface (ESAI) Controller + +The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port +for serial communication with a variety of serial devices, including industry +standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and +other DSPs. It has up to six transmitters and four receivers. + +Required properties: + + - compatible : Compatible list, must contain "fsl,imx35-esai". + + - reg : Offset and length of the register set for the device. + + - interrupts : Contains the spdif interrupt. + + - dmas : Generic dma devicetree binding as described in + Documentation/devicetree/bindings/dma/dma.txt. + + - dma-names : Two dmas have to be defined, "tx" and "rx". + + - clocks: Contains an entry for each entry in clock-names. + + - clock-names : Includes the following entries: + "core" The core clock used to access registers + "extal" The esai baud clock for esai controller used to derive + HCK, SCK and FS. + "fsys" The system clock derived from ahb clock used to derive + HCK, SCK and FS. + + - fsl,fifo-depth: The number of elements in the transmit and receive FIFOs. + This number is the maximum allowed value for TFCR[TFWM] or RFCR[RFWM]. + + - fsl,esai-synchronous: This is a boolean property. If present, indicating + that ESAI would work in the synchronous mode, which means all the settings + for Receiving would be duplicated from Transmition related registers. + + - fsl,esai-network-mode: This is a boolean property. If present, indicating + that ESAI controller would work in its network mode. + +Example: + +esai: esai@02024000 { + compatible = "fsl,imx35-esai"; + reg = <0x02024000 0x4000>; + interrupts = <0 51 0x04>; + clocks = <&clks 208>, <&clks 118>, <&clks 208>; + clock-names = "core", "extal", "fsys"; + dmas = <&sdma 23 21 0>, <&sdma 24 21 0>; + dma-names = "rx", "tx"; + fsl,fifo-depth = <128>; + fsl,esai-synchronous; + fsl,esai-network-mode; + status = "disabled"; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 514c275..324988d 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -8,6 +8,9 @@ config SND_SOC_FSL_SSI config SND_SOC_FSL_SPDIF tristate +config SND_SOC_FSL_ESAI + tristate + config SND_SOC_FSL_UTILS tristate diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index aaccbee..b12ad4b 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -14,11 +14,13 @@ obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-objs := fsl_ssi.o snd-soc-fsl-spdif-objs := fsl_spdif.o +snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o +obj-$(CONFIG_SND_SOC_FSL_ESAI) += snd-soc-fsl-esai.o obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c new file mode 100644 index 0000000..a30b1fd --- /dev/null +++ b/sound/soc/fsl/fsl_esai.c @@ -0,0 +1,815 @@ +/* + * Freescale ESAI ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/dmaengine.h> +#include <linux/module.h> +#include <linux/of_irq.h> +#include <linux/of_platform.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> + +#include "fsl_esai.h" +#include "imx-pcm.h" + +#define FSL_ESAI_RATES SNDRV_PCM_RATE_8000_192000 +#define FSL_ESAI_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/** + * fsl_esai: ESAI private data + * + * @dma_params_rx: DMA parameters for receive channel + * @dma_params_tx: DMA parameters for transmit channel + * @pdev: platform device pointer + * @regmap: regmap handler + * @coreclk: clock source to access register + * @extalclk: esai clock source to derive HCK, SCK and FS + * @fsysclk: system clock source to derive HCK, SCK and FS + * @fifo_depth: depth of tx/rx FIFO + * @sck_div: if using PSR/PM dividers for SCK clock + * @network_mode: if using network mode + * @synchronous: if using tx/rx synchronous mode + * @name: driver name + */ +struct fsl_esai { + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; + struct platform_device *pdev; + struct regmap *regmap; + struct clk *coreclk; + struct clk *extalclk; + struct clk *fsysclk; + u32 fifo_depth; + u32 slot_width; + bool sck_div; + bool network_mode; + bool synchronous; + char name[32]; +}; + +static irqreturn_t esai_isr(int irq, void *devid) +{ + struct fsl_esai *esai_priv = (struct fsl_esai *)devid; + struct platform_device *pdev = esai_priv->pdev; + u32 esr; + + regmap_read(esai_priv->regmap, REG_ESAI_ESR, &esr); + + if (esr & ESAI_ESR_TINIT_MASK) + dev_dbg(&pdev->dev, "isr: Transmition Initialized\n"); + + if (esr & ESAI_ESR_RFF_MASK) + dev_warn(&pdev->dev, "isr: Receiving overrun\n"); + + if (esr & ESAI_ESR_TFE_MASK) + dev_warn(&pdev->dev, "isr: Transmition underrun\n"); + + if (esr & ESAI_ESR_TLS_MASK) + dev_dbg(&pdev->dev, "isr: Just transmitted the last slot\n"); + + if (esr & ESAI_ESR_TDE_MASK) + dev_dbg(&pdev->dev, "isr: Transmition data exception\n"); + + if (esr & ESAI_ESR_TED_MASK) + dev_dbg(&pdev->dev, "isr: Transmitting even slots\n"); + + if (esr & ESAI_ESR_TD_MASK) + dev_dbg(&pdev->dev, "isr: Transmitting data\n"); + + if (esr & ESAI_ESR_RLS_MASK) + dev_dbg(&pdev->dev, "isr: Just received the last slot\n"); + + if (esr & ESAI_ESR_RDE_MASK) + dev_dbg(&pdev->dev, "isr: Receiving data exception\n"); + + if (esr & ESAI_ESR_RED_MASK) + dev_dbg(&pdev->dev, "isr: Receiving even slots\n"); + + if (esr & ESAI_ESR_RD_MASK) + dev_dbg(&pdev->dev, "isr: Receiving data\n"); + + return IRQ_HANDLED; +} + +/** + * This function is used to calculate the divisors of psr, pm, fp and it is + * supposed to be called in set_dai_sysclk() and set_bclk_ratio(). + * + * @ratio: desired overall ratio for the paticipating dividers + * @usefp: for HCK setting, there is no need to set fp divider + * @fp: bypass other dividers by setting fp directly if fp != 0 + */ +static int fsl_esai_divisor_cal(struct snd_soc_dai *dai, u32 ratio, + bool usefp, u32 fp) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + u32 psr, pm = 999, maxfp, prod, sub, savesub, i, j; + + maxfp = usefp ? 16 : 1; + + if (usefp && fp) + goto out_fp; + + if (ratio > 2 * 8 * 256 * maxfp || ratio < 2) { + dev_err(dai->dev, "the ratio is out of range (2 ~ %d)\n", + 2 * 8 * 256 * maxfp); + return -EINVAL; + } else if (ratio % 2) { + dev_err(dai->dev, "the raio must be even if using upper divider\n"); + return -EINVAL; + } + + ratio /= 2; + + psr = ratio <= 256 * maxfp ? ESAI_xCCR_xPSR_BYPASS : ESAI_xCCR_xPSR_DIV8; + + /* Set the max fluctuation -- 0.1% of the max devisor */ + savesub = (psr ? 1 : 8) * 256 * maxfp / 1000; + + /* Find the best value for PM */ + for (i = 1; i <= 256; i++) { + for (j = 1; j <= maxfp; j++) { + /* PSR (1 or 8) * PM (1 ~ 256) * FP (1 ~ 16) */ + prod = (psr ? 1 : 8) * i * j; + + if (prod == ratio) + sub = 0; + else if (prod / ratio == 1) + sub = prod - ratio; + else if (ratio / prod == 1) + sub = ratio - prod; + else + continue; + + /* Calculate the fraction */ + sub = sub * 1000 / ratio; + if (sub < savesub) { + savesub = sub; + pm = i; + fp = j; + } + + /* We are lucky */ + if (savesub == 0) + goto out; + } + } + + if (pm == 999) { + dev_err(dai->dev, "failed to calculate proper divisors\n"); + return -EINVAL; + } + +out: + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, + ESAI_xCCR_xPSR_MASK | ESAI_xCCR_xPM_MASK, + psr | ESAI_xCCR_xPM(pm)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, + ESAI_xCCR_xPSR_MASK | ESAI_xCCR_xPM_MASK, + psr | ESAI_xCCR_xPM(pm)); + +out_fp: + /* Bypass fp if not being required */ + if (maxfp <= 1) + return 0; + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, + ESAI_xCCR_xFP_MASK, ESAI_xCCR_xFP(fp)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, + ESAI_xCCR_xFP_MASK, ESAI_xCCR_xFP(fp)); + + return 0; +} + +/** + * This function mainly configures the clock frequency of MCLK (HCK) + * + * @Parameters: + * clk_id: The clock source of HCK + * (Input from outside; output from inside, FSYS or EXTAL) + * freq: The required clock rate of HCK + * dir: The clock direction of HCK + * + * Note: If the direction is input, we do not care about clk_id. + */ +static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + bool in = dir == SND_SOC_CLOCK_IN; + u32 ret, ratio, ecr = 0; + unsigned long clk_rate; + + /* sck_div can be only bypassed if ETO/ERO=0 and SNC_SOC_CLOCK_OUT */ + esai_priv->sck_div = true; + + /* Set the direction of HCK pins */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, + ESAI_xCCR_xHCKD, in ? 0 : ESAI_xCCR_xHCKD); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, + ESAI_xCCR_xHCKD, in ? 0 : ESAI_xCCR_xHCKD); + + if (in) + goto out; + + switch (clk_id) { + case ESAI_HCK_FSYS: + if (IS_ERR(esai_priv->fsysclk)) { + dev_err(dai->dev, "no assigned fsys clock\n"); + return -EINVAL; + } + clk_rate = clk_get_rate(esai_priv->fsysclk); + break; + case ESAI_HCK_EXTAL: + if (IS_ERR(esai_priv->extalclk)) { + dev_err(dai->dev, "no assigned extal clock\n"); + return -EINVAL; + } + clk_rate = clk_get_rate(esai_priv->extalclk); + + ecr |= ESAI_ECR_ETI | ESAI_ECR_ERI; + break; + default: + return -EINVAL; + } + + ratio = clk_rate / freq; + if (ratio * freq > clk_rate) + ret = ratio * freq - clk_rate; + else if (ratio * freq < clk_rate) + ret = clk_rate - ratio * freq; + else + ret = 0; + + /* Block if clock source can not be divided into the required rate */ + if (ret != 0 && clk_rate / ret < 1000) { + dev_err(dai->dev, "failed to derive the required frequency\n"); + return -EINVAL; + } + + if (ratio == 1) { + /* Bypass all the dividers if not being needed */ + ecr |= ESAI_ECR_ETO | ESAI_ECR_ERO; + goto out; + } + + ret = fsl_esai_divisor_cal(dai, ratio, false, 0); + if (ret) + return ret; + + esai_priv->sck_div = false; + +out: + regmap_update_bits(esai_priv->regmap, REG_ESAI_ECR, + ESAI_ECR_ETI | ESAI_ECR_ETO | + ESAI_ECR_ERI | ESAI_ECR_ERO, ecr); + + return 0; +} + +/** + * This function configures the ratio between MCLK (HCK) and BCLK (SCK) + * (For DAI Master Mode only) + * + * Note: Machine driver should calculate the ratio to call this function. + * Only effective after calling set_dai_sysclk() to set HCK direction. + */ +static int fsl_esai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + u32 fp = esai_priv->sck_div ? 0 : ratio; + + if (!esai_priv->sck_div && (ratio > 16 || ratio == 0)) { + dev_err(dai->dev, "the ratio is out of range (1 ~ 16)\n"); + return -EINVAL; + } + + return fsl_esai_divisor_cal(dai, ratio, true, fp); +} + +/** + * This function provides a manual way to configure all the dividers directly + * (For DAI Master Mode only) + * + * By using this function, the driver would allow user to configure a flexible + * combination of divisors for both TX and RX route separately + * + * Note: Only effective after calling set_dai_sysclk() to set HCK direction + */ +static int fsl_esai_set_dai_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + bool tx = div_id <= ESAI_TX_DIV_FP; + u32 mask, val; + + switch (div_id) { + case ESAI_TX_DIV_PSR: + case ESAI_RX_DIV_PSR: + mask = ESAI_xCCR_xPSR_MASK; + val = div ? ESAI_xCCR_xPSR_BYPASS : 0; + break; + case ESAI_TX_DIV_PM: + case ESAI_RX_DIV_PM: + mask = ESAI_xCCR_xPM_MASK; + val = ESAI_xCCR_xPM(div); + break; + case ESAI_TX_DIV_FP: + case ESAI_RX_DIV_FP: + mask = ESAI_xCCR_xFP_MASK; + val = ESAI_xCCR_xFP(div); + break; + default: + return -EINVAL; + } + + return regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx), mask, val); +} + +static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, + u32 rx_mask, int slots, int slot_width) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, + ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, + ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(tx_mask)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, + ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, + ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(rx_mask)); + + esai_priv->slot_width = slot_width; + + return 0; +} + +static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + u32 xcr = 0, xccr = 0, mask; + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* Data on rising edge of bclk, frame low, 1clk before data */ + xcr |= ESAI_xCR_xFSR; + xccr |= ESAI_xCCR_xFSP | ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* Data on rising edge of bclk, frame high */ + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + /* Data on rising edge of bclk, frame high, right aligned */ + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Data on rising edge of bclk, frame high, 1clk before data */ + xcr |= ESAI_xCR_xFSL | ESAI_xCR_xFSR; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_DSP_B: + /* Data on rising edge of bclk, frame high */ + xcr |= ESAI_xCR_xFSL; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do for both normal cases */ + break; + case SND_SOC_DAIFMT_IB_NF: + /* Invert bit clock */ + xccr ^= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_NB_IF: + /* Invert frame clock */ + xccr ^= ESAI_xCCR_xFSP; + break; + case SND_SOC_DAIFMT_IB_IF: + /* Invert both clocks */ + xccr ^= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP; + break; + default: + return -EINVAL; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFM: + xccr |= ESAI_xCCR_xCKD; + break; + case SND_SOC_DAIFMT_CBM_CFS: + xccr |= ESAI_xCCR_xFSD; + break; + case SND_SOC_DAIFMT_CBS_CFS: + xccr |= ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; + break; + default: + return -EINVAL; + } + + mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR; + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr); + + mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP | + ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA; + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr); + + return 0; +} + +static int fsl_esai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + /* + * Some platforms might use the same bit to gate all three or two of + * clocks, so keep all clocks open/close at the same time for safety + */ + clk_prepare_enable(esai_priv->coreclk); + if (!IS_ERR(esai_priv->extalclk)) + clk_prepare_enable(esai_priv->extalclk); + if (!IS_ERR(esai_priv->fsysclk)) + clk_prepare_enable(esai_priv->fsysclk); + + if (!dai->active) { + /* Reset Port C */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC, + ESAI_PRRC_PDC_MASK, ESAI_PRRC_PDC(ESAI_GPIO)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC, + ESAI_PCRC_PC_MASK, ESAI_PCRC_PC(ESAI_GPIO)); + + /* Set synchronous mode */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_SAICR, + ESAI_SAICR_SYNC, esai_priv->synchronous ? + ESAI_SAICR_SYNC : 0); + + /* Set network mode */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_xMOD_MASK, esai_priv->network_mode ? + ESAI_xCR_xMOD_NETWORK : 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, + ESAI_xCR_xMOD_MASK, esai_priv->network_mode ? + ESAI_xCR_xMOD_NETWORK : 0); + + /* Set a default slot number -- 2 */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + } + + return 0; +} + +static int fsl_esai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 width = snd_pcm_format_width(params_format(params)); + u32 channels = params_channels(params); + u32 mask, val; + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFR_MASK, ESAI_xFCR_xFR); + + mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK | + (tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK); + val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth / 2) | + (tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val); + + mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0); + val = ESAI_xCR_xSWS(esai_priv->slot_width, width) | (tx ? ESAI_xCR_PADC : 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val); + + return 0; +} + +static void fsl_esai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + if (!IS_ERR(esai_priv->fsysclk)) + clk_disable_unprepare(esai_priv->fsysclk); + if (!IS_ERR(esai_priv->extalclk)) + clk_disable_unprepare(esai_priv->extalclk); + clk_disable_unprepare(esai_priv->coreclk); +} + +static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u8 i, channels = substream->runtime->channels; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFEN_MASK, ESAI_xFCR_xFEN); + + /* Write initial words reqiured by ESAI as normal procedure */ + for (i = 0; tx && i < channels; i++) + regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, + tx ? ESAI_xCR_TE(channels) : ESAI_xCR_RE(channels)); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0); + + /* Disable and reset FIFO */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFR | ESAI_xFCR_xFEN, ESAI_xFCR_xFR); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFR, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops fsl_esai_dai_ops = { + .startup = fsl_esai_startup, + .shutdown = fsl_esai_shutdown, + .trigger = fsl_esai_trigger, + .hw_params = fsl_esai_hw_params, + .set_sysclk = fsl_esai_set_dai_sysclk, + .set_bclk_ratio = fsl_esai_set_bclk_ratio, + .set_clkdiv = fsl_esai_set_dai_clkdiv, + .set_fmt = fsl_esai_set_dai_fmt, + .set_tdm_slot = fsl_esai_set_dai_tdm_slot, +}; + +static int fsl_esai_dai_probe(struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &esai_priv->dma_params_tx, + &esai_priv->dma_params_rx); + + return 0; +} + +static struct snd_soc_dai_driver fsl_esai_dai = { + .probe = fsl_esai_dai_probe, + .playback = { + .channels_min = 1, + .channels_max = 12, + .rates = FSL_ESAI_RATES, + .formats = FSL_ESAI_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = FSL_ESAI_RATES, + .formats = FSL_ESAI_FORMATS, + }, + .ops = &fsl_esai_dai_ops, +}; + +static const struct snd_soc_component_driver fsl_esai_component = { + .name = "fsl-esai", +}; + +static bool fsl_esai_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ESAI_ERDR: + case REG_ESAI_ECR: + case REG_ESAI_ESR: + case REG_ESAI_TFCR: + case REG_ESAI_TFSR: + case REG_ESAI_RFCR: + case REG_ESAI_RFSR: + case REG_ESAI_RX0: + case REG_ESAI_RX1: + case REG_ESAI_RX2: + case REG_ESAI_RX3: + case REG_ESAI_SAISR: + case REG_ESAI_SAICR: + case REG_ESAI_TCR: + case REG_ESAI_TCCR: + case REG_ESAI_RCR: + case REG_ESAI_RCCR: + case REG_ESAI_TSMA: + case REG_ESAI_TSMB: + case REG_ESAI_RSMA: + case REG_ESAI_RSMB: + case REG_ESAI_PRRC: + case REG_ESAI_PCRC: + return true; + default: + return false; + } +} + +static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ESAI_ETDR: + case REG_ESAI_ECR: + case REG_ESAI_TFCR: + case REG_ESAI_RFCR: + case REG_ESAI_TX0: + case REG_ESAI_TX1: + case REG_ESAI_TX2: + case REG_ESAI_TX3: + case REG_ESAI_TX4: + case REG_ESAI_TX5: + case REG_ESAI_TSR: + case REG_ESAI_SAICR: + case REG_ESAI_TCR: + case REG_ESAI_TCCR: + case REG_ESAI_RCR: + case REG_ESAI_RCCR: + case REG_ESAI_TSMA: + case REG_ESAI_TSMB: + case REG_ESAI_RSMA: + case REG_ESAI_RSMB: + case REG_ESAI_PRRC: + case REG_ESAI_PCRC: + return true; + default: + return false; + } +} + +static const struct regmap_config fsl_esai_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + + .max_register = REG_ESAI_PCRC, + .readable_reg = fsl_esai_readable_reg, + .writeable_reg = fsl_esai_writeable_reg, +}; + +static int fsl_esai_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct fsl_esai *esai_priv; + struct resource *res; + const uint32_t *iprop; + void __iomem *regs; + int irq, ret; + + esai_priv = devm_kzalloc(&pdev->dev, sizeof(*esai_priv), GFP_KERNEL); + if (!esai_priv) + return -ENOMEM; + + esai_priv->pdev = pdev; + strcpy(esai_priv->name, np->name); + + /* Get the addresses and IRQ */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + esai_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "core", regs, &fsl_esai_regmap_config); + if (IS_ERR(esai_priv->regmap)) { + dev_err(&pdev->dev, "failed to init regmap: %ld\n", + PTR_ERR(esai_priv->regmap)); + return PTR_ERR(esai_priv->regmap); + } + + esai_priv->coreclk = devm_clk_get(&pdev->dev, "core"); + if (IS_ERR(esai_priv->coreclk)) { + dev_err(&pdev->dev, "failed to get core clock: %ld\n", + PTR_ERR(esai_priv->coreclk)); + return PTR_ERR(esai_priv->coreclk); + } + + esai_priv->extalclk = devm_clk_get(&pdev->dev, "extal"); + if (IS_ERR(esai_priv->extalclk)) + dev_warn(&pdev->dev, "failed to get extal clock: %ld\n", + PTR_ERR(esai_priv->extalclk)); + + esai_priv->fsysclk = devm_clk_get(&pdev->dev, "fsys"); + if (IS_ERR(esai_priv->fsysclk)) + dev_warn(&pdev->dev, "failed to get fsys clock: %ld\n", + PTR_ERR(esai_priv->fsysclk)); + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + return irq; + } + + ret = devm_request_irq(&pdev->dev, irq, esai_isr, 0, + esai_priv->name, esai_priv); + if (ret) { + dev_err(&pdev->dev, "failed to claim irq %u\n", irq); + return ret; + } + + /* Set a default slot size */ + esai_priv->slot_width = 32; + + /* Determine the FIFO depth */ + iprop = of_get_property(np, "fsl,fifo-depth", NULL); + if (iprop) + esai_priv->fifo_depth = be32_to_cpup(iprop); + else + esai_priv->fifo_depth = 64; + + esai_priv->dma_params_tx.maxburst = esai_priv->fifo_depth > 1; + esai_priv->dma_params_rx.maxburst = esai_priv->fifo_depth > 1; + esai_priv->dma_params_tx.addr = res->start + REG_ESAI_ETDR; + esai_priv->dma_params_rx.addr = res->start + REG_ESAI_ERDR; + + esai_priv->synchronous = + of_property_read_bool(np, "fsl,esai-synchronous"); + esai_priv->network_mode = + of_property_read_bool(np, "fsl,esai-network-mode"); + + dev_set_drvdata(&pdev->dev, esai_priv); + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component, + &fsl_esai_dai, 1); + if (ret) { + dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); + return ret; + } + + ret = imx_pcm_dma_init(pdev); + if (ret) + dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); + + /* Reset ESAI unit */ + regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ERST); + + /* + * We need to enable ESAI so as to access some of its registers. + * Otherwise, we would fail to dump regmap from user space. + */ + regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ESAIEN); + + return ret; +} + +static const struct of_device_id fsl_esai_dt_ids[] = { + { .compatible = "fsl,imx35-esai", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids); + +static struct platform_driver fsl_esai_driver = { + .probe = fsl_esai_probe, + .driver = { + .name = "fsl-esai-dai", + .owner = THIS_MODULE, + .of_match_table = fsl_esai_dt_ids, + }, +}; + +module_platform_driver(fsl_esai_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale ESAI CPU DAI driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:fsl-esai-dai"); diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h new file mode 100644 index 0000000..680aac0 --- /dev/null +++ b/sound/soc/fsl/fsl_esai.h @@ -0,0 +1,352 @@ +/* + * fsl_esai.h - ALSA ESAI interface for the Freescale i.MX SoC + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _FSL_ESAI_DAI_H +#define _FSL_ESAI_DAI_H + +/* ESAI Register Map */ +#define REG_ESAI_ETDR 0x00 +#define REG_ESAI_ERDR 0x04 +#define REG_ESAI_ECR 0x08 +#define REG_ESAI_ESR 0x0C +#define REG_ESAI_TFCR 0x10 +#define REG_ESAI_TFSR 0x14 +#define REG_ESAI_RFCR 0x18 +#define REG_ESAI_RFSR 0x1C +#define REG_ESAI_xFCR(tx) (tx ? REG_ESAI_TFCR : REG_ESAI_RFCR) +#define REG_ESAI_xFSR(tx) (tx ? REG_ESAI_TFSR : REG_ESAI_RFSR) +#define REG_ESAI_TX0 0x80 +#define REG_ESAI_TX1 0x84 +#define REG_ESAI_TX2 0x88 +#define REG_ESAI_TX3 0x8C +#define REG_ESAI_TX4 0x90 +#define REG_ESAI_TX5 0x94 +#define REG_ESAI_TSR 0x98 +#define REG_ESAI_RX0 0xA0 +#define REG_ESAI_RX1 0xA4 +#define REG_ESAI_RX2 0xA8 +#define REG_ESAI_RX3 0xAC +#define REG_ESAI_SAISR 0xCC +#define REG_ESAI_SAICR 0xD0 +#define REG_ESAI_TCR 0xD4 +#define REG_ESAI_TCCR 0xD8 +#define REG_ESAI_RCR 0xDC +#define REG_ESAI_RCCR 0xE0 +#define REG_ESAI_xCR(tx) (tx ? REG_ESAI_TCR : REG_ESAI_RCR) +#define REG_ESAI_xCCR(tx) (tx ? REG_ESAI_TCCR : REG_ESAI_RCCR) +#define REG_ESAI_TSMA 0xE4 +#define REG_ESAI_TSMB 0xE8 +#define REG_ESAI_RSMA 0xEC +#define REG_ESAI_RSMB 0xF0 +#define REG_ESAI_xSMA(tx) (tx ? REG_ESAI_TSMA : REG_ESAI_RSMA) +#define REG_ESAI_xSMB(tx) (tx ? REG_ESAI_TSMB : REG_ESAI_RSMB) +#define REG_ESAI_PRRC 0xF8 +#define REG_ESAI_PCRC 0xFC + +/* ESAI Control Register -- REG_ESAI_ECR 0x8 */ +#define ESAI_ECR_ETI_SHIFT 19 +#define ESAI_ECR_ETI_MASK (1 << ESAI_ECR_ETI_SHIFT) +#define ESAI_ECR_ETI (1 << ESAI_ECR_ETI_SHIFT) +#define ESAI_ECR_ETO_SHIFT 18 +#define ESAI_ECR_ETO_MASK (1 << ESAI_ECR_ETO_SHIFT) +#define ESAI_ECR_ETO (1 << ESAI_ECR_ETO_SHIFT) +#define ESAI_ECR_ERI_SHIFT 17 +#define ESAI_ECR_ERI_MASK (1 << ESAI_ECR_ERI_SHIFT) +#define ESAI_ECR_ERI (1 << ESAI_ECR_ERI_SHIFT) +#define ESAI_ECR_ERO_SHIFT 16 +#define ESAI_ECR_ERO_MASK (1 << ESAI_ECR_ERO_SHIFT) +#define ESAI_ECR_ERO (1 << ESAI_ECR_ERO_SHIFT) +#define ESAI_ECR_ERST_SHIFT 1 +#define ESAI_ECR_ERST_MASK (1 << ESAI_ECR_ERST_SHIFT) +#define ESAI_ECR_ERST (1 << ESAI_ECR_ERST_SHIFT) +#define ESAI_ECR_ESAIEN_SHIFT 0 +#define ESAI_ECR_ESAIEN_MASK (1 << ESAI_ECR_ESAIEN_SHIFT) +#define ESAI_ECR_ESAIEN (1 << ESAI_ECR_ESAIEN_SHIFT) + +/* ESAI Status Register -- REG_ESAI_ESR 0xC */ +#define ESAI_ESR_TINIT_SHIFT 10 +#define ESAI_ESR_TINIT_MASK (1 << ESAI_ESR_TINIT_SHIFT) +#define ESAI_ESR_TINIT (1 << ESAI_ESR_TINIT_SHIFT) +#define ESAI_ESR_RFF_SHIFT 9 +#define ESAI_ESR_RFF_MASK (1 << ESAI_ESR_RFF_SHIFT) +#define ESAI_ESR_RFF (1 << ESAI_ESR_RFF_SHIFT) +#define ESAI_ESR_TFE_SHIFT 8 +#define ESAI_ESR_TFE_MASK (1 << ESAI_ESR_TFE_SHIFT) +#define ESAI_ESR_TFE (1 << ESAI_ESR_TFE_SHIFT) +#define ESAI_ESR_TLS_SHIFT 7 +#define ESAI_ESR_TLS_MASK (1 << ESAI_ESR_TLS_SHIFT) +#define ESAI_ESR_TLS (1 << ESAI_ESR_TLS_SHIFT) +#define ESAI_ESR_TDE_SHIFT 6 +#define ESAI_ESR_TDE_MASK (1 << ESAI_ESR_TDE_SHIFT) +#define ESAI_ESR_TDE (1 << ESAI_ESR_TDE_SHIFT) +#define ESAI_ESR_TED_SHIFT 5 +#define ESAI_ESR_TED_MASK (1 << ESAI_ESR_TED_SHIFT) +#define ESAI_ESR_TED (1 << ESAI_ESR_TED_SHIFT) +#define ESAI_ESR_TD_SHIFT 4 +#define ESAI_ESR_TD_MASK (1 << ESAI_ESR_TD_SHIFT) +#define ESAI_ESR_TD (1 << ESAI_ESR_TD_SHIFT) +#define ESAI_ESR_RLS_SHIFT 3 +#define ESAI_ESR_RLS_MASK (1 << ESAI_ESR_RLS_SHIFT) +#define ESAI_ESR_RLS (1 << ESAI_ESR_RLS_SHIFT) +#define ESAI_ESR_RDE_SHIFT 2 +#define ESAI_ESR_RDE_MASK (1 << ESAI_ESR_RDE_SHIFT) +#define ESAI_ESR_RDE (1 << ESAI_ESR_RDE_SHIFT) +#define ESAI_ESR_RED_SHIFT 1 +#define ESAI_ESR_RED_MASK (1 << ESAI_ESR_RED_SHIFT) +#define ESAI_ESR_RED (1 << ESAI_ESR_RED_SHIFT) +#define ESAI_ESR_RD_SHIFT 0 +#define ESAI_ESR_RD_MASK (1 << ESAI_ESR_RD_SHIFT) +#define ESAI_ESR_RD (1 << ESAI_ESR_RD_SHIFT) + +/* + * Transmit FIFO Configuration Register -- REG_ESAI_TFCR 0x10 + * Receive FIFO Configuration Register -- REG_ESAI_RFCR 0x18 + */ +#define ESAI_xFCR_TIEN_SHIFT 19 +#define ESAI_xFCR_TIEN_MASK (1 << ESAI_xFCR_TIEN_SHIFT) +#define ESAI_xFCR_TIEN (1 << ESAI_xFCR_TIEN_SHIFT) +#define ESAI_xFCR_REXT_SHIFT 19 +#define ESAI_xFCR_REXT_MASK (1 << ESAI_xFCR_REXT_SHIFT) +#define ESAI_xFCR_REXT (1 << ESAI_xFCR_REXT_SHIFT) +#define ESAI_xFCR_xWA_SHIFT 16 +#define ESAI_xFCR_xWA_WIDTH 3 +#define ESAI_xFCR_xWA_MASK (((1 << ESAI_xFCR_xWA_WIDTH) - 1) << ESAI_xFCR_xWA_SHIFT) +#define ESAI_xFCR_xWA(v) (((8 - (v >> 2)) << ESAI_xFCR_xWA_SHIFT) & ESAI_xFCR_xWA_MASK) +#define ESAI_xFCR_xFWM_SHIFT 8 +#define ESAI_xFCR_xFWM_WIDTH 8 +#define ESAI_xFCR_xFWM_MASK (((1 << ESAI_xFCR_xFWM_WIDTH) - 1) << ESAI_xFCR_xFWM_SHIFT) +#define ESAI_xFCR_xFWM(v) (((v - 1) << ESAI_xFCR_xFWM_SHIFT) & ESAI_xFCR_xFWM_MASK) +#define ESAI_xFCR_xE_SHIFT 2 +#define ESAI_xFCR_TE_WIDTH 6 +#define ESAI_xFCR_RE_WIDTH 4 +#define ESAI_xFCR_TE_MASK (((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) +#define ESAI_xFCR_RE_MASK (((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) +#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_TE_MASK) +#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_RE_MASK) +#define ESAI_xFCR_xFR_SHIFT 1 +#define ESAI_xFCR_xFR_MASK (1 << ESAI_xFCR_xFR_SHIFT) +#define ESAI_xFCR_xFR (1 << ESAI_xFCR_xFR_SHIFT) +#define ESAI_xFCR_xFEN_SHIFT 0 +#define ESAI_xFCR_xFEN_MASK (1 << ESAI_xFCR_xFEN_SHIFT) +#define ESAI_xFCR_xFEN (1 << ESAI_xFCR_xFEN_SHIFT) + +/* + * Transmit FIFO Status Register -- REG_ESAI_TFSR 0x14 + * Receive FIFO Status Register --REG_ESAI_RFSR 0x1C + */ +#define ESAI_xFSR_NTFO_SHIFT 12 +#define ESAI_xFSR_NRFI_SHIFT 12 +#define ESAI_xFSR_NTFI_SHIFT 8 +#define ESAI_xFSR_NRFO_SHIFT 8 +#define ESAI_xFSR_NTFx_WIDTH 3 +#define ESAI_xFSR_NRFx_WIDTH 2 +#define ESAI_xFSR_NTFO_MASK (((1 << ESAI_xFSR_NTFx_WIDTH) - 1) << ESAI_xFSR_NTFO_SHIFT) +#define ESAI_xFSR_NTFI_MASK (((1 << ESAI_xFSR_NTFx_WIDTH) - 1) << ESAI_xFSR_NTFI_SHIFT) +#define ESAI_xFSR_NRFO_MASK (((1 << ESAI_xFSR_NRFx_WIDTH) - 1) << ESAI_xFSR_NRFO_SHIFT) +#define ESAI_xFSR_NRFI_MASK (((1 << ESAI_xFSR_NRFx_WIDTH) - 1) << ESAI_xFSR_NRFI_SHIFT) +#define ESAI_xFSR_xFCNT_SHIFT 0 +#define ESAI_xFSR_xFCNT_WIDTH 8 +#define ESAI_xFSR_xFCNT_MASK (((1 << ESAI_xFSR_xFCNT_WIDTH) - 1) << ESAI_xFSR_xFCNT_SHIFT) + +/* ESAI Transmit Slot Register -- REG_ESAI_TSR 0x98 */ +#define ESAI_TSR_SHIFT 0 +#define ESAI_TSR_WIDTH 24 +#define ESAI_TSR_MASK (((1 << ESAI_TSR_WIDTH) - 1) << ESAI_TSR_SHIFT) + +/* Serial Audio Interface Status Register -- REG_ESAI_SAISR 0xCC */ +#define ESAI_SAISR_TODFE_SHIFT 17 +#define ESAI_SAISR_TODFE_MASK (1 << ESAI_SAISR_TODFE_SHIFT) +#define ESAI_SAISR_TODFE (1 << ESAI_SAISR_TODFE_SHIFT) +#define ESAI_SAISR_TEDE_SHIFT 16 +#define ESAI_SAISR_TEDE_MASK (1 << ESAI_SAISR_TEDE_SHIFT) +#define ESAI_SAISR_TEDE (1 << ESAI_SAISR_TEDE_SHIFT) +#define ESAI_SAISR_TDE_SHIFT 15 +#define ESAI_SAISR_TDE_MASK (1 << ESAI_SAISR_TDE_SHIFT) +#define ESAI_SAISR_TDE (1 << ESAI_SAISR_TDE_SHIFT) +#define ESAI_SAISR_TUE_SHIFT 14 +#define ESAI_SAISR_TUE_MASK (1 << ESAI_SAISR_TUE_SHIFT) +#define ESAI_SAISR_TUE (1 << ESAI_SAISR_TUE_SHIFT) +#define ESAI_SAISR_TFS_SHIFT 13 +#define ESAI_SAISR_TFS_MASK (1 << ESAI_SAISR_TFS_SHIFT) +#define ESAI_SAISR_TFS (1 << ESAI_SAISR_TFS_SHIFT) +#define ESAI_SAISR_RODF_SHIFT 10 +#define ESAI_SAISR_RODF_MASK (1 << ESAI_SAISR_RODF_SHIFT) +#define ESAI_SAISR_RODF (1 << ESAI_SAISR_RODF_SHIFT) +#define ESAI_SAISR_REDF_SHIFT 9 +#define ESAI_SAISR_REDF_MASK (1 << ESAI_SAISR_REDF_SHIFT) +#define ESAI_SAISR_REDF (1 << ESAI_SAISR_REDF_SHIFT) +#define ESAI_SAISR_RDF_SHIFT 8 +#define ESAI_SAISR_RDF_MASK (1 << ESAI_SAISR_RDF_SHIFT) +#define ESAI_SAISR_RDF (1 << ESAI_SAISR_RDF_SHIFT) +#define ESAI_SAISR_ROE_SHIFT 7 +#define ESAI_SAISR_ROE_MASK (1 << ESAI_SAISR_ROE_SHIFT) +#define ESAI_SAISR_ROE (1 << ESAI_SAISR_ROE_SHIFT) +#define ESAI_SAISR_RFS_SHIFT 6 +#define ESAI_SAISR_RFS_MASK (1 << ESAI_SAISR_RFS_SHIFT) +#define ESAI_SAISR_RFS (1 << ESAI_SAISR_RFS_SHIFT) +#define ESAI_SAISR_IF2_SHIFT 2 +#define ESAI_SAISR_IF2_MASK (1 << ESAI_SAISR_IF2_SHIFT) +#define ESAI_SAISR_IF2 (1 << ESAI_SAISR_IF2_SHIFT) +#define ESAI_SAISR_IF1_SHIFT 1 +#define ESAI_SAISR_IF1_MASK (1 << ESAI_SAISR_IF1_SHIFT) +#define ESAI_SAISR_IF1 (1 << ESAI_SAISR_IF1_SHIFT) +#define ESAI_SAISR_IF0_SHIFT 0 +#define ESAI_SAISR_IF0_MASK (1 << ESAI_SAISR_IF0_SHIFT) +#define ESAI_SAISR_IF0 (1 << ESAI_SAISR_IF0_SHIFT) + +/* Serial Audio Interface Control Register -- REG_ESAI_SAICR 0xD0 */ +#define ESAI_SAICR_ALC_SHIFT 8 +#define ESAI_SAICR_ALC_MASK (1 << ESAI_SAICR_ALC_SHIFT) +#define ESAI_SAICR_ALC (1 << ESAI_SAICR_ALC_SHIFT) +#define ESAI_SAICR_TEBE_SHIFT 7 +#define ESAI_SAICR_TEBE_MASK (1 << ESAI_SAICR_TEBE_SHIFT) +#define ESAI_SAICR_TEBE (1 << ESAI_SAICR_TEBE_SHIFT) +#define ESAI_SAICR_SYNC_SHIFT 6 +#define ESAI_SAICR_SYNC_MASK (1 << ESAI_SAICR_SYNC_SHIFT) +#define ESAI_SAICR_SYNC (1 << ESAI_SAICR_SYNC_SHIFT) +#define ESAI_SAICR_OF2_SHIFT 2 +#define ESAI_SAICR_OF2_MASK (1 << ESAI_SAICR_OF2_SHIFT) +#define ESAI_SAICR_OF2 (1 << ESAI_SAICR_OF2_SHIFT) +#define ESAI_SAICR_OF1_SHIFT 1 +#define ESAI_SAICR_OF1_MASK (1 << ESAI_SAICR_OF1_SHIFT) +#define ESAI_SAICR_OF1 (1 << ESAI_SAICR_OF1_SHIFT) +#define ESAI_SAICR_OF0_SHIFT 0 +#define ESAI_SAICR_OF0_MASK (1 << ESAI_SAICR_OF0_SHIFT) +#define ESAI_SAICR_OF0 (1 << ESAI_SAICR_OF0_SHIFT) + +/* + * Transmit Control Register -- REG_ESAI_TCR 0xD4 + * Receive Control Register -- REG_ESAI_RCR 0xDC + */ +#define ESAI_xCR_xLIE_SHIFT 23 +#define ESAI_xCR_xLIE_MASK (1 << ESAI_xCR_xLIE_SHIFT) +#define ESAI_xCR_xLIE (1 << ESAI_xCR_xLIE_SHIFT) +#define ESAI_xCR_xIE_SHIFT 22 +#define ESAI_xCR_xIE_MASK (1 << ESAI_xCR_xIE_SHIFT) +#define ESAI_xCR_xIE (1 << ESAI_xCR_xIE_SHIFT) +#define ESAI_xCR_xEDIE_SHIFT 21 +#define ESAI_xCR_xEDIE_MASK (1 << ESAI_xCR_xEDIE_SHIFT) +#define ESAI_xCR_xEDIE (1 << ESAI_xCR_xEDIE_SHIFT) +#define ESAI_xCR_xEIE_SHIFT 20 +#define ESAI_xCR_xEIE_MASK (1 << ESAI_xCR_xEIE_SHIFT) +#define ESAI_xCR_xEIE (1 << ESAI_xCR_xEIE_SHIFT) +#define ESAI_xCR_xPR_SHIFT 19 +#define ESAI_xCR_xPR_MASK (1 << ESAI_xCR_xPR_SHIFT) +#define ESAI_xCR_xPR (1 << ESAI_xCR_xPR_SHIFT) +#define ESAI_xCR_PADC_SHIFT 17 +#define ESAI_xCR_PADC_MASK (1 << ESAI_xCR_PADC_SHIFT) +#define ESAI_xCR_PADC (1 << ESAI_xCR_PADC_SHIFT) +#define ESAI_xCR_xFSR_SHIFT 16 +#define ESAI_xCR_xFSR_MASK (1 << ESAI_xCR_xFSR_SHIFT) +#define ESAI_xCR_xFSR (1 << ESAI_xCR_xFSR_SHIFT) +#define ESAI_xCR_xFSL_SHIFT 15 +#define ESAI_xCR_xFSL_MASK (1 << ESAI_xCR_xFSL_SHIFT) +#define ESAI_xCR_xFSL (1 << ESAI_xCR_xFSL_SHIFT) +#define ESAI_xCR_xSWS_SHIFT 10 +#define ESAI_xCR_xSWS_WIDTH 5 +#define ESAI_xCR_xSWS_MASK (((1 << ESAI_xCR_xSWS_WIDTH) - 1) << ESAI_xCR_xSWS_SHIFT) +#define ESAI_xCR_xSWS(s, w) ((w < 24 ? (s - w + ((w - 8) >> 2)) : (s < 32 ? 0x1e : 0x1f)) << ESAI_xCR_xSWS_SHIFT) +#define ESAI_xCR_xMOD_SHIFT 8 +#define ESAI_xCR_xMOD_WIDTH 2 +#define ESAI_xCR_xMOD_MASK (((1 << ESAI_xCR_xMOD_WIDTH) - 1) << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xMOD_ONDEMAND (0x1 << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xMOD_NETWORK (0x1 << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xMOD_AC97 (0x3 << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xWA_SHIFT 7 +#define ESAI_xCR_xWA_MASK (1 << ESAI_xCR_xWA_SHIFT) +#define ESAI_xCR_xWA (1 << ESAI_xCR_xWA_SHIFT) +#define ESAI_xCR_xSHFD_SHIFT 6 +#define ESAI_xCR_xSHFD_MASK (1 << ESAI_xCR_xSHFD_SHIFT) +#define ESAI_xCR_xSHFD (1 << ESAI_xCR_xSHFD_SHIFT) +#define ESAI_xCR_xE_SHIFT 0 +#define ESAI_xCR_TE_WIDTH 6 +#define ESAI_xCR_RE_WIDTH 4 +#define ESAI_xCR_TE_MASK (((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) +#define ESAI_xCR_RE_MASK (((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) +#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_TE_MASK) +#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_RE_MASK) + +/* + * Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8 + * Receive Clock Control Register -- REG_ESAI_RCCR 0xE0 + */ +#define ESAI_xCCR_xHCKD_SHIFT 23 +#define ESAI_xCCR_xHCKD_MASK (1 << ESAI_xCCR_xHCKD_SHIFT) +#define ESAI_xCCR_xHCKD (1 << ESAI_xCCR_xHCKD_SHIFT) +#define ESAI_xCCR_xFSD_SHIFT 22 +#define ESAI_xCCR_xFSD_MASK (1 << ESAI_xCCR_xFSD_SHIFT) +#define ESAI_xCCR_xFSD (1 << ESAI_xCCR_xFSD_SHIFT) +#define ESAI_xCCR_xCKD_SHIFT 21 +#define ESAI_xCCR_xCKD_MASK (1 << ESAI_xCCR_xCKD_SHIFT) +#define ESAI_xCCR_xCKD (1 << ESAI_xCCR_xCKD_SHIFT) +#define ESAI_xCCR_xHCKP_SHIFT 20 +#define ESAI_xCCR_xHCKP_MASK (1 << ESAI_xCCR_xHCKP_SHIFT) +#define ESAI_xCCR_xHCKP (1 << ESAI_xCCR_xHCKP_SHIFT) +#define ESAI_xCCR_xFSP_SHIFT 19 +#define ESAI_xCCR_xFSP_MASK (1 << ESAI_xCCR_xFSP_SHIFT) +#define ESAI_xCCR_xFSP (1 << ESAI_xCCR_xFSP_SHIFT) +#define ESAI_xCCR_xCKP_SHIFT 18 +#define ESAI_xCCR_xCKP_MASK (1 << ESAI_xCCR_xCKP_SHIFT) +#define ESAI_xCCR_xCKP (1 << ESAI_xCCR_xCKP_SHIFT) +#define ESAI_xCCR_xFP_SHIFT 14 +#define ESAI_xCCR_xFP_WIDTH 4 +#define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT) +#define ESAI_xCCR_xFP(v) (((v - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK) +#define ESAI_xCCR_xDC_SHIFT 9 +#define ESAI_xCCR_xDC_WIDTH 4 +#define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT) +#define ESAI_xCCR_xDC(v) (((v - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK) +#define ESAI_xCCR_xPSR_SHIFT 8 +#define ESAI_xCCR_xPSR_MASK (1 << ESAI_xCCR_xPSR_SHIFT) +#define ESAI_xCCR_xPSR_BYPASS (1 << ESAI_xCCR_xPSR_SHIFT) +#define ESAI_xCCR_xPSR_DIV8 (0 << ESAI_xCCR_xPSR_SHIFT) +#define ESAI_xCCR_xPM_SHIFT 0 +#define ESAI_xCCR_xPM_WIDTH 8 +#define ESAI_xCCR_xPM_MASK (((1 << ESAI_xCCR_xPM_WIDTH) - 1) << ESAI_xCCR_xPM_SHIFT) +#define ESAI_xCCR_xPM(v) (((v - 1) << ESAI_xCCR_xPM_SHIFT) & ESAI_xCCR_xPM_MASK) + +/* Transmit Slot Mask Register A/B -- REG_ESAI_TSMA/B 0xE4 ~ 0xF0 */ +#define ESAI_xSMA_xS_SHIFT 0 +#define ESAI_xSMA_xS_WIDTH 16 +#define ESAI_xSMA_xS_MASK (((1 << ESAI_xSMA_xS_WIDTH) - 1) << ESAI_xSMA_xS_SHIFT) +#define ESAI_xSMA_xS(v) (v & ESAI_xSMA_xS_MASK) +#define ESAI_xSMB_xS_SHIFT 0 +#define ESAI_xSMB_xS_WIDTH 16 +#define ESAI_xSMB_xS_MASK (((1 << ESAI_xSMB_xS_WIDTH) - 1) << ESAI_xSMB_xS_SHIFT) +#define ESAI_xSMB_xS(v) ((v >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMA_xS_MASK) + +/* Port C Direction Register -- REG_ESAI_PRRC 0xF8 */ +#define ESAI_PRRC_PDC_SHIFT 0 +#define ESAI_PRRC_PDC_WIDTH 12 +#define ESAI_PRRC_PDC_MASK (((1 << ESAI_PRRC_PDC_WIDTH) - 1) << ESAI_PRRC_PDC_SHIFT) +#define ESAI_PRRC_PDC(v) (v & ESAI_PRRC_PDC_MASK) + +/* Port C Control Register -- REG_ESAI_PCRC 0xFC */ +#define ESAI_PCRC_PC_SHIFT 0 +#define ESAI_PCRC_PC_WIDTH 12 +#define ESAI_PCRC_PC_MASK (((1 << ESAI_PCRC_PC_WIDTH) - 1) << ESAI_PCRC_PC_SHIFT) +#define ESAI_PCRC_PC(v) (v & ESAI_PCRC_PC_MASK) + +#define ESAI_GPIO 0xfff + +/* ESAI clock source */ +#define ESAI_HCK_FSYS 0 +#define ESAI_HCK_EXTAL 1 + +/* ESAI clock divider */ +#define ESAI_TX_DIV_PSR 0 +#define ESAI_TX_DIV_PM 1 +#define ESAI_TX_DIV_FP 2 +#define ESAI_RX_DIV_PSR 3 +#define ESAI_RX_DIV_PM 4 +#define ESAI_RX_DIV_FP 5 +#endif /* _FSL_ESAI_DAI_H */ -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: simple-card: fix the cinfo error check
by Xiubo Li 09 Jan '14

09 Jan '14

If the dt is used and the cinfo is NULL, the -ENOMEM should be return. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 51a4784..406e4ea 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -223,15 +223,17 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dev_err(dev, "parse error %d\n", ret); return ret; } + } else { + return -ENOMEM; } } else { cinfo = pdev->dev.platform_data; - cinfo->snd_card.dev = &pdev->dev; - } + if (!cinfo) { + dev_err(dev, "no info for asoc-simple-card\n"); + return -EINVAL; + } - if (!cinfo) { - dev_err(dev, "no info for asoc-simple-card\n"); - return -EINVAL; + cinfo->snd_card.dev = &pdev->dev; } if (!cinfo->name || -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: simple-card: fix a bug where cinfo will be NULL before using it
by Xiubo Li 09 Jan '14

09 Jan '14

If the dt is not used, the cinfo will be always NULL before using it. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index c67d2db..51a4784 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -225,8 +225,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } } } else { - cinfo->snd_card.dev = &pdev->dev; cinfo = pdev->dev.platform_data; + cinfo->snd_card.dev = &pdev->dev; } if (!cinfo) { -- 1.8.4

2 1

[alsa-devel] [PATCH v2] ASoC: fsl_ssi: Set default slot number for common cases
by Nicolin Chen 09 Jan '14

09 Jan '14

For those platforms using DAI master mode like I2S, it's better to pre-set a default slot number so that there's no need for these common cases to set the slot number from its machine driver any more. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v2: * Correct coding style for multi-line comment. sound/soc/fsl/fsl_ssi.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 94dedcb..aad2a1f 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -711,6 +711,17 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) if (ssi_private->imx_ac97) fsl_ssi_setup_ac97(ssi_private); + /* + * Set a default slot number so that there is no need for those common + * cases like I2S mode to call the extra set_tdm_slot() any more. + */ + if (!ssi_private->imx_ac97) { + write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(2)); + write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(2)); + } + return 0; } -- 1.8.4

2 1

[alsa-devel] [PATCH -next] ASoC: fsl-ssi: Add missing clk_disable_unprepare() on error in fsl_ssi_probe()
by Wei Yongjun 09 Jan '14

09 Jan '14

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Add the missing clk_disable_unprepare() before return from fsl_ssi_probe() in the request irq error handling case. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index e18b4b3..4c6818d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1464,7 +1464,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (ret < 0) { dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); - goto error_irqmap; + goto error_clk; } }

2 1

[alsa-devel] [PATCH] ASoC: wm8804: Allow fine-grained control of the PLL generation
by Daniel Matuschek 09 Jan '14

09 Jan '14

Signed-off-by: Daniel Matuschek <daniel(a)matuschek.net> WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using the "pllid" argument of the "set_pll" function is is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. When pllid=0, the behaviour of the driver does not change. --- sound/soc/codecs/wm8804.c | 18 +++++++++++++----- sound/soc/codecs/wm8804.h | 7 +++++++ 2 files changed, 20 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 1704b1e..e79c408 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -2,6 +2,8 @@ * wm8804.c -- WM8804 S/PDIF transceiver driver * * Copyright 2010-11 Wolfson Microelectronics plc + * patched by Daniel Matuschek <info(a)crazyaudio.com> to allow + * fine-grained control of PLL * * Author: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> * @@ -318,7 +320,7 @@ static struct { #define FIXED_PLL_SIZE ((1ULL << 22) * 10) static int pll_factors(struct pll_div *pll_div, unsigned int target, - unsigned int source) + unsigned int source, int mclk_div) { u64 Kpart; unsigned long int K, Ndiv, Nmod, tmp; @@ -332,9 +334,15 @@ static int pll_factors(struct pll_div *pll_div, unsigned int target, tmp = target * post_table[i].div; if (tmp >= 90000000 && tmp <= 100000000) { pll_div->freqmode = post_table[i].freqmode; - pll_div->mclkdiv = post_table[i].mclkdiv; - target *= post_table[i].div; - break; + if ((mclk_div == WM8804_MCLKDIV_DONTCARE) || + ((post_table[i].mclkdiv == 1) && + (mclk_div == WM8804_MCLKDIV_1)) || + ((post_table[i].mclkdiv == 0) && + (mclk_div == WM8804_MCLKDIV_0))) { + pll_div->mclkdiv = post_table[i].mclkdiv; + target *= post_table[i].div; + break; + } } } @@ -388,7 +396,7 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, int ret; struct pll_div pll_div; - ret = pll_factors(&pll_div, freq_out, freq_in); + ret = pll_factors(&pll_div, freq_out, freq_in, pll_id); if (ret) return ret; diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index 8ec14f5..0365177 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -58,4 +58,11 @@ #define WM8804_CLKOUT_DIV 1 +#define WM8804_MCLKDIV_DONTCARE 0 +#define WM8804_MCLKDIV_0 1 +#define WM8804_MCLKDIV_1 2 +#define WM8804_PLL_MCLKDIV_DONTCARE WM8804_MCLKDIV_DONTCARE +#define WM8804_PLL_MCLKDIV_0 WM8804_MCLKDIV_0 +#define WM8804_PLL_MCLKDIV_1 WM8804_MCLKDIV_1 + #endif /* _WM8804_H */ -- 1.7.9.5

4 5

[alsa-devel] [PATCH -next] ASoC: wm_adsp: Fix sparse non static symbol warning
by Wei Yongjun 09 Jan '14

09 Jan '14

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fixes the following sparse warning: sound/soc/codecs/wm_adsp.c:1500:6: warning: symbol 'wm_adsp2_boot_work' was not declared. Should it be static? Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 41669ad..444626f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1497,7 +1497,7 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) return 0; } -void wm_adsp2_boot_work(struct work_struct *work) +static void wm_adsp2_boot_work(struct work_struct *work) { struct wm_adsp *dsp = container_of(work, struct wm_adsp,

2 1

[alsa-devel] [PATCH v2 01/11] ASoC: dpcm: Add helper function for initializing runtime pcm
by Lars-Peter Clausen 09 Jan '14

09 Jan '14

We have the same code for initializing the runtime pcm on both the playback and the capture path. Factor this out into a common helper function. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Jean-Francois Moine <moinejf(a)free.fr> --- New in v2 --- sound/soc/soc-pcm.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 604e7e9..898149e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1235,6 +1235,17 @@ unwind: return err; } +static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, + struct snd_soc_pcm_stream *stream) +{ + runtime->hw.rate_min = stream->rate_min; + runtime->hw.rate_max = stream->rate_max; + runtime->hw.channels_min = stream->channels_min; + runtime->hw.channels_max = stream->channels_max; + runtime->hw.formats &= stream->formats; + runtime->hw.rates = stream->rates; +} + static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -1242,21 +1253,10 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - runtime->hw.rate_min = cpu_dai_drv->playback.rate_min; - runtime->hw.rate_max = cpu_dai_drv->playback.rate_max; - runtime->hw.channels_min = cpu_dai_drv->playback.channels_min; - runtime->hw.channels_max = cpu_dai_drv->playback.channels_max; - runtime->hw.formats &= cpu_dai_drv->playback.formats; - runtime->hw.rates = cpu_dai_drv->playback.rates; - } else { - runtime->hw.rate_min = cpu_dai_drv->capture.rate_min; - runtime->hw.rate_max = cpu_dai_drv->capture.rate_max; - runtime->hw.channels_min = cpu_dai_drv->capture.channels_min; - runtime->hw.channels_max = cpu_dai_drv->capture.channels_max; - runtime->hw.formats &= cpu_dai_drv->capture.formats; - runtime->hw.rates = cpu_dai_drv->capture.rates; - } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback); + else + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture); } static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) -- 1.8.0

3 21

[alsa-devel] [PATCH 1/5] ASoC: ad1836: Use params_width() rather than explicit memory format
by Mark Brown 09 Jan '14

09 Jan '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ad1836.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index d7c983862cf0..77f459868579 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -168,15 +168,15 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, int word_len = 0; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: word_len = AD1836_WORD_LEN_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: word_len = AD1836_WORD_LEN_20; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 24: + case 32: word_len = AD1836_WORD_LEN_24; break; default: -- 1.8.5.2

2 6

[alsa-devel] [PATCH] ASoC: wm_adsp: Mark wm_adsp2_boot_work as static
by Charles Keepax 09 Jan '14

09 Jan '14

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 41669ad..444626f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1497,7 +1497,7 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) return 0; } -void wm_adsp2_boot_work(struct work_struct *work) +static void wm_adsp2_boot_work(struct work_struct *work) { struct wm_adsp *dsp = container_of(work, struct wm_adsp, -- 1.7.2.5

2 1

[alsa-devel] [PATCH 1/2] mfd: wm5110: Add registers for headphone short circuit control
by Charles Keepax 09 Jan '14

09 Jan '14

Add the registers necessary to enable/disable the headphone short circuit protection. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5110-tables.c | 6 ++++++ include/linux/mfd/arizona/registers.h | 27 +++++++++++++++++++++++++++ 2 files changed, 33 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 24ec464..30f8163 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -637,6 +637,9 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000491, 0x0000 }, /* R1169 - PDM SPK1 CTRL 2 */ { 0x00000492, 0x0069 }, /* R1170 - PDM SPK2 CTRL 1 */ { 0x00000493, 0x0000 }, /* R1171 - PDM SPK2 CTRL 2 */ + { 0x000004A0, 0x3480 }, /* R1184 - HP1 Short Circuit Ctrl */ + { 0x000004A1, 0x3480 }, /* R1185 - HP2 Short Circuit Ctrl */ + { 0x000004A2, 0x3480 }, /* R1186 - HP3 Short Circuit Ctrl */ { 0x00000500, 0x000C }, /* R1280 - AIF1 BCLK Ctrl */ { 0x00000501, 0x0008 }, /* R1281 - AIF1 Tx Pin Ctrl */ { 0x00000502, 0x0000 }, /* R1282 - AIF1 Rx Pin Ctrl */ @@ -1633,6 +1636,9 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_PDM_SPK1_CTRL_2: case ARIZONA_PDM_SPK2_CTRL_1: case ARIZONA_PDM_SPK2_CTRL_2: + case ARIZONA_HP1_SHORT_CIRCUIT_CTRL: + case ARIZONA_HP2_SHORT_CIRCUIT_CTRL: + case ARIZONA_HP3_SHORT_CIRCUIT_CTRL: case ARIZONA_AIF1_BCLK_CTRL: case ARIZONA_AIF1_TX_PIN_CTRL: case ARIZONA_AIF1_RX_PIN_CTRL: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index b40960d..2d3ac50 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -226,6 +226,9 @@ #define ARIZONA_PDM_SPK1_CTRL_2 0x491 #define ARIZONA_PDM_SPK2_CTRL_1 0x492 #define ARIZONA_PDM_SPK2_CTRL_2 0x493 +#define ARIZONA_HP1_SHORT_CIRCUIT_CTRL 0x4A0 +#define ARIZONA_HP2_SHORT_CIRCUIT_CTRL 0x4A1 +#define ARIZONA_HP3_SHORT_CIRCUIT_CTRL 0x4A2 #define ARIZONA_SPK_CTRL_2 0x4B5 #define ARIZONA_SPK_CTRL_3 0x4B6 #define ARIZONA_DAC_COMP_1 0x4DC @@ -3297,6 +3300,30 @@ #define ARIZONA_SPK2_FMT_WIDTH 1 /* SPK2_FMT */ /* + * R1184 (0x4A0) - HP1 Short Circuit Ctrl + */ +#define ARIZONA_HP1_SC_ENA 0x1000 /* HP1_SC_ENA */ +#define ARIZONA_HP1_SC_ENA_MASK 0x1000 /* HP1_SC_ENA */ +#define ARIZONA_HP1_SC_ENA_SHIFT 12 /* HP1_SC_ENA */ +#define ARIZONA_HP1_SC_ENA_WIDTH 1 /* HP1_SC_ENA */ + +/* + * R1185 (0x4A1) - HP2 Short Circuit Ctrl + */ +#define ARIZONA_HP2_SC_ENA 0x1000 /* HP2_SC_ENA */ +#define ARIZONA_HP2_SC_ENA_MASK 0x1000 /* HP2_SC_ENA */ +#define ARIZONA_HP2_SC_ENA_SHIFT 12 /* HP2_SC_ENA */ +#define ARIZONA_HP2_SC_ENA_WIDTH 1 /* HP2_SC_ENA */ + +/* + * R1186 (0x4A2) - HP3 Short Circuit Ctrl + */ +#define ARIZONA_HP3_SC_ENA 0x1000 /* HP3_SC_ENA */ +#define ARIZONA_HP3_SC_ENA_MASK 0x1000 /* HP3_SC_ENA */ +#define ARIZONA_HP3_SC_ENA_SHIFT 12 /* HP3_SC_ENA */ +#define ARIZONA_HP3_SC_ENA_WIDTH 1 /* HP3_SC_ENA */ + +/* * R1244 (0x4DC) - DAC comp 1 */ #define ARIZONA_OUT_COMP_COEFF_MASK 0xFFFF /* OUT_COMP_COEFF - [15:0] */ -- 1.7.2.5

2 2

[alsa-devel] [PATCH] ASoC: bcm: Remove obsoleted Kconfig dependency
by Takashi Iwai 09 Jan '14

09 Jan '14

CONFIG_SND_SOC_DMAENGINE_PCM was renamed to CONFIG_SND_DMAENGINE_PCM recently. And yet we don't have to select it since CONFIG_SND_GENERIC_DMAENGINE_PCM selects the dependency by itself, so just rip it off. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/bcm/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig index 3d82a29ce3a8..6a834e109f1d 100644 --- a/sound/soc/bcm/Kconfig +++ b/sound/soc/bcm/Kconfig @@ -1,7 +1,6 @@ config SND_BCM2835_SOC_I2S tristate "SoC Audio support for the Broadcom BCM2835 I2S module" depends on ARCH_BCM2835 || COMPILE_TEST - select SND_SOC_DMAENGINE_PCM select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help -- 1.8.5.2

3 8

[alsa-devel] [PATCH] ASoC: add RT5670 CODEC driver
by bardliao＠realtek.com 09 Jan '14

09 Jan '14

From: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- This is the initial codec driver of ALC5670 I have tested playback to Headphone, Line out and PDM interface. But not any capture path. --- include/sound/rt5670.h | 20 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5670-dsp.c | 817 +++++++++++ sound/soc/codecs/rt5670-dsp.h | 69 + sound/soc/codecs/rt5670.c | 3053 +++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5670.h | 1966 ++++++++++++++++++++++++++ 7 files changed, 5931 insertions(+) create mode 100644 include/sound/rt5670.h create mode 100644 sound/soc/codecs/rt5670-dsp.c create mode 100644 sound/soc/codecs/rt5670-dsp.h create mode 100644 sound/soc/codecs/rt5670.c create mode 100644 sound/soc/codecs/rt5670.h diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h new file mode 100644 index 0000000..2397fb8 --- /dev/null +++ b/include/sound/rt5670.h @@ -0,0 +1,20 @@ +/* + * linux/sound/rt5670.h -- Platform data for RT5670 + * + * Copyright 2011 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT5670_H +#define __LINUX_SND_RT5670_H + +struct rt5670_platform_data { + bool asrc_en; + bool jd1_en; + bool in2_diff; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..4400433 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -61,6 +61,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C + select SND_SOC_RT5670 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -319,6 +320,9 @@ config SND_SOC_RT5631 config SND_SOC_RT5640 tristate +config SND_SOC_RT5670 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc12676..f95e65a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -48,6 +48,7 @@ snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o +snd-soc-rt5670-objs := rt5670.o rt5670-dsp.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -181,6 +182,7 @@ obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o +obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o diff --git a/sound/soc/codecs/rt5670-dsp.c b/sound/soc/codecs/rt5670-dsp.c new file mode 100644 index 0000000..ed01fa6 --- /dev/null +++ b/sound/soc/codecs/rt5670-dsp.c @@ -0,0 +1,817 @@ +/* + * rt5670.c -- RT5670 ALSA SoC DSP driver + * + * Copyright 2011 Realtek Semiconductor Corp. + * Author: Johnny Hsu <johnnyhsu(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "rt5670.h" +#include "rt5670-dsp.h" + +#define DSP_CLK_RATE RT5670_DSP_CLK_96K + +/* DSP init */ +static unsigned short rt5670_dsp_init[][2] = { + {0x2260, 0x30d9}, {0x2261, 0x30d9}, {0x2289, 0x7fff}, {0x2290, 0x7fff}, + {0x2288, 0x0002}, {0x22b2, 0x0002}, {0x2295, 0x0001}, {0x22b3, 0x0001}, + {0x22d7, 0x0008}, {0x22d8, 0x0009}, {0x22d9, 0x0000}, {0x22da, 0x0001}, + {0x22fd, 0x009e}, {0x22c1, 0x1006}, {0x22c2, 0x1006}, {0x22c3, 0x1007}, + {0x22c4, 0x1007} +}; +#define RT5670_DSP_INIT_NUM \ + (sizeof(rt5670_dsp_init) / sizeof(rt5670_dsp_init[0])) + +/* MCLK rate */ +static unsigned short rt5670_dsp_4096000[][2] = { + {0x226c, 0x000c}, {0x226d, 0x000c}, {0x226e, 0x0022}, +}; +#define RT5670_DSP_4096000_NUM \ + (sizeof(rt5670_dsp_4096000) / sizeof(rt5670_dsp_4096000[0])) + +static unsigned short rt5670_dsp_12288000[][2] = { + {0x226c, 0x000c}, {0x226d, 0x000c}, {0x226e, 0x0026}, +}; +#define RT5670_DSP_12288000_NUM \ + (sizeof(rt5670_dsp_12288000) / sizeof(rt5670_dsp_12288000[0])) + +static unsigned short rt5670_dsp_11289600[][2] = { + {0x226c, 0x0031}, {0x226d, 0x0050}, {0x226e, 0x0009}, +}; +#define RT5670_DSP_11289600_NUM \ + (sizeof(rt5670_dsp_11289600) / sizeof(rt5670_dsp_11289600[0])) + +static unsigned short rt5670_dsp_24576000[][2] = { + {0x226c, 0x000c}, {0x226d, 0x000c}, {0x226e, 0x002c}, +}; +#define RT5670_DSP_24576000_NUM \ + (sizeof(rt5670_dsp_24576000) / sizeof(rt5670_dsp_24576000[0])) + +/* sample rate */ +static unsigned short rt5670_dsp_48_441[][2] = { + {0x22f2, 0x0058}, {0x2301, 0x0016} +}; +#define RT5670_DSP_48_441_NUM \ + (sizeof(rt5670_dsp_48_441) / sizeof(rt5670_dsp_48_441[0])) + +static unsigned short rt5670_dsp_24[][2] = { + {0x22f2, 0x0058}, {0x2301, 0x0004} +}; +#define RT5670_DSP_24_NUM (sizeof(rt5670_dsp_24) / sizeof(rt5670_dsp_24[0])) + +static unsigned short rt5670_dsp_16[][2] = { + {0x22f2, 0x004c}, {0x2301, 0x0002} +}; +#define RT5670_DSP_16_NUM (sizeof(rt5670_dsp_16) / sizeof(rt5670_dsp_16[0])) + +static unsigned short rt5670_dsp_8[][2] = { + {0x22f2, 0x004c}, {0x2301, 0x0000} +}; +#define RT5670_DSP_8_NUM (sizeof(rt5670_dsp_8) / sizeof(rt5670_dsp_8[0])) + +/* DSP mode */ +static unsigned short rt5670_dsp_ns[][2] = { + {0x22f8, 0x8005}, {0x2375, 0x7ff0}, {0x2376, 0x7990}, {0x2377, 0x7332}, + {0x2388, 0x7fff}, {0x2389, 0x6000}, {0x238a, 0x0000}, {0x238b, 0x1000}, + {0x238c, 0x1000}, {0x23a1, 0x2000}, {0x2303, 0x0200}, {0x2304, 0x0032}, + {0x2305, 0x0000}, {0x230c, 0x0200}, {0x22fb, 0x0000} +}; +#define RT5670_DSP_NS_NUM \ + (sizeof(rt5670_dsp_ns) / sizeof(rt5670_dsp_ns[0])) + +static unsigned short rt5670_dsp_aec[][2] = { + {0x22f8, 0x8003}, {0x232f, 0x00d0}, {0x2355, 0x2666}, {0x2356, 0x2666}, + {0x2357, 0x2666}, {0x2358, 0x6666}, {0x2359, 0x6666}, {0x235a, 0x6666}, + {0x235b, 0x7fff}, {0x235c, 0x7fff}, {0x235d, 0x7fff}, {0x235e, 0x7fff}, + {0x235f, 0x7fff}, {0x2360, 0x7fff}, {0x2361, 0x7fff}, {0x2362, 0x1000}, + {0x2367, 0x0007}, {0x2368, 0x4000}, {0x2369, 0x0008}, {0x236a, 0x2000}, + {0x236b, 0x0009}, {0x236c, 0x003c}, {0x236d, 0x0000}, {0x236f, 0x2000}, + {0x2370, 0x0008}, {0x2371, 0x000a}, {0x2373, 0x0000}, {0x2374, 0x7fff}, + {0x2375, 0x7ff0}, {0x2376, 0x7990}, {0x2377, 0x7332}, {0x2379, 0x1000}, + {0x237a, 0x1000}, {0x2388, 0x7fff}, {0x2389, 0x6000}, {0x238a, 0x4000}, + {0x238b, 0x1000}, {0x238c, 0x1000}, {0x2398, 0x4668}, {0x23a1, 0x2000}, + {0x23a3, 0x4000}, {0x23ad, 0x2000}, {0x23ae, 0x2000}, {0x23af, 0x2000}, + {0x23b4, 0x2000}, {0x23b5, 0x2000}, {0x23b6, 0x2000}, {0x23bb, 0x6000}, + {0x2303, 0x0710}, {0x2304, 0x0332}, {0x230c, 0x0200}, {0x230d, 0x0080}, + {0x2310, 0x0010}, {0x22fb, 0x0000} +}; +#define RT5670_DSP_AEC_NUM \ + (sizeof(rt5670_dsp_aec) / sizeof(rt5670_dsp_aec[0])) + +static unsigned short rt5670_dsp_vt[][2] = { + {0x22f8, 0x8003}, {0x2371, 0x000a}, {0x2373, 0x0000}, {0x2374, 0x7fff}, + {0x2375, 0x7ff0}, {0x2376, 0x7990}, {0x2377, 0x7332}, {0x2379, 0x1000}, + {0x237a, 0x1000}, {0x2388, 0x7fff}, {0x2389, 0x6000}, {0x238b, 0x1000}, + {0x238c, 0x1000}, {0x23a1, 0x2000}, {0x2304, 0x0332}, {0x230c, 0x0200}, + {0x22fb, 0x0000} +}; +#define RT5670_DSP_VT_NUM (sizeof(rt5670_dsp_vt) / sizeof(rt5670_dsp_vt[0])) + +static unsigned short rt5670_dsp_vr[][2] = { + {0x22f8, 0x8003}, {0x2304, 0x0332}, {0x2305, 0x0000}, {0x2309, 0x0400}, + {0x230c, 0x0200}, {0x230d, 0x0080}, {0x2310, 0x0004}, {0x232f, 0x0100}, + {0x2371, 0x000a}, {0x2373, 0x3000}, {0x2374, 0x5000}, {0x2375, 0x7ff0}, + {0x2376, 0x7990}, {0x2377, 0x7332}, {0x2379, 0x1000}, {0x237a, 0x1000}, + {0x2386, 0x0200}, {0x2388, 0x4000}, {0x2389, 0x4000}, {0x238a, 0x0000}, + {0x238b, 0x2000}, {0x238c, 0x1800}, {0x239b, 0x0000}, {0x239c, 0x0a00}, + {0x239d, 0x0000}, {0x239e, 0x7fff}, {0x239f, 0x0001}, {0x23a1, 0x3000}, + {0x23a2, 0x1000}, {0x23ad, 0x0000}, {0x23ae, 0x0000}, {0x23af, 0x0000}, + {0x23c4, 0x2000}, {0x23b0, 0x0000}, {0x23b1, 0x0000}, {0x23b2, 0x0000}, + {0x23b3, 0x0000}, {0x23b4, 0x0000}, {0x23b5, 0x0000}, {0x23b6, 0x0000}, + {0x23b7, 0x0000}, {0x23b8, 0x0000}, {0x23b9, 0x0000}, {0x23ba, 0x0000}, + {0x23bb, 0x0000}, {0x22fb, 0x0000} +}; +#define RT5670_DSP_VR_NUM (sizeof(rt5670_dsp_vr) / sizeof(rt5670_dsp_vr[0])) + +static unsigned short rt5670_dsp_ffp_ns[][2] = { + {0x22f8, 0x8003}, {0x2304, 0x8332}, {0x2371, 0x000a}, {0x2373, 0x0000}, + {0x2374, 0x7fff}, {0x2379, 0x1800}, {0x237a, 0x1800}, {0x230c, 0x0200}, + {0x23a2, 0x1000}, {0x2388, 0x7000}, {0x238b, 0x2000}, {0x238c, 0x2000}, + {0x23a8, 0x2000}, {0x23a9, 0x4000}, {0x23aa, 0x0100}, {0x23ab, 0x7800}, + {0x22fb, 0x0000} +}; +#define RT5670_DSP_FFP_NS_NUM \ + (sizeof(rt5670_dsp_ffp_ns) / sizeof(rt5670_dsp_ffp_ns[0])) + +static unsigned short rt5670_dsp_48k_sto_ffp[][2] = { + {0x22c1, 0x1025}, {0x22c2, 0x1026}, {0x22f8, 0x8004}, {0x22ea, 0x0001}, + {0x230c, 0x0100}, {0x230d, 0x0100}, {0x2301, 0x0010}, {0x2303, 0x0200}, + {0x2304, 0x8000}, {0x2305, 0x0000}, {0x2388, 0x6500}, {0x238b, 0x4000}, + {0x238c, 0x4000}, {0x23a9, 0x2000}, {0x23aa, 0x0200}, {0x23ab, 0x7c00}, + {0x22fb, 0x0000} +}; +#define RT5670_DSP_48K_STO_FFP_NUM \ + (sizeof(rt5670_dsp_48k_sto_ffp) / sizeof(rt5670_dsp_48k_sto_ffp[0])) + +static unsigned short rt5670_dsp_2mic_handset[][2] = { + {0x22f8, 0x8002}, {0x2301, 0x0002}, {0x2302, 0x0002}, {0x2303, 0x0710}, + {0x2304, 0x4332}, {0x2305, 0x206c}, {0x236e, 0x0000}, {0x236f, 0x0001}, + {0x237e, 0x0001}, {0x237f, 0x3800}, {0x2380, 0x3000}, {0x2381, 0x0005}, + {0x2382, 0x0040}, {0x2383, 0x7fff}, {0x2388, 0x2c00}, {0x2389, 0x2800}, + {0x238b, 0x1800}, {0x238c, 0x1800}, {0x238f, 0x2000}, {0x239b, 0x0002}, + {0x239c, 0x0a00}, {0x239f, 0x0001}, {0x230c, 0x0200}, {0x22fb, 0x0000} +}; +#define RT5670_DSP_2MIC_HANDSET_NUM \ + (sizeof(rt5670_dsp_2mic_handset) / sizeof(rt5670_dsp_2mic_handset[0])) + +static unsigned short rt5670_dsp_2mic_handsfree[][2] = { + {0x22f8, 0x8003}, {0x2371, 0x000a}, {0x2373, 0x0000}, {0x2374, 0x7fff}, + {0x2375, 0x7ff0}, {0x2376, 0x7990}, {0x2377, 0x7332}, {0x2379, 0x1000}, + {0x237a, 0x1000}, {0x2388, 0x7fff}, {0x2389, 0x6000}, {0x238b, 0x1000}, + {0x238c, 0x1000}, {0x23a1, 0x2000}, {0x2304, 0x0332}, {0x230c, 0x0200}, + {0x22fb, 0x0000} +}; +#define RT5670_DSP_2MIC_HANDSFREE_NUM \ + (sizeof(rt5670_dsp_2mic_handsfree) / \ + sizeof(rt5670_dsp_2mic_handsfree[0])) + +static unsigned short rt5670_dsp_aec_handsfree[][2] = { + {0x22f8, 0x8003}, {0x232f, 0x00d0}, {0x2355, 0x2666}, {0x2356, 0x2666}, + {0x2357, 0x2666}, {0x2358, 0x6666}, {0x2359, 0x6666}, {0x235a, 0x6666}, + {0x235b, 0x7fff}, {0x235c, 0x7fff}, {0x235d, 0x7fff}, {0x235e, 0x7fff}, + {0x235f, 0x7fff}, {0x2360, 0x7fff}, {0x2361, 0x7fff}, {0x2362, 0x1000}, + {0x2367, 0x0007}, {0x2368, 0x4000}, {0x2369, 0x0008}, {0x236a, 0x2000}, + {0x236b, 0x0009}, {0x236c, 0x003c}, {0x236d, 0x0000}, {0x236f, 0x2000}, + {0x2370, 0x0008}, {0x2371, 0x000a}, {0x2373, 0x0000}, {0x2374, 0x7fff}, + {0x2375, 0x7ff0}, {0x2376, 0x7990}, {0x2377, 0x7332}, {0x2379, 0x1000}, + {0x237a, 0x1000}, {0x2388, 0x7fff}, {0x2389, 0x6000}, {0x238a, 0x4000}, + {0x238b, 0x1000}, {0x238c, 0x1000}, {0x2398, 0x4668}, {0x23a1, 0x2000}, + {0x23a3, 0x4000}, {0x23ad, 0x2000}, {0x23ae, 0x2000}, {0x23af, 0x2000}, + {0x23b4, 0x2000}, {0x23b5, 0x2000}, {0x23b6, 0x2000}, {0x23bb, 0x6000}, + {0x2303, 0x0710}, {0x2304, 0x0332}, {0x230c, 0x0200}, {0x230d, 0x0080}, + {0x2310, 0x0010}, {0x22fb, 0x0000} +}; +#define RT5670_DSP_AEC_HANDSFREE_NUM \ + (sizeof(rt5670_dsp_aec_handsfree) / sizeof(rt5670_dsp_aec_handsfree[0])) + +/** + * rt5670_dsp_done - Wait until DSP is ready. + * @codec: SoC Audio Codec device. + * + * To check voice DSP status and confirm it's ready for next work. + * + * Returns 0 for success or negative error code. + */ +static int rt5670_dsp_done(struct snd_soc_codec *codec) +{ + unsigned int count = 0, dsp_val; + + dsp_val = snd_soc_read(codec, RT5670_DSP_CTRL1); + while (dsp_val & RT5670_DSP_BUSY_MASK) { + if (count > 10) + return -EBUSY; + dsp_val = snd_soc_read(codec, RT5670_DSP_CTRL1); + count++; + } + + return 0; +} + +/** + * rt5670_dsp_write - Write DSP register. + * @codec: SoC audio codec device. + * @param: DSP parameters. + * + * Modify voice DSP register for sound effect. The DSP can be controlled + * through DSP command format (0xfc), addr (0xc4), data (0xc5) and cmd (0xc6) + * register. It has to wait until the DSP is ready. + * + * Returns 0 for success or negative error code. + */ +static int rt5670_dsp_write(struct snd_soc_codec *codec, + unsigned int addr, unsigned int data) +{ + unsigned int dsp_val; + int ret; + + ret = snd_soc_write(codec, RT5670_DSP_CTRL2, addr); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP addr reg: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_DSP_CTRL3, data); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP data reg: %d\n", ret); + goto err; + } + dsp_val = RT5670_DSP_I2C_AL_16 | RT5670_DSP_DL_2 | + RT5670_DSP_CMD_MW | DSP_CLK_RATE | RT5670_DSP_CMD_EN; + + ret = snd_soc_write(codec, RT5670_DSP_CTRL1, dsp_val); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP cmd reg: %d\n", ret); + goto err; + } + ret = rt5670_dsp_done(codec); + if (ret < 0) { + dev_err(codec->dev, "DSP is busy: %d\n", ret); + goto err; + } + + return 0; + +err: + return ret; +} + +/** + * rt5670_dsp_read - Read DSP register. + * @codec: SoC audio codec device. + * @reg: DSP register index. + * + * Read DSP setting value from voice DSP. The DSP can be controlled + * through DSP addr (0xc4), data (0xc5) and cmd (0xc6) register. Each + * command has to wait until the DSP is ready. + * + * Returns DSP register value or negative error code. + */ +static unsigned int rt5670_dsp_read( + struct snd_soc_codec *codec, unsigned int reg) +{ + unsigned int value; + unsigned int dsp_val; + int ret = 0; + + ret = rt5670_dsp_done(codec); + if (ret < 0) { + dev_err(codec->dev, "DSP is busy: %d\n", ret); + goto err; + } + + ret = snd_soc_write(codec, RT5670_DSP_CTRL2, reg); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP addr reg: %d\n", ret); + goto err; + } + dsp_val = RT5670_DSP_I2C_AL_16 | RT5670_DSP_DL_0 | RT5670_DSP_RW_MASK | + RT5670_DSP_CMD_MR | DSP_CLK_RATE | RT5670_DSP_CMD_EN; + + ret = snd_soc_write(codec, RT5670_DSP_CTRL1, dsp_val); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP cmd reg: %d\n", ret); + goto err; + } + + ret = rt5670_dsp_done(codec); + if (ret < 0) { + dev_err(codec->dev, "DSP is busy: %d\n", ret); + goto err; + } + + ret = snd_soc_write(codec, RT5670_DSP_CTRL2, 0x26); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP addr reg: %d\n", ret); + goto err; + } + dsp_val = RT5670_DSP_DL_1 | RT5670_DSP_CMD_RR | RT5670_DSP_RW_MASK | + DSP_CLK_RATE | RT5670_DSP_CMD_EN; + + ret = snd_soc_write(codec, RT5670_DSP_CTRL1, dsp_val); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP cmd reg: %d\n", ret); + goto err; + } + + ret = rt5670_dsp_done(codec); + if (ret < 0) { + dev_err(codec->dev, "DSP is busy: %d\n", ret); + goto err; + } + + ret = snd_soc_write(codec, RT5670_DSP_CTRL2, 0x25); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP addr reg: %d\n", ret); + goto err; + } + + dsp_val = RT5670_DSP_DL_1 | RT5670_DSP_CMD_RR | RT5670_DSP_RW_MASK | + DSP_CLK_RATE | RT5670_DSP_CMD_EN; + + ret = snd_soc_write(codec, RT5670_DSP_CTRL1, dsp_val); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP cmd reg: %d\n", ret); + goto err; + } + + ret = rt5670_dsp_done(codec); + if (ret < 0) { + dev_err(codec->dev, "DSP is busy: %d\n", ret); + goto err; + } + + ret = snd_soc_read(codec, RT5670_DSP_CTRL5); + if (ret < 0) { + dev_err(codec->dev, "Failed to read DSP data reg: %d\n", ret); + goto err; + } + + value = ret; + return value; + +err: + return ret; +} + +static int rt5670_dsp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = rt5670->dsp_sw; + + return 0; +} + +static int rt5670_dsp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + if (rt5670->dsp_sw != ucontrol->value.integer.value[0]) + rt5670->dsp_sw = ucontrol->value.integer.value[0]; + + return 0; +} + +static int rt5670_dsp_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + switch (rt5670->dsp_rate) { + case 8000: + ucontrol->value.integer.value[0] = 0; + break; + case 16000: + ucontrol->value.integer.value[0] = 1; + break; + case 24000: + ucontrol->value.integer.value[0] = 2; + break; + case 44100: + ucontrol->value.integer.value[0] = 3; + break; + case 48000: + ucontrol->value.integer.value[0] = 4; + break; + default: + ucontrol->value.integer.value[0] = 1; + break; + } + + return 0; +} + +static int rt5670_dsp_rate_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + switch (ucontrol->value.integer.value[0]) { + case 0: + rt5670->dsp_rate = 8000; + break; + case 1: + rt5670->dsp_rate = 16000; + break; + case 2: + rt5670->dsp_rate = 24000; + break; + case 3: + rt5670->dsp_rate = 44100; + break; + case 4: + rt5670->dsp_rate = 48000; + break; + default: + pr_err("unsupported dsp rate, assume it is 16 KHz\n"); + rt5670->dsp_rate = 16000; + break; + } + + return 0; +} + +/* DSP Path Control 1 */ +static const char * const rt5670_src_rxdp_mode[] = { + "Normal", "Divided by 2", "Divided by 3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_src_rxdp_enum, RT5670_DSP_PATH1, + RT5670_RXDP_SRC_SFT, rt5670_src_rxdp_mode); + +static const char * const rt5670_src_txdp_mode[] = { + "Normal", "Multiplied by 2", "Multiplied by 3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_src_txdp_enum, RT5670_DSP_PATH1, + RT5670_TXDP_SRC_SFT, rt5670_src_txdp_mode); + +/* Sound Effect */ +static const char * const rt5670_dsp_mode[] = { + "Disable", "NS", "AEC", "VT", "VR", "FFP+NS", "48K-stereo+FFP", + "2MIC Handset", "2MIC Handsfree", "AEC Handsfree" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5670_dsp_enum, SND_SOC_NOPM, 0, + rt5670_dsp_mode); + +static const char * const rt5670_dsp_rate_mode[] = { + "8K", "16K", "24K", "44.1K", "48K" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5670_dsp_rate_enum, SND_SOC_NOPM, 0, + rt5670_dsp_rate_mode); + +static const struct snd_kcontrol_new rt5670_dsp_snd_controls[] = { + SOC_ENUM("SRC for RxDP", rt5670_src_rxdp_enum), + SOC_ENUM("SRC for TxDP", rt5670_src_txdp_enum), + SOC_ENUM_EXT("DSP Function Switch", rt5670_dsp_enum, + rt5670_dsp_get, rt5670_dsp_put), + SOC_ENUM_EXT("DSP Rate", rt5670_dsp_rate_enum, + rt5670_dsp_rate_get, rt5670_dsp_rate_put), +}; + +/** + * rt5670_dsp_conf - Set DSP basic setting. + * + * @codec: SoC audio codec device. + * + * Set parameters of basic setting to DSP. + * + * Returns 0 for success or negative error code. + */ +static int rt5670_dsp_conf(struct snd_soc_codec *codec) +{ + int ret, i; + + for (i = 0; i < RT5670_DSP_INIT_NUM; i++) { + ret = rt5670_dsp_write(codec, rt5670_dsp_init[i][0], + rt5670_dsp_init[i][1]); + if (ret < 0) { + dev_err(codec->dev, "Fail to config Dsp: %d\n", ret); + goto conf_err; + } + } + + return 0; + +conf_err: + + return ret; +} + +/** + * rt5670_dsp_rate - Set DSP rate setting. + * + * @codec: SoC audio codec device. + * @sys_clk: System clock. + * @srate: Sampling rate. + * + * Set parameters of system clock and sampling rate to DSP. + * + * Returns 0 for success or negative error code. + */ +static int rt5670_dsp_rate(struct snd_soc_codec *codec, int sys_clk, + int srate) +{ + int ret, i, tab_num; + unsigned short (*rate_tab)[2]; + + dev_dbg(codec->dev, "%s sys:%d srate:%d\n", __func__, sys_clk, srate); + + switch (sys_clk) { + case 4096000: + rate_tab = rt5670_dsp_4096000; + tab_num = RT5670_DSP_4096000_NUM; + break; + case 11289600: + rate_tab = rt5670_dsp_11289600; + tab_num = RT5670_DSP_11289600_NUM; + break; + case 12288000: + rate_tab = rt5670_dsp_12288000; + tab_num = RT5670_DSP_12288000_NUM; + break; + case 24576000: + rate_tab = rt5670_dsp_24576000; + tab_num = RT5670_DSP_24576000_NUM; + break; + default: + return -EINVAL; + break; + } + + for (i = 0; i < tab_num; i++) { + ret = rt5670_dsp_write(codec, rate_tab[i][0], rate_tab[i][1]); + if (ret < 0) + goto rate_err; + } + + switch (srate) { + case 8000: + rate_tab = rt5670_dsp_8; + tab_num = RT5670_DSP_8_NUM; + break; + case 16000: + rate_tab = rt5670_dsp_16; + tab_num = RT5670_DSP_16_NUM; + break; + case 24000: + rate_tab = rt5670_dsp_24; + tab_num = RT5670_DSP_24_NUM; + break; + case 44100: + case 48000: + rate_tab = rt5670_dsp_48_441; + tab_num = RT5670_DSP_48_441_NUM; + break; + default: + return -EINVAL; + break; + } + + for (i = 0; i < tab_num; i++) { + ret = rt5670_dsp_write(codec, rate_tab[i][0], rate_tab[i][1]); + if (ret < 0) + goto rate_err; + } + + return 0; + +rate_err: + + dev_err(codec->dev, "Fail to set rate parameters: %d\n", ret); + return ret; +} + +/** + * rt5670_dsp_set_mode - Set DSP mode parameters. + * + * @codec: SoC audio codec device. + * @mode: DSP mode. + * + * Set parameters of mode to DSP. + * There are three modes which includes " mic AEC + NS + FENS", + * "HFBF" and "Far-field pickup". + * + * Returns 0 for success or negative error code. + */ +static int rt5670_dsp_set_mode(struct snd_soc_codec *codec, int mode) +{ + int ret, i, tab_num; + unsigned short (*mode_tab)[2]; + + switch (mode) { + case RT5670_DSP_NS: + dev_info(codec->dev, "NS\n"); + mode_tab = rt5670_dsp_ns; + tab_num = RT5670_DSP_NS_NUM; + break; + + case RT5670_DSP_AEC: + dev_info(codec->dev, "AEC\n"); + mode_tab = rt5670_dsp_aec; + tab_num = RT5670_DSP_AEC_NUM; + break; + + case RT5670_DSP_VT: + dev_info(codec->dev, "VT\n"); + mode_tab = rt5670_dsp_vt; + tab_num = RT5670_DSP_VT_NUM; + break; + + case RT5670_DSP_VR: + dev_info(codec->dev, "VR\n"); + mode_tab = rt5670_dsp_vr; + tab_num = RT5670_DSP_VR_NUM; + break; + + case RT5670_DSP_FFP_NS: + dev_info(codec->dev, "FFP_NS\n"); + mode_tab = rt5670_dsp_ffp_ns; + tab_num = RT5670_DSP_FFP_NS_NUM; + break; + + case RT5670_DSP_48K_STO_FFP: + dev_info(codec->dev, "48K_STO_FFP\n"); + mode_tab = rt5670_dsp_48k_sto_ffp; + tab_num = RT5670_DSP_48K_STO_FFP_NUM; + break; + + case RT5670_DSP_2MIC_HANDSET: + dev_info(codec->dev, "2MIC_HANDSET\n"); + mode_tab = rt5670_dsp_2mic_handset; + tab_num = RT5670_DSP_2MIC_HANDSET_NUM; + break; + + case RT5670_DSP_2MIC_HANDSFREE: + dev_info(codec->dev, "2MIC_HANDSFREE\n"); + mode_tab = rt5670_dsp_2mic_handsfree; + tab_num = RT5670_DSP_2MIC_HANDSFREE_NUM; + break; + + case RT5670_DSP_AEC_HANDSFREE: + dev_info(codec->dev, "AEC_HANDSFREE\n"); + mode_tab = rt5670_dsp_aec_handsfree; + tab_num = RT5670_DSP_AEC_HANDSFREE_NUM; + break; + + case RT5670_DSP_DIS: + default: + dev_info(codec->dev, "Disable\n"); + return 0; + } + + for (i = 0; i < tab_num; i++) { + ret = rt5670_dsp_write(codec, mode_tab[i][0], mode_tab[i][1]); + if (ret < 0) + goto mode_err; + } + + return 0; + +mode_err: + + dev_err(codec->dev, "Fail to set mode %d parameters: %d\n", mode, ret); + return ret; +} + +/** + * rt5670_dsp_snd_effect - Set DSP sound effect. + * + * Set parameters of sound effect to DSP. + * + * Returns 0 for success or negative error code. + */ +static int rt5670_dsp_snd_effect(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + int ret; + + snd_soc_update_bits(codec, RT5670_DIG_MISC, RT5670_RST_DSP, + RT5670_RST_DSP); + snd_soc_update_bits(codec, RT5670_DIG_MISC, RT5670_RST_DSP, 0); + + mdelay(10); + + ret = rt5670_dsp_rate(codec, + rt5670->sysclk ? + rt5670->sysclk : 11289600, + rt5670->dsp_rate ? + rt5670->dsp_rate : 16000); + if (ret < 0) + goto effect_err; + + ret = rt5670_dsp_conf(codec); + if (ret < 0) + goto effect_err; + + ret = rt5670_dsp_set_mode(codec, rt5670->dsp_sw); + if (ret < 0) + goto effect_err; + + return 0; + +effect_err: + + return ret; +} + +static int rt5670_dsp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + dev_dbg(codec->dev, "%s(): PMD\n", __func__); + rt5670_dsp_write(codec, 0x22f9, 1); + break; + + case SND_SOC_DAPM_POST_PMU: + dev_dbg(codec->dev, "%s(): PMU\n", __func__); + rt5670_dsp_snd_effect(codec); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5670_dsp_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY_S("Voice DSP", 1, SND_SOC_NOPM, + 0, 0, rt5670_dsp_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA("DSP Downstream", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DSP Upstream", SND_SOC_NOPM, + 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route rt5670_dsp_dapm_routes[] = { + {"DSP Downstream", NULL, "Voice DSP"}, + {"DSP Downstream", NULL, "RxDP Mux"}, + {"DSP Upstream", NULL, "Voice DSP"}, + {"DSP Upstream", NULL, "8CH TDM Data"}, + {"DSP DL Mux", "DSP", "DSP Downstream"}, + {"DSP UL Mux", "DSP", "DSP Upstream"}, +}; + +/** + * rt5670_dsp_probe - register DSP for rt5670 + * @codec: audio codec + * + * To register DSP function for rt5670. + * + * Returns 0 for success or negative error code. + */ +int rt5670_dsp_probe(struct snd_soc_codec *codec) +{ + if (codec == NULL) + return -EINVAL; + + snd_soc_update_bits(codec, RT5670_PWR_DIG2, + RT5670_PWR_I2S_DSP, RT5670_PWR_I2S_DSP); + + snd_soc_update_bits(codec, RT5670_DIG_MISC, RT5670_RST_DSP, + RT5670_RST_DSP); + snd_soc_update_bits(codec, RT5670_DIG_MISC, RT5670_RST_DSP, 0); + + mdelay(10); + + rt5670_dsp_write(codec, 0x22fb, 0); + /* power down DSP*/ + rt5670_dsp_write(codec, 0x22f9, 1); + + snd_soc_update_bits(codec, RT5670_PWR_DIG2, + RT5670_PWR_I2S_DSP, 0); + + snd_soc_add_codec_controls(codec, rt5670_dsp_snd_controls, + ARRAY_SIZE(rt5670_dsp_snd_controls)); + snd_soc_dapm_new_controls(&codec->dapm, rt5670_dsp_dapm_widgets, + ARRAY_SIZE(rt5670_dsp_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, rt5670_dsp_dapm_routes, + ARRAY_SIZE(rt5670_dsp_dapm_routes)); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5670_dsp_probe); + diff --git a/sound/soc/codecs/rt5670-dsp.h b/sound/soc/codecs/rt5670-dsp.h new file mode 100644 index 0000000..784544c --- /dev/null +++ b/sound/soc/codecs/rt5670-dsp.h @@ -0,0 +1,69 @@ +/* + * rt5670-dsp.h -- RT5670 ALSA SoC DSP driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5670_DSP_H__ +#define __RT5670_DSP_H__ + +#define RT5670_DSP_CTRL1 0xe0 +#define RT5670_DSP_CTRL2 0xe1 +#define RT5670_DSP_CTRL3 0xe2 +#define RT5670_DSP_CTRL4 0xe3 +#define RT5670_DSP_CTRL5 0xe4 + +/* DSP Control 1 (0xe0) */ +#define RT5670_DSP_CMD_MASK (0xff << 8) +#define RT5670_DSP_CMD_PE (0x0d << 8) /* Patch Entry */ +#define RT5670_DSP_CMD_MW (0x3b << 8) /* Memory Write */ +#define RT5670_DSP_CMD_MR (0x37 << 8) /* Memory Read */ +#define RT5670_DSP_CMD_RR (0x60 << 8) /* Register Read */ +#define RT5670_DSP_CMD_RW (0x68 << 8) /* Register Write */ +#define RT5670_DSP_REG_DATHI (0x26 << 8) /* High Data Addr */ +#define RT5670_DSP_REG_DATLO (0x25 << 8) /* Low Data Addr */ +#define RT5670_DSP_CLK_MASK (0x3 << 6) +#define RT5670_DSP_CLK_SFT 6 +#define RT5670_DSP_CLK_768K (0x0 << 6) +#define RT5670_DSP_CLK_384K (0x1 << 6) +#define RT5670_DSP_CLK_192K (0x2 << 6) +#define RT5670_DSP_CLK_96K (0x3 << 6) +#define RT5670_DSP_BUSY_MASK (0x1 << 5) +#define RT5670_DSP_RW_MASK (0x1 << 4) +#define RT5670_DSP_DL_MASK (0x3 << 2) +#define RT5670_DSP_DL_0 (0x0 << 2) +#define RT5670_DSP_DL_1 (0x1 << 2) +#define RT5670_DSP_DL_2 (0x2 << 2) +#define RT5670_DSP_DL_3 (0x3 << 2) +#define RT5670_DSP_I2C_AL_16 (0x1 << 1) +#define RT5670_DSP_CMD_EN (0x1) + +enum { + RT5670_DSP_DIS, + RT5670_DSP_NS, + RT5670_DSP_AEC, + RT5670_DSP_VT, + RT5670_DSP_VR, + RT5670_DSP_FFP_NS, + RT5670_DSP_48K_STO_FFP, + RT5670_DSP_2MIC_HANDSET, + RT5670_DSP_2MIC_HANDSFREE, + RT5670_DSP_AEC_HANDSFREE, +}; + +struct rt5670_dsp_param { + u16 cmd_fmt; + u16 addr; + u16 data; + u8 cmd; +}; + +int rt5670_dsp_probe(struct snd_soc_codec *codec); + +#endif /* __RT5670_DSP_H__ */ + diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c new file mode 100644 index 0000000..b1a8517 --- /dev/null +++ b/sound/soc/codecs/rt5670.c @@ -0,0 +1,3053 @@ +/* + * rt5670.c -- RT5670 ALSA SoC audio codec driver + * + * Copyright 2012 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5670.h" +#include "rt5670-dsp.h" + +#define RT5670_DEVICE_ID 0x6271 + +#define RT5670_PR_RANGE_BASE (0xff + 1) +#define RT5670_PR_SPACING 0x100 + +#define RT5670_PR_BASE (RT5670_PR_RANGE_BASE + (0 * RT5670_PR_SPACING)) + +static const struct regmap_range_cfg rt5670_ranges[] = { + { .name = "PR", .range_min = RT5670_PR_BASE, + .range_max = RT5670_PR_BASE + 0xf8, + .selector_reg = RT5670_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5670_PRIV_DATA, + .window_len = 0x1, }, +}; + +static struct reg_default init_list[] = { + { RT5670_PR_BASE + 0x14, 0x9a8a }, + { RT5670_PR_BASE + 0x3d, 0x3640 }, +}; +#define RT5670_INIT_REG_LEN ARRAY_SIZE(init_list) + +static const struct reg_default rt5670_reg[] = { + { 0x00, 0x0000 }, + { 0x02, 0x8888 }, + { 0x03, 0x8888 }, + { 0x0a, 0x0001 }, + { 0x0b, 0x0827 }, + { 0x0c, 0x0000 }, + { 0x0d, 0x0008 }, + { 0x0e, 0x0000 }, + { 0x0f, 0x0808 }, + { 0x19, 0xafaf }, + { 0x1a, 0xafaf }, + { 0x1b, 0x0011 }, + { 0x1c, 0x2f2f }, + { 0x1d, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x1f, 0x2f2f }, + { 0x20, 0x0000 }, + { 0x26, 0x7860 }, + { 0x27, 0x7860 }, + { 0x28, 0x7871 }, + { 0x29, 0x8080 }, + { 0x2a, 0x5656 }, + { 0x2b, 0x5454 }, + { 0x2c, 0xaaa0 }, + { 0x2d, 0x0000 }, + { 0x2e, 0x2f2f }, + { 0x2f, 0x1002 }, + { 0x30, 0x0000 }, + { 0x31, 0x5f00 }, + { 0x32, 0x0000 }, + { 0x33, 0x0000 }, + { 0x34, 0x0000 }, + { 0x35, 0x0000 }, + { 0x36, 0x0000 }, + { 0x37, 0x0000 }, + { 0x38, 0x0000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x007f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x007f }, + { 0x45, 0xe00f }, + { 0x4c, 0x5380 }, + { 0x4f, 0x0073 }, + { 0x52, 0x00d3 }, + { 0x53, 0xf0f0 }, + { 0x61, 0x0000 }, + { 0x62, 0x0001 }, + { 0x63, 0x00c3 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x6f, 0x8000 }, + { 0x70, 0x8000 }, + { 0x71, 0x8000 }, + { 0x72, 0x8000 }, + { 0x73, 0x1110 }, + { 0x74, 0x0e00 }, + { 0x75, 0x1505 }, + { 0x76, 0x0015 }, + { 0x77, 0x0c00 }, + { 0x78, 0x4000 }, + { 0x79, 0x0123 }, + { 0x7f, 0x1100 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x83, 0x0000 }, + { 0x84, 0x0000 }, + { 0x85, 0x0000 }, + { 0x86, 0x0008 }, + { 0x87, 0x0000 }, + { 0x88, 0x0000 }, + { 0x89, 0x0000 }, + { 0x8a, 0x0000 }, + { 0x8b, 0x0000 }, + { 0x8c, 0x0007 }, + { 0x8d, 0x0000 }, + { 0x8e, 0x0004 }, + { 0x8f, 0x1100 }, + { 0x90, 0x0646 }, + { 0x91, 0x0c06 }, + { 0x93, 0x0000 }, + { 0x94, 0x0000 }, + { 0x95, 0x0000 }, + { 0x97, 0x0000 }, + { 0x98, 0x0000 }, + { 0x99, 0x0000 }, + { 0x9a, 0x2184 }, + { 0x9b, 0x010a }, + { 0x9c, 0x0aea }, + { 0x9d, 0x000c }, + { 0x9e, 0x0400 }, + { 0xae, 0x7000 }, + { 0xaf, 0x0000 }, + { 0xb0, 0x6000 }, + { 0xb1, 0x0000 }, + { 0xb2, 0x0000 }, + { 0xb3, 0x001f }, + { 0xb4, 0x2206 }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xb7, 0x0000 }, + { 0xbb, 0x0000 }, + { 0xbc, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x0000 }, + { 0xc0, 0x0000 }, + { 0xc1, 0x0000 }, + { 0xc2, 0x0000 }, + { 0xcd, 0x0000 }, + { 0xce, 0x0000 }, + { 0xcf, 0x1813 }, + { 0xd0, 0x0690 }, + { 0xd1, 0x1c17 }, + { 0xd3, 0xb320 }, + { 0xd4, 0x0000 }, + { 0xd6, 0x0400 }, + { 0xd9, 0x0809 }, + { 0xda, 0x0000 }, + { 0xdb, 0x0001 }, + { 0xdc, 0x0049 }, + { 0xdd, 0x0009 }, + { 0xe6, 0x8000 }, + { 0xe7, 0x0000 }, + { 0xec, 0xb300 }, + { 0xed, 0x0000 }, + { 0xee, 0xb300 }, + { 0xef, 0x0000 }, + { 0xf8, 0x0000 }, + { 0xf9, 0x0000 }, + { 0xfa, 0x8010 }, + { 0xfb, 0x0033 }, + { 0xfc, 0x0080 }, +}; + +unsigned rt5670_pdm1_read(struct snd_soc_codec *codec, unsigned int reg) +{ + int ret; + + ret = snd_soc_write(codec, RT5670_PDM1_DATA_CTRL2, reg<<8); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_PDM_DATA_CTRL1, 0x0200); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + do + ret = snd_soc_read(codec, RT5670_PDM_DATA_CTRL1); + while (ret & 0x0100); + return snd_soc_read(codec, RT5670_PDM1_DATA_CTRL4); +err: + return ret; +} +EXPORT_SYMBOL(rt5670_pdm1_read); + +int rt5670_pdm1_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + int ret; + + ret = snd_soc_write(codec, RT5670_PDM1_DATA_CTRL3, value); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_PDM1_DATA_CTRL2, reg<<8); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_PDM_DATA_CTRL1, 0x0600); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_PDM_DATA_CTRL1, 0x3600); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + do + ret = snd_soc_read(codec, RT5670_PDM_DATA_CTRL1); + while (ret & 0x0100); + return 0; + +err: + return ret; +} +EXPORT_SYMBOL(rt5670_pdm1_write); + +unsigned rt5670_pdm2_read(struct snd_soc_codec *codec, unsigned int reg) +{ + int ret; + + ret = snd_soc_write(codec, RT5670_PDM2_DATA_CTRL2, reg<<8); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_PDM_DATA_CTRL1, 0x0002); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + do + ret = snd_soc_read(codec, RT5670_PDM_DATA_CTRL1); + while (ret & 0x0001); + return snd_soc_read(codec, RT5670_PDM2_DATA_CTRL4); +err: + return ret; +} +EXPORT_SYMBOL(rt5670_pdm2_read); + +int rt5670_pdm2_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + int ret; + + ret = snd_soc_write(codec, RT5670_PDM2_DATA_CTRL3, value); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_PDM2_DATA_CTRL2, reg<<8); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_PDM_DATA_CTRL1, 0x0006); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_PDM_DATA_CTRL1, 0x0036); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + do + ret = snd_soc_read(codec, RT5670_PDM_DATA_CTRL1); + while (ret & 0x0001); + + return 0; + +err: + return ret; +} +EXPORT_SYMBOL(rt5670_pdm2_write); + +static bool rt5670_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5670_ranges); i++) + if ((reg >= rt5670_ranges[i].window_start && + reg <= rt5670_ranges[i].window_start + + rt5670_ranges[i].window_len) || + (reg >= rt5670_ranges[i].range_min && + reg <= rt5670_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5670_RESET: + case RT5670_PDM_DATA_CTRL1: + case RT5670_PDM1_DATA_CTRL4: + case RT5670_PDM2_DATA_CTRL4: + case RT5670_PRIV_DATA: + case RT5670_ASRC_5: + case RT5670_CJ_CTRL1: + case RT5670_CJ_CTRL2: + case RT5670_CJ_CTRL3: + case RT5670_A_JD_CTRL1: + case RT5670_A_JD_CTRL2: + case RT5670_VAD_CTRL5: + case RT5670_ADC_EQ_CTRL1: + case RT5670_EQ_CTRL1: + case RT5670_ALC_CTRL_1: + case RT5670_IRQ_CTRL2: + case RT5670_IRQ_CTRL3: + case RT5670_INT_IRQ_ST: + case RT5670_IL_CMD: + case RT5670_DSP_CTRL1: + case RT5670_DSP_CTRL2: + case RT5670_DSP_CTRL3: + case RT5670_DSP_CTRL4: + case RT5670_DSP_CTRL5: + case RT5670_VENDOR_ID: + case RT5670_VENDOR_ID1: + case RT5670_VENDOR_ID2: + return true; + default: + return false; + } +} + +static bool rt5670_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5670_ranges); i++) + if ((reg >= rt5670_ranges[i].window_start && + reg <= rt5670_ranges[i].window_start + + rt5670_ranges[i].window_len) || + (reg >= rt5670_ranges[i].range_min && + reg <= rt5670_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5670_RESET: + case RT5670_HP_VOL: + case RT5670_LOUT1: + case RT5670_CJ_CTRL1: + case RT5670_CJ_CTRL2: + case RT5670_CJ_CTRL3: + case RT5670_IN2: + case RT5670_INL1_INR1_VOL: + case RT5670_DAC1_DIG_VOL: + case RT5670_DAC2_DIG_VOL: + case RT5670_DAC_CTRL: + case RT5670_STO1_ADC_DIG_VOL: + case RT5670_MONO_ADC_DIG_VOL: + case RT5670_STO2_ADC_DIG_VOL: + case RT5670_ADC_BST_VOL1: + case RT5670_ADC_BST_VOL2: + case RT5670_STO2_ADC_MIXER: + case RT5670_STO1_ADC_MIXER: + case RT5670_MONO_ADC_MIXER: + case RT5670_AD_DA_MIXER: + case RT5670_STO_DAC_MIXER: + case RT5670_DD_MIXER: + case RT5670_DIG_MIXER: + case RT5670_DSP_PATH1: + case RT5670_DSP_PATH2: + case RT5670_DIG_INF1_DATA: + case RT5670_DIG_INF2_DATA: + case RT5670_PDM_OUT_CTRL: + case RT5670_PDM_DATA_CTRL1: + case RT5670_PDM1_DATA_CTRL2: + case RT5670_PDM1_DATA_CTRL3: + case RT5670_PDM1_DATA_CTRL4: + case RT5670_PDM2_DATA_CTRL2: + case RT5670_PDM2_DATA_CTRL3: + case RT5670_PDM2_DATA_CTRL4: + case RT5670_REC_L1_MIXER: + case RT5670_REC_L2_MIXER: + case RT5670_REC_R1_MIXER: + case RT5670_REC_R2_MIXER: + case RT5670_HPO_MIXER: + case RT5670_MONO_MIXER: + case RT5670_OUT_L1_MIXER: + case RT5670_OUT_R1_MIXER: + case RT5670_LOUT_MIXER: + case RT5670_PWR_DIG1: + case RT5670_PWR_DIG2: + case RT5670_PWR_ANLG1: + case RT5670_PWR_ANLG2: + case RT5670_PWR_MIXER: + case RT5670_PWR_VOL: + case RT5670_PRIV_INDEX: + case RT5670_PRIV_DATA: + case RT5670_I2S4_SDP: + case RT5670_I2S1_SDP: + case RT5670_I2S2_SDP: + case RT5670_I2S3_SDP: + case RT5670_ADDA_CLK1: + case RT5670_ADDA_CLK2: + case RT5670_DMIC_CTRL1: + case RT5670_DMIC_CTRL2: + case RT5670_TDM_CTRL_1: + case RT5670_TDM_CTRL_2: + case RT5670_TDM_CTRL_3: + case RT5670_DSP_CLK: + case RT5670_GLB_CLK: + case RT5670_PLL_CTRL1: + case RT5670_PLL_CTRL2: + case RT5670_ASRC_1: + case RT5670_ASRC_2: + case RT5670_ASRC_3: + case RT5670_ASRC_4: + case RT5670_ASRC_5: + case RT5670_ASRC_7: + case RT5670_ASRC_8: + case RT5670_ASRC_9: + case RT5670_ASRC_10: + case RT5670_ASRC_11: + case RT5670_ASRC_12: + case RT5670_ASRC_13: + case RT5670_ASRC_14: + case RT5670_DEPOP_M1: + case RT5670_DEPOP_M2: + case RT5670_DEPOP_M3: + case RT5670_CHARGE_PUMP: + case RT5670_MICBIAS: + case RT5670_A_JD_CTRL1: + case RT5670_A_JD_CTRL2: + case RT5670_VAD_CTRL1: + case RT5670_VAD_CTRL2: + case RT5670_VAD_CTRL3: + case RT5670_VAD_CTRL4: + case RT5670_VAD_CTRL5: + case RT5670_ADC_EQ_CTRL1: + case RT5670_ADC_EQ_CTRL2: + case RT5670_EQ_CTRL1: + case RT5670_EQ_CTRL2: + case RT5670_ALC_DRC_CTRL1: + case RT5670_ALC_DRC_CTRL2: + case RT5670_ALC_CTRL_1: + case RT5670_ALC_CTRL_2: + case RT5670_ALC_CTRL_3: + case RT5670_JD_CTRL: + case RT5670_IRQ_CTRL1: + case RT5670_IRQ_CTRL2: + case RT5670_IRQ_CTRL3: + case RT5670_INT_IRQ_ST: + case RT5670_GPIO_CTRL1: + case RT5670_GPIO_CTRL2: + case RT5670_GPIO_CTRL3: + case RT5670_SCRABBLE_FUN: + case RT5670_SCRABBLE_CTRL: + case RT5670_BASE_BACK: + case RT5670_MP3_PLUS1: + case RT5670_MP3_PLUS2: + case RT5670_ADJ_HPF1: + case RT5670_ADJ_HPF2: + case RT5670_HP_CALIB_AMP_DET: + case RT5670_SV_ZCD1: + case RT5670_SV_ZCD2: + case RT5670_IL_CMD: + case RT5670_IL_CMD2: + case RT5670_IL_CMD3: + case RT5670_DRC_HL_CTRL1: + case RT5670_DRC_HL_CTRL2: + case RT5670_ADC_MONO_HP_CTRL1: + case RT5670_ADC_MONO_HP_CTRL2: + case RT5670_ADC_STO2_HP_CTRL1: + case RT5670_ADC_STO2_HP_CTRL2: + case RT5670_JD_CTRL3: + case RT5670_JD_CTRL4: + case RT5670_DIG_MISC: + case RT5670_DSP_CTRL1: + case RT5670_DSP_CTRL2: + case RT5670_DSP_CTRL3: + case RT5670_DSP_CTRL4: + case RT5670_DSP_CTRL5: + case RT5670_GEN_CTRL2: + case RT5670_GEN_CTRL3: + case RT5670_VENDOR_ID: + case RT5670_VENDOR_ID1: + case RT5670_VENDOR_ID2: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static unsigned int bst_tlv[] = { + TLV_DB_RANGE_HEAD(7), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), +}; + +/* Interface data select */ +static const char * const rt5670_data_select[] = { + "Normal", "Swap", "left copy to right", "right copy to left" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5670_if2_dac_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_DAC_SEL_SFT, rt5670_data_select); + +static const SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_ADC_SEL_SFT, rt5670_data_select); + +static const struct snd_kcontrol_new rt5670_snd_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE("HP Playback Switch", RT5670_HP_VOL, + RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 39, 0, out_vol_tlv), + /* OUTPUT Control */ + SOC_DOUBLE("OUT Playback Switch", RT5670_LOUT1, + RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("OUT Channel Switch", RT5670_LOUT1, + RT5670_VOL_L_SFT, RT5670_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5670_LOUT1, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, 39, 1, out_vol_tlv), + /* DAC Digital Volume */ + SOC_DOUBLE("DAC2 Playback Switch", RT5670_DAC_CTRL, + RT5670_M_DAC_L2_VOL_SFT, RT5670_M_DAC_R2_VOL_SFT, 1, 1), + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5670_DAC1_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 175, 0, dac_vol_tlv), + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5670_DAC2_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 175, 0, dac_vol_tlv), + /* IN1/IN2 Control */ + SOC_SINGLE_TLV("IN1 Boost", RT5670_CJ_CTRL1, + RT5670_BST_SFT1, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN2 Boost", RT5670_IN2, + RT5670_BST_SFT1, 8, 0, bst_tlv), + /* INL/INR Volume Control */ + SOC_DOUBLE_TLV("IN Capture Volume", RT5670_INL1_INR1_VOL, + RT5670_INL_VOL_SFT, RT5670_INR_VOL_SFT, + 31, 1, in_vol_tlv), + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5670_STO1_ADC_DIG_VOL, + RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5670_STO1_ADC_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 127, 0, adc_vol_tlv), + + SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5670_MONO_ADC_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5670_ADC_BST_VOL1, + RT5670_STO1_ADC_L_BST_SFT, RT5670_STO1_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), + + SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5670_ADC_BST_VOL1, + RT5670_STO2_ADC_L_BST_SFT, RT5670_STO2_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), + + SOC_ENUM("ADC IF2 Data Switch", rt5670_if2_adc_enum), + SOC_ENUM("DAC IF2 Data Switch", rt5670_if2_dac_enum), +}; + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + int div[] = {2, 3, 4, 6, 8, 12}; + int idx = -EINVAL, i; + int rate, red, bound, temp; + + rate = rt5670->lrck[RT5670_AIF1] << 8; + red = 2000000 * 12; + for (i = 0; i < ARRAY_SIZE(div); i++) { + bound = div[i] * 2000000; + if (rate > bound) + continue; + temp = bound - rate; + if (temp < red) { + red = temp; + idx = i; + } + } + if (rt5670->pdata.asrc_en) + idx = 5; + + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5670_DMIC_CTRL1, + RT5670_DMIC_CLK_MASK, idx << RT5670_DMIC_CLK_SFT); + return idx; +} + +static int check_sysclk1_source(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + + val = snd_soc_read(source->codec, RT5670_GLB_CLK); + val &= RT5670_SCLK_SRC_MASK; + if (val == RT5670_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto2_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto2_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER, + RT5670_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER, + RT5670_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER, + RT5670_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER, + RT5670_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L1_STO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L1_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L2_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R2_MONO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R1_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R2_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L2_MONO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dig_l_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5670_DIG_MIXER, + RT5670_M_STO_L_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_L2_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_R2_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dig_r_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5670_DIG_MIXER, + RT5670_M_STO_R_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_R2_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_L2_DAC_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5670_rec_l_mix[] = { + SOC_DAPM_SINGLE("INL Switch", RT5670_REC_L2_MIXER, + RT5670_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5670_REC_L2_MIXER, + RT5670_M_BST2_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5670_REC_L2_MIXER, + RT5670_M_BST1_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_rec_r_mix[] = { + SOC_DAPM_SINGLE("INR Switch", RT5670_REC_R2_MIXER, + RT5670_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5670_REC_R2_MIXER, + RT5670_M_BST2_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5670_REC_R2_MIXER, + RT5670_M_BST1_RM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5670_OUT_L1_MIXER, + RT5670_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_DAC_L2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_BST2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5670_OUT_R1_MIXER, + RT5670_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_DAC_R2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_DAC_R1_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpo_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPVOL Switch", RT5670_HPO_MIXER, + RT5670_M_HPVOL_HM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpvoll_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACL1_HML_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5670_HPO_MIXER, + RT5670_M_INL1_HML_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpvolr_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACR1_HMR_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5670_HPO_MIXER, + RT5670_M_INR1_HMR_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_lout_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_LOUT_MIXER, + RT5670_M_DAC_L1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_LOUT_MIXER, + RT5670_M_DAC_R1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX L Switch", RT5670_LOUT_MIXER, + RT5670_M_OV_L_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX R Switch", RT5670_LOUT_MIXER, + RT5670_M_OV_R_LM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpl_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACL1_HML_SFT, 1, 1), + SOC_DAPM_SINGLE("INL1 Switch", RT5670_HPO_MIXER, + RT5670_M_INL1_HML_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpr_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACR1_HMR_SFT, 1, 1), + SOC_DAPM_SINGLE("INR1 Switch", RT5670_HPO_MIXER, + RT5670_M_INR1_HMR_SFT, 1, 1), +}; + +/* DAC1 L/R source */ /* MX-29 [9:8] [11:10] */ +static const char * const rt5670_dac1_src[] = { + "IF1 DAC", "IF2 DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac1l_enum, RT5670_AD_DA_MIXER, + RT5670_DAC1_L_SEL_SFT, rt5670_dac1_src); + +static const struct snd_kcontrol_new rt5670_dac1l_mux = + SOC_DAPM_ENUM("DAC1 L source", rt5670_dac1l_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac1r_enum, RT5670_AD_DA_MIXER, + RT5670_DAC1_R_SEL_SFT, rt5670_dac1_src); + +static const struct snd_kcontrol_new rt5670_dac1r_mux = + SOC_DAPM_ENUM("DAC1 R source", rt5670_dac1r_enum); + +/*DAC2 L/R source*/ /* MX-1B [6:4] [2:0] */ +/* TODO Use SOC_VALUE_ENUM_SINGLE_DECL */ +static const char * const rt5670_dac12_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC", "TxDC DAC", + "Bass", "VAD_ADC", "IF4 DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac2l_enum, RT5670_DAC_CTRL, + RT5670_DAC2_L_SEL_SFT, rt5670_dac12_src); + +static const struct snd_kcontrol_new rt5670_dac_l2_mux = + SOC_DAPM_ENUM("DAC2 L source", rt5670_dac2l_enum); + +static const char * const rt5670_dacr2_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC", "TxDC DAC", "TxDP ADC", "IF4 DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac2r_enum, RT5670_DAC_CTRL, + RT5670_DAC2_R_SEL_SFT, rt5670_dacr2_src); + +static const struct snd_kcontrol_new rt5670_dac_r2_mux = + SOC_DAPM_ENUM("DAC2 R source", rt5670_dac2r_enum); + +/*RxDP source*/ /* MX-2D [15:13] */ +static const char * const rt5670_rxdp_src[] = { + "IF2 DAC", "IF1 DAC", "STO1 ADC Mixer", "STO2 ADC Mixer", + "Mono ADC Mixer L", "Mono ADC Mixer R", "DAC1" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_rxdp_enum, RT5670_DSP_PATH1, + RT5670_RXDP_SEL_SFT, rt5670_rxdp_src); + +static const struct snd_kcontrol_new rt5670_rxdp_mux = + SOC_DAPM_ENUM("DAC2 L source", rt5670_rxdp_enum); + +/* MX-2D [1] [0] */ +static const char * const rt5670_dsp_bypass_src[] = { + "DSP", "Bypass" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dsp_ul_enum, RT5670_DSP_PATH1, + RT5670_DSP_UL_SFT, rt5670_dsp_bypass_src); + +static const struct snd_kcontrol_new rt5670_dsp_ul_mux = + SOC_DAPM_ENUM("DSP UL source", rt5670_dsp_ul_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dsp_dl_enum, RT5670_DSP_PATH1, + RT5670_DSP_DL_SFT, rt5670_dsp_bypass_src); + +static const struct snd_kcontrol_new rt5670_dsp_dl_mux = + SOC_DAPM_ENUM("DSP DL source", rt5670_dsp_dl_enum); + +/* Stereo2 ADC source */ +/* MX-26 [15] */ +static const char * const rt5670_stereo2_adc_lr_src[] = { + "L", "LR" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc_lr_enum, RT5670_STO2_ADC_MIXER, + RT5670_STO2_ADC_SRC_SFT, rt5670_stereo2_adc_lr_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_lr_mux = + SOC_DAPM_ENUM("Stereo2 ADC LR source", rt5670_stereo2_adc_lr_enum); + +/* Stereo1 ADC source */ +/* MX-27 MX-26 [12] */ +static const char * const rt5670_stereo_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_adc1_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_l1_mux = + SOC_DAPM_ENUM("Stereo1 ADC L1 source", rt5670_stereo1_adc1_enum); + +static const struct snd_kcontrol_new rt5670_sto_adc_r1_mux = + SOC_DAPM_ENUM("Stereo1 ADC R1 source", rt5670_stereo1_adc1_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc1_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_l1_mux = + SOC_DAPM_ENUM("Stereo2 ADC L1 source", rt5670_stereo2_adc1_enum); + +static const struct snd_kcontrol_new rt5670_sto2_adc_r1_mux = + SOC_DAPM_ENUM("Stereo2 ADC R1 source", rt5670_stereo2_adc1_enum); + +/* MX-27 MX-26 [11] */ +static const char * const rt5670_stereo_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_adc2_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_l2_mux = + SOC_DAPM_ENUM("Stereo1 ADC L2 source", rt5670_stereo1_adc2_enum); + +static const struct snd_kcontrol_new rt5670_sto_adc_r2_mux = + SOC_DAPM_ENUM("Stereo1 ADC R2 source", rt5670_stereo1_adc2_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc2_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_l2_mux = + SOC_DAPM_ENUM("Stereo2 ADC L2 source", rt5670_stereo2_adc2_enum); + +static const struct snd_kcontrol_new rt5670_sto2_adc_r2_mux = + SOC_DAPM_ENUM("Stereo2 ADC R2 source", rt5670_stereo2_adc2_enum); + +/* MX-27 MX26 [10] */ +static const char * const rt5670_stereo_adc_src[] = { + "ADC1L ADC2R", "ADC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_adc_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_SRC_SFT, rt5670_stereo_adc_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_mux = + SOC_DAPM_ENUM("Stereo1 ADC source", rt5670_stereo1_adc_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_SRC_SFT, rt5670_stereo_adc_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_mux = + SOC_DAPM_ENUM("Stereo2 ADC source", rt5670_stereo2_adc_enum); + +/* MX-27 MX-26 [9:8] */ +static const char * const rt5670_stereo_dmic_src[] = { + "DMIC1", "DMIC2", "DMIC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_dmic_enum, RT5670_STO1_ADC_MIXER, + RT5670_DMIC_SRC_SFT, rt5670_stereo_dmic_src); + +static const struct snd_kcontrol_new rt5670_sto1_dmic_mux = + SOC_DAPM_ENUM("Stereo1 DMIC source", rt5670_stereo1_dmic_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_dmic_enum, RT5670_STO2_ADC_MIXER, + RT5670_DMIC_SRC_SFT, rt5670_stereo_dmic_src); + +static const struct snd_kcontrol_new rt5670_sto2_dmic_mux = + SOC_DAPM_ENUM("Stereo2 DMIC source", rt5670_stereo2_dmic_enum); + +/* MX-27 [0] */ +static const char * const rt5670_stereo_dmic3_src[] = { + "DMIC3", "PDM ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo_dmic3_enum, RT5670_STO1_ADC_MIXER, + RT5670_DMIC3_SRC_SFT, rt5670_stereo_dmic3_src); + +static const struct snd_kcontrol_new rt5670_sto_dmic3_mux = + SOC_DAPM_ENUM("Stereo DMIC3 source", rt5670_stereo_dmic3_enum); + +/* Mono ADC source */ +/* MX-28 [12] */ +static const char * const rt5670_mono_adc_l1_src[] = { + "Mono DAC MIXL", "ADC1" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_l1_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_L1_SRC_SFT, rt5670_mono_adc_l1_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_l1_mux = + SOC_DAPM_ENUM("Mono ADC1 left source", rt5670_mono_adc_l1_enum); +/* MX-28 [11] */ +static const char * const rt5670_mono_adc_l2_src[] = { + "Mono DAC MIXL", "DMIC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_l2_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_L2_SRC_SFT, rt5670_mono_adc_l2_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_l2_mux = + SOC_DAPM_ENUM("Mono ADC2 left source", rt5670_mono_adc_l2_enum); + +/* MX-28 [9:8] */ +static const char * const rt5670_mono_dmic_src[] = { + "DMIC1", "DMIC2", "DMIC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_dmic_l_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_DMIC_L_SRC_SFT, rt5670_mono_dmic_src); + +static const struct snd_kcontrol_new rt5670_mono_dmic_l_mux = + SOC_DAPM_ENUM("Mono DMIC left source", rt5670_mono_dmic_l_enum); +/* MX-28 [1:0] */ +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_dmic_r_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_DMIC_R_SRC_SFT, rt5670_mono_dmic_src); + +static const struct snd_kcontrol_new rt5670_mono_dmic_r_mux = + SOC_DAPM_ENUM("Mono DMIC Right source", rt5670_mono_dmic_r_enum); +/* MX-28 [4] */ +static const char * const rt5670_mono_adc_r1_src[] = { + "Mono DAC MIXR", "ADC2" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_r1_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_R1_SRC_SFT, rt5670_mono_adc_r1_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_r1_mux = + SOC_DAPM_ENUM("Mono ADC1 right source", rt5670_mono_adc_r1_enum); +/* MX-28 [3] */ +static const char * const rt5670_mono_adc_r2_src[] = { + "Mono DAC MIXR", "DMIC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_r2_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_R2_SRC_SFT, rt5670_mono_adc_r2_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_r2_mux = + SOC_DAPM_ENUM("Mono ADC2 right source", rt5670_mono_adc_r2_enum); + +/* MX-2F [15] */ +static const char * const rt5670_if1_adc2_in_src[] = { + "IF_ADC2", "VAD_ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc2_in_enum, RT5670_DIG_INF1_DATA, + RT5670_IF1_ADC2_IN_SFT, rt5670_if1_adc2_in_src); + +static const struct snd_kcontrol_new rt5670_if1_adc2_in_mux = + SOC_DAPM_ENUM("IF1 ADC2 IN source", rt5670_if1_adc2_in_enum); + +/* MX-2F [14:12] */ +static const char * const rt5670_if2_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "IF_ADC3", "TxDC_DAC", "TxDP_ADC", "VAD_ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if2_adc_in_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_ADC_IN_SFT, rt5670_if2_adc_in_src); + +static const struct snd_kcontrol_new rt5670_if2_adc_in_mux = + SOC_DAPM_ENUM("IF2 ADC IN source", rt5670_if2_adc_in_enum); + +/* MX-30 [5:4] */ +static const char * const rt5670_if4_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "IF_ADC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if4_adc_in_enum, RT5670_DIG_INF2_DATA, + RT5670_IF4_ADC_IN_SFT, rt5670_if4_adc_in_src); + +static const struct snd_kcontrol_new rt5670_if4_adc_in_mux = + SOC_DAPM_ENUM("IF4 ADC IN source", rt5670_if4_adc_in_enum); + +/* MX-31 [15] [13] [11] [9] */ +static const char * const rt5670_pdm_src[] = { + "Mono DAC", "Stereo DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm1_l_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM1_L_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm1_l_mux = + SOC_DAPM_ENUM("PDM1 L source", rt5670_pdm1_l_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm1_r_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM1_R_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm1_r_mux = + SOC_DAPM_ENUM("PDM1 R source", rt5670_pdm1_r_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm2_l_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM2_L_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm2_l_mux = + SOC_DAPM_ENUM("PDM2 L source", rt5670_pdm2_l_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm2_r_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM2_R_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm2_r_mux = + SOC_DAPM_ENUM("PDM2 R source", rt5670_pdm2_r_enum); + +/* MX-FA [12] */ +static const char * const rt5670_if1_adc1_in1_src[] = { + "IF_ADC1", "IF1_ADC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc1_in1_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC1_IN1_SFT, rt5670_if1_adc1_in1_src); + +static const struct snd_kcontrol_new rt5670_if1_adc1_in1_mux = + SOC_DAPM_ENUM("IF1 ADC1 IN1 source", rt5670_if1_adc1_in1_enum); + +/* MX-FA [11] */ +static const char * const rt5670_if1_adc1_in2_src[] = { + "IF1_ADC1_IN1", "IF1_ADC4" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc1_in2_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC1_IN2_SFT, rt5670_if1_adc1_in2_src); + +static const struct snd_kcontrol_new rt5670_if1_adc1_in2_mux = + SOC_DAPM_ENUM("IF1 ADC1 IN2 source", rt5670_if1_adc1_in2_enum); + +/* MX-FA [10] */ +static const char * const rt5670_if1_adc2_in1_src[] = { + "IF1_ADC2_IN", "IF1_ADC4" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc2_in1_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC2_IN1_SFT, rt5670_if1_adc2_in1_src); + +static const struct snd_kcontrol_new rt5670_if1_adc2_in1_mux = + SOC_DAPM_ENUM("IF1 ADC2 IN1 source", rt5670_if1_adc2_in1_enum); + +/* MX-9D [9:8] */ +static const char * const rt5670_vad_adc_src[] = { + "Sto1 ADC L", "Mono ADC L", "Mono ADC R", "Sto2 ADC L" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_vad_adc_enum, RT5670_VAD_CTRL4, + RT5670_VAD_SEL_SFT, rt5670_vad_adc_src); + +static const struct snd_kcontrol_new rt5670_vad_adc_mux = + SOC_DAPM_ENUM("VAD ADC source", rt5670_vad_adc_enum); + +static int rt5670_adc_clk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(rt5670->regmap, RT5670_PR_BASE + + RT5670_CHOP_DAC_ADC, 0x1000, 0x1000); + break; + + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(rt5670->regmap, RT5670_PR_BASE + + RT5670_CHOP_DAC_ADC, 0x1000, 0x0000); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_sto1_adcl_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_STO1_ADC_DIG_VOL, + RT5670_L_MUTE, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_STO1_ADC_DIG_VOL, + RT5670_L_MUTE, + RT5670_L_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_sto1_adcr_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_STO1_ADC_DIG_VOL, + RT5670_R_MUTE, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_STO1_ADC_DIG_VOL, + RT5670_R_MUTE, + RT5670_R_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_mono_adcl_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_MONO_ADC_DIG_VOL, + RT5670_L_MUTE, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_MONO_ADC_DIG_VOL, + RT5670_L_MUTE, + RT5670_L_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_mono_adcr_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_MONO_ADC_DIG_VOL, + RT5670_R_MUTE, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_MONO_ADC_DIG_VOL, + RT5670_R_MUTE, + RT5670_R_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +void hp_amp_power(struct snd_soc_codec *codec, int on) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + if (on) { + regmap_update_bits(rt5670->regmap, RT5670_CHARGE_PUMP, + RT5670_PM_HP_MASK, RT5670_PM_HP_HV); + regmap_update_bits(rt5670->regmap, RT5670_GEN_CTRL2, + 0x0400, 0x0400); + /* headphone amp power on */ + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, + RT5670_PWR_HA | RT5670_PWR_FV1 | + RT5670_PWR_FV2, RT5670_PWR_HA | + RT5670_PWR_FV1 | RT5670_PWR_FV2); + /* depop parameters */ + regmap_write(rt5670->regmap, RT5670_DEPOP_M2, 0x3100); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8009); + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_HP_DCC_INT1, 0x9f00); + mdelay(20); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); + } else { + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x0004); + msleep(30); + } +} + +static void rt5670_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + /* headphone unmute sequence */ + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xb400); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0772); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x805d); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x831d); + regmap_update_bits(rt5670->regmap, RT5670_GEN_CTRL2, + 0x0300, 0x0300); + regmap_update_bits(rt5670->regmap, RT5670_HP_VOL, + RT5670_L_MUTE | RT5670_R_MUTE, 0); + msleep(80); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); +} + +static void rt5670_pmd_depop(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + /* headphone mute sequence */ + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xb400); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0772); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x803d); + mdelay(10); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x831d); + mdelay(10); + regmap_update_bits(rt5670->regmap, RT5670_HP_VOL, + RT5670_L_MUTE | RT5670_R_MUTE, RT5670_L_MUTE | RT5670_R_MUTE); + msleep(20); + regmap_update_bits(rt5670->regmap, RT5670_GEN_CTRL2, 0x0300, 0x0); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0707); + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xfc00); +} + +static int rt5670_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power(codec, 1); + break; + case SND_SOC_DAPM_PRE_PMD: + hp_amp_power(codec, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt5670_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt5670_pmu_depop(codec); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt5670_pmd_depop(codec); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_lout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_LOUT1, + RT5670_L_MUTE | RT5670_R_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_LOUT1, + RT5670_L_MUTE | RT5670_R_MUTE, + RT5670_L_MUTE | RT5670_R_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_bst1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x0); + snd_soc_update_bits(codec, RT5670_CHARGE_PUMP, + RT5670_OSW_L_MASK | RT5670_OSW_R_MASK, + RT5670_OSW_L_DIS | RT5670_OSW_R_DIS); + + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST1_P, RT5670_PWR_BST1_P); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4); + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST1_P, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_bst2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST2_P, RT5670_PWR_BST2_P); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST2_P, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_pdm1_l_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PDM_OUT_CTRL, + RT5670_M_PDM1_L, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PDM_OUT_CTRL, + RT5670_M_PDM1_L, RT5670_M_PDM1_L); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_pdm1_r_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PDM_OUT_CTRL, + RT5670_M_PDM1_R, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PDM_OUT_CTRL, + RT5670_M_PDM1_R, RT5670_M_PDM1_R); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_pdm2_l_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_L, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_L, RT5670_M_PDM2_L); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_pdm2_r_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_R, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_R, RT5670_M_PDM2_R); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_asrc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(w->codec, RT5670_ASRC_1, 0xffff); + snd_soc_write(w->codec, RT5670_ASRC_2, 0x1221); + snd_soc_write(w->codec, RT5670_ASRC_3, 0x0022); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(w->codec, RT5670_ASRC_1, 0); + snd_soc_write(w->codec, RT5670_ASRC_2, 0); + snd_soc_write(w->codec, RT5670_ASRC_3, 0); + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL1", RT5670_PWR_ANLG2, + RT5670_PWR_PLL_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S DSP", RT5670_PWR_DIG2, + RT5670_PWR_I2S_DSP_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("JD Power", RT5670_PWR_ANLG2, + RT5670_PWR_JD1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5670_PWR_VOL, + RT5670_PWR_MIC_DET_BIT, 0, NULL, 0), + + /* Input Side */ + /* micbias */ + SND_SOC_DAPM_MICBIAS("micbias1", RT5670_PWR_ANLG2, + RT5670_PWR_MB1_BIT, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + SND_SOC_DAPM_INPUT("DMIC L2"), + SND_SOC_DAPM_INPUT("DMIC R2"), + SND_SOC_DAPM_INPUT("DMIC L3"), + SND_SOC_DAPM_INPUT("DMIC R3"), + + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC3", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_1_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_2_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC3 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_3_EN_SFT, 0, NULL, 0), + /* Boost */ + SND_SOC_DAPM_PGA_E("BST1", RT5670_PWR_ANLG2, + RT5670_PWR_BST1_BIT, 0, NULL, 0, rt5670_bst1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("BST2", RT5670_PWR_ANLG2, + RT5670_PWR_BST2_BIT, 0, NULL, 0, rt5670_bst2_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + /* Input Volume */ + SND_SOC_DAPM_PGA("INL VOL", RT5670_PWR_VOL, + RT5670_PWR_IN_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR VOL", RT5670_PWR_VOL, + RT5670_PWR_IN_R_BIT, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5670_PWR_MIXER, RT5670_PWR_RM_L_BIT, 0, + rt5670_rec_l_mix, ARRAY_SIZE(rt5670_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5670_PWR_MIXER, RT5670_PWR_RM_R_BIT, 0, + rt5670_rec_r_mix, ARRAY_SIZE(rt5670_rec_r_mix)), + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, + 0, 0), + SND_SOC_DAPM_ADC("ADC 2", NULL, SND_SOC_NOPM, + 0, 0), + + SND_SOC_DAPM_PGA("ADC 1_2", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("ADC 1 power", RT5670_PWR_DIG1, + RT5670_PWR_ADC_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC 2 power", RT5670_PWR_DIG1, + RT5670_PWR_ADC_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC clock", SND_SOC_NOPM, 0, 0, + rt5670_adc_clk_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU), + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto1_dmic_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_l2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_r2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_r1_mux), + SND_SOC_DAPM_MUX("Stereo2 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_dmic_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_l2_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_r2_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_r1_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC LR Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_lr_mux), + SND_SOC_DAPM_MUX("Mono DMIC L Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_dmic_l_mux), + SND_SOC_DAPM_MUX("Mono DMIC R Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_dmic_r_mux), + SND_SOC_DAPM_MUX("Mono ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_l2_mux), + SND_SOC_DAPM_MUX("Mono ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_l1_mux), + SND_SOC_DAPM_MUX("Mono ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_r1_mux), + SND_SOC_DAPM_MUX("Mono ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_r2_mux), + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("adc stereo1 filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("adc stereo2 filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_S2F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("Sto1 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_sto1_adc_l_mix, ARRAY_SIZE(rt5670_sto1_adc_l_mix), + rt5670_sto1_adcl_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MIXER_E("Sto1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_sto1_adc_r_mix, ARRAY_SIZE(rt5670_sto1_adc_r_mix), + rt5670_sto1_adcr_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MIXER("Sto2 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_sto2_adc_l_mix, ARRAY_SIZE(rt5670_sto2_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto2 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_sto2_adc_r_mix, ARRAY_SIZE(rt5670_sto2_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("adc mono left filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("Mono ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_mono_adc_l_mix, ARRAY_SIZE(rt5670_mono_adc_l_mix), + rt5670_mono_adcl_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("adc mono right filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("Mono ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_mono_adc_r_mix, ARRAY_SIZE(rt5670_mono_adc_r_mix), + rt5670_mono_adcr_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo1 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Sto2 ADC LR MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("VAD_ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DSP */ + SND_SOC_DAPM_PGA("TxDP_ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDP_ADC_L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDP_ADC_R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDC_DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_PGA("8CH TDM Data", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DSP UL Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dsp_ul_mux), + SND_SOC_DAPM_MUX("DSP DL Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dsp_dl_mux), + + SND_SOC_DAPM_MUX("RxDP Mux", SND_SOC_NOPM, 0, 0, + &rt5670_rxdp_mux), + + /* IF2 Mux */ + SND_SOC_DAPM_MUX("IF2 ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if2_adc_in_mux), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5670_PWR_DIG1, + RT5670_PWR_I2S1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5670_PWR_DIG1, + RT5670_PWR_I2S2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("IF1 ADC1 IN1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc1_in1_mux), + SND_SOC_DAPM_MUX("IF1 ADC1 IN2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc1_in2_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 IN Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc2_in_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 IN1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc2_in1_mux), + SND_SOC_DAPM_MUX("VAD ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_vad_adc_mux), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Audio DSP */ + SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5670_dac_l_mix, ARRAY_SIZE(rt5670_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5670_dac_r_mix, ARRAY_SIZE(rt5670_dac_r_mix)), + SND_SOC_DAPM_PGA("DAC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dac_r2_mux), + SND_SOC_DAPM_PGA("DAC L2 Volume", RT5670_PWR_DIG1, + RT5670_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC R2 Volume", RT5670_PWR_DIG1, + RT5670_PWR_DAC_R2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DAC1 L Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dac1l_mux), + SND_SOC_DAPM_MUX("DAC1 R Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dac1r_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono left filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono right filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_sto_dac_l_mix, ARRAY_SIZE(rt5670_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_sto_dac_r_mix, ARRAY_SIZE(rt5670_sto_dac_r_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_mono_dac_l_mix, ARRAY_SIZE(rt5670_mono_dac_l_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_mono_dac_r_mix, ARRAY_SIZE(rt5670_mono_dac_r_mix)), + SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_dig_l_mix, ARRAY_SIZE(rt5670_dig_l_mix)), + SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_dig_r_mix, ARRAY_SIZE(rt5670_dig_r_mix)), + + /* DACs */ + SND_SOC_DAPM_SUPPLY("DAC L1 Power", RT5670_PWR_DIG1, + RT5670_PWR_DAC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R1 Power", RT5670_PWR_DIG1, + RT5670_PWR_DAC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_DAC("DAC L1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC L2", NULL, RT5670_PWR_DIG1, + RT5670_PWR_DAC_L2_BIT, 0), + + SND_SOC_DAPM_DAC("DAC R2", NULL, RT5670_PWR_DIG1, + RT5670_PWR_DAC_R2_BIT, 0), + /* OUT Mixer */ + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5670_PWR_MIXER, RT5670_PWR_OM_L_BIT, + 0, rt5670_out_l_mix, ARRAY_SIZE(rt5670_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5670_PWR_MIXER, RT5670_PWR_OM_R_BIT, + 0, rt5670_out_r_mix, ARRAY_SIZE(rt5670_out_r_mix)), + /* Ouput Volume */ + SND_SOC_DAPM_MIXER("HPOVOL MIXL", RT5670_PWR_VOL, RT5670_PWR_HV_L_BIT, + 0, rt5670_hpvoll_mix, ARRAY_SIZE(rt5670_hpvoll_mix)), + SND_SOC_DAPM_MIXER("HPOVOL MIXR", RT5670_PWR_VOL, RT5670_PWR_HV_R_BIT, + 0, rt5670_hpvolr_mix, ARRAY_SIZE(rt5670_hpvolr_mix)), + SND_SOC_DAPM_PGA("DAC 1", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC 2", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0, + rt5670_hpo_mix, ARRAY_SIZE(rt5670_hpo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", RT5670_PWR_ANLG1, RT5670_PWR_LM_BIT, + 0, rt5670_lout_mix, ARRAY_SIZE(rt5670_lout_mix)), + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5670_hp_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5670_PWR_ANLG1, + RT5670_PWR_HP_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5670_PWR_ANLG1, + RT5670_PWR_HP_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5670_hp_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("LOUT Amp", 1, SND_SOC_NOPM, 0, 0, + rt5670_lout_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* PDM */ + SND_SOC_DAPM_SUPPLY("PDM1 Power", RT5670_PWR_DIG2, + RT5670_PWR_PDM1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2, + RT5670_PWR_PDM2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MUX_E("PDM1 L Mux", SND_SOC_NOPM, 0, 0, + &rt5670_pdm1_l_mux, rt5670_pdm1_l_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("PDM1 R Mux", SND_SOC_NOPM, 0, 0, + &rt5670_pdm1_r_mux, rt5670_pdm1_r_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("PDM2 L Mux", SND_SOC_NOPM, 0, 0, + &rt5670_pdm2_l_mux, rt5670_pdm2_l_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("PDM2 R Mux", SND_SOC_NOPM, 0, 0, + &rt5670_pdm2_r_mux, rt5670_pdm2_r_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("PDM1L"), + SND_SOC_DAPM_OUTPUT("PDM1R"), + SND_SOC_DAPM_OUTPUT("PDM2L"), + SND_SOC_DAPM_OUTPUT("PDM2R"), +}; + +static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { + { "micbias1", NULL, "DAC L1 Power" }, + { "micbias1", NULL, "DAC R1 Power" }, + + { "DMIC1", NULL, "DMIC L1" }, + { "DMIC1", NULL, "DMIC R1" }, + { "DMIC2", NULL, "DMIC L2" }, + { "DMIC2", NULL, "DMIC R2" }, + { "DMIC3", NULL, "DMIC L3" }, + { "DMIC3", NULL, "DMIC R3" }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST1", NULL, "JD Power" }, + { "BST1", NULL, "Mic Det Power" }, + { "BST2", NULL, "IN2P" }, + { "BST2", NULL, "IN2N" }, + + { "INL VOL", NULL, "IN2P" }, + { "INR VOL", NULL, "IN2N" }, + + { "RECMIXL", "INL Switch", "INL VOL" }, + { "RECMIXL", "BST2 Switch", "BST2" }, + { "RECMIXL", "BST1 Switch", "BST1" }, + + { "RECMIXR", "INR Switch", "INR VOL" }, + { "RECMIXR", "BST2 Switch", "BST2" }, + { "RECMIXR", "BST1 Switch", "BST1" }, + + { "ADC 1", NULL, "RECMIXL" }, + { "ADC 1", NULL, "ADC 1 power" }, + { "ADC 1", NULL, "ADC clock" }, + { "ADC 2", NULL, "RECMIXR" }, + { "ADC 2", NULL, "ADC 2 power" }, + { "ADC 2", NULL, "ADC clock" }, + + { "DMIC L1", NULL, "DMIC CLK" }, + { "DMIC L1", NULL, "DMIC1 Power" }, + { "DMIC R1", NULL, "DMIC CLK" }, + { "DMIC R1", NULL, "DMIC1 Power" }, + { "DMIC L2", NULL, "DMIC CLK" }, + { "DMIC L2", NULL, "DMIC2 Power" }, + { "DMIC R2", NULL, "DMIC CLK" }, + { "DMIC R2", NULL, "DMIC2 Power" }, + { "DMIC L3", NULL, "DMIC CLK" }, + { "DMIC L3", NULL, "DMIC3 Power" }, + { "DMIC R3", NULL, "DMIC CLK" }, + { "DMIC R3", NULL, "DMIC3 Power" }, + + { "Stereo1 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo1 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo1 DMIC Mux", "DMIC3", "DMIC3" }, + + { "Stereo2 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo2 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo2 DMIC Mux", "DMIC3", "DMIC3" }, + + { "Mono DMIC L Mux", "DMIC1", "DMIC L1" }, + { "Mono DMIC L Mux", "DMIC2", "DMIC L2" }, + { "Mono DMIC L Mux", "DMIC3", "DMIC L3" }, + + { "Mono DMIC R Mux", "DMIC1", "DMIC R1" }, + { "Mono DMIC R Mux", "DMIC2", "DMIC R2" }, + { "Mono DMIC R Mux", "DMIC3", "DMIC R3" }, + + { "ADC 1_2", NULL, "ADC 1" }, + { "ADC 1_2", NULL, "ADC 2" }, + + { "Stereo1 ADC L2 Mux", "DMIC", "Stereo1 DMIC Mux" }, + { "Stereo1 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, + { "Stereo1 ADC L1 Mux", "ADC", "ADC 1_2" }, + { "Stereo1 ADC L1 Mux", "DAC MIX", "DAC MIXL" }, + + { "Stereo1 ADC R1 Mux", "ADC", "ADC 1_2" }, + { "Stereo1 ADC R1 Mux", "DAC MIX", "DAC MIXR" }, + { "Stereo1 ADC R2 Mux", "DMIC", "Stereo1 DMIC Mux" }, + { "Stereo1 ADC R2 Mux", "DAC MIX", "DAC MIXR" }, + + { "Mono ADC L2 Mux", "DMIC", "Mono DMIC L Mux" }, + { "Mono ADC L2 Mux", "Mono DAC MIXL", "Mono DAC MIXL" }, + { "Mono ADC L1 Mux", "Mono DAC MIXL", "Mono DAC MIXL" }, + { "Mono ADC L1 Mux", "ADC1", "ADC 1" }, + + { "Mono ADC R1 Mux", "Mono DAC MIXR", "Mono DAC MIXR" }, + { "Mono ADC R1 Mux", "ADC2", "ADC 2" }, + { "Mono ADC R2 Mux", "DMIC", "Mono DMIC R Mux" }, + { "Mono ADC R2 Mux", "Mono DAC MIXR", "Mono DAC MIXR" }, + + { "Sto1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux" }, + { "Sto1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux" }, + { "Sto1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux" }, + { "Sto1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux" }, + + { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" }, + { "Stereo1 ADC MIXL", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", check_sysclk1_source }, + + { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" }, + { "Stereo1 ADC MIXR", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", check_sysclk1_source }, + + { "Mono ADC MIXL", "ADC1 Switch", "Mono ADC L1 Mux" }, + { "Mono ADC MIXL", "ADC2 Switch", "Mono ADC L2 Mux" }, + { "Mono ADC MIXL", NULL, "adc mono left filter" }, + { "adc mono left filter", NULL, "PLL1", check_sysclk1_source }, + + { "Mono ADC MIXR", "ADC1 Switch", "Mono ADC R1 Mux" }, + { "Mono ADC MIXR", "ADC2 Switch", "Mono ADC R2 Mux" }, + { "Mono ADC MIXR", NULL, "adc mono right filter" }, + { "adc mono right filter", NULL, "PLL1", check_sysclk1_source }, + + { "Stereo2 ADC L2 Mux", "DMIC", "Stereo2 DMIC Mux" }, + { "Stereo2 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, + { "Stereo2 ADC L1 Mux", "ADC", "ADC 1_2" }, + { "Stereo2 ADC L1 Mux", "DAC MIX", "DAC MIXL" }, + + { "Stereo2 ADC R1 Mux", "ADC", "ADC 1_2" }, + { "Stereo2 ADC R1 Mux", "DAC MIX", "DAC MIXR" }, + { "Stereo2 ADC R2 Mux", "DMIC", "Stereo2 DMIC Mux" }, + { "Stereo2 ADC R2 Mux", "DAC MIX", "DAC MIXR" }, + + { "Sto2 ADC MIXL", "ADC1 Switch", "Stereo2 ADC L1 Mux" }, + { "Sto2 ADC MIXL", "ADC2 Switch", "Stereo2 ADC L2 Mux" }, + { "Sto2 ADC MIXR", "ADC1 Switch", "Stereo2 ADC R1 Mux" }, + { "Sto2 ADC MIXR", "ADC2 Switch", "Stereo2 ADC R2 Mux" }, + + { "Sto2 ADC LR MIX", NULL, "Sto2 ADC MIXL" }, + { "Sto2 ADC LR MIX", NULL, "Sto2 ADC MIXR" }, + + { "Stereo2 ADC LR Mux", "L", "Sto2 ADC MIXL" }, + { "Stereo2 ADC LR Mux", "LR", "Sto2 ADC LR MIX" }, + + { "Stereo2 ADC MIXL", NULL, "Stereo2 ADC LR Mux" }, + { "Stereo2 ADC MIXL", NULL, "adc stereo2 filter" }, + { "adc stereo2 filter", NULL, "PLL1", check_sysclk1_source }, + + { "Stereo2 ADC MIXR", NULL, "Sto2 ADC MIXR" }, + { "Stereo2 ADC MIXR", NULL, "adc stereo2 filter" }, + { "adc stereo2 filter", NULL, "PLL1", check_sysclk1_source }, + + { "VAD ADC Mux", "Sto1 ADC L", "Stereo1 ADC MIXL" }, + { "VAD ADC Mux", "Mono ADC L", "Mono ADC MIXL" }, + { "VAD ADC Mux", "Mono ADC R", "Mono ADC MIXR" }, + { "VAD ADC Mux", "Sto2 ADC L", "Sto2 ADC MIXL" }, + + { "VAD_ADC", NULL, "VAD ADC Mux" }, + + { "IF_ADC1", NULL, "Stereo1 ADC MIXL" }, + { "IF_ADC1", NULL, "Stereo1 ADC MIXR" }, + { "IF_ADC2", NULL, "Mono ADC MIXL" }, + { "IF_ADC2", NULL, "Mono ADC MIXR" }, + { "IF_ADC3", NULL, "Stereo2 ADC MIXL" }, + { "IF_ADC3", NULL, "Stereo2 ADC MIXR" }, + + { "IF1 ADC1 IN1 Mux", "IF_ADC1", "IF_ADC1" }, + { "IF1 ADC1 IN1 Mux", "IF1_ADC3", "IF1_ADC3" }, + + { "IF1 ADC1 IN2 Mux", "IF1_ADC1_IN1", "IF1 ADC1 IN1 Mux" }, + { "IF1 ADC1 IN2 Mux", "IF1_ADC4", "IF1_ADC4" }, + + { "IF1 ADC2 IN Mux", "IF_ADC2", "IF_ADC2" }, + { "IF1 ADC2 IN Mux", "VAD_ADC", "VAD_ADC" }, + + { "IF1 ADC2 IN1 Mux", "IF1_ADC2_IN", "IF1 ADC2 IN Mux" }, + { "IF1 ADC2 IN1 Mux", "IF1_ADC4", "IF1_ADC4" }, + + { "IF1_ADC1" , NULL, "IF1 ADC1 IN2 Mux" }, + { "IF1_ADC2" , NULL, "IF1 ADC2 IN1 Mux" }, + + { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL" }, + { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR" }, + { "Stereo2 ADC MIX", NULL, "Sto2 ADC MIXL" }, + { "Stereo2 ADC MIX", NULL, "Sto2 ADC MIXR" }, + + { "RxDP Mux", "IF2 DAC", "IF2 DAC" }, + { "RxDP Mux", "IF1 DAC", "IF1 DAC2" }, + { "RxDP Mux", "STO1 ADC Mixer", "Stereo1 ADC MIX" }, + { "RxDP Mux", "STO2 ADC Mixer", "Stereo2 ADC MIX" }, + { "RxDP Mux", "Mono ADC Mixer L", "Mono ADC MIXL" }, + { "RxDP Mux", "Mono ADC Mixer R", "Mono ADC MIXR" }, + { "RxDP Mux", "DAC1", "DAC MIX" }, + + { "8CH TDM Data", NULL, "Stereo1 ADC MIXL" }, + { "8CH TDM Data", NULL, "Stereo1 ADC MIXR" }, + { "8CH TDM Data", NULL, "Mono ADC MIXL" }, + { "8CH TDM Data", NULL, "Mono ADC MIXR" }, + { "8CH TDM Data", NULL, "Sto2 ADC MIXL" }, + { "8CH TDM Data", NULL, "Sto2 ADC MIXR" }, + { "8CH TDM Data", NULL, "IF2 DAC L" }, + { "8CH TDM Data", NULL, "IF2 DAC R" }, + + { "DSP UL Mux", "Bypass", "8CH TDM Data" }, + { "DSP UL Mux", NULL, "I2S DSP" }, + { "DSP DL Mux", "Bypass", "RxDP Mux" }, + { "DSP DL Mux", NULL, "I2S DSP" }, + + { "TxDP_ADC_L", NULL, "DSP UL Mux" }, + { "TxDP_ADC_R", NULL, "DSP UL Mux" }, + { "TxDC_DAC", NULL, "DSP DL Mux" }, + + { "TxDP_ADC", NULL, "TxDP_ADC_L" }, + { "TxDP_ADC", NULL, "TxDP_ADC_R" }, + + { "IF1 ADC", NULL, "I2S1" }, + { "IF1 ADC", NULL, "IF1_ADC1" }, + { "IF1 ADC", NULL, "IF1_ADC2" }, + { "IF1 ADC", NULL, "IF_ADC3" }, + { "IF1 ADC", NULL, "TxDP_ADC" }, + + { "IF2 ADC Mux", "IF_ADC1", "IF_ADC1" }, + { "IF2 ADC Mux", "IF_ADC2", "IF_ADC2" }, + { "IF2 ADC Mux", "IF_ADC3", "IF_ADC3" }, + { "IF2 ADC Mux", "TxDC_DAC", "TxDC_DAC" }, + { "IF2 ADC Mux", "TxDP_ADC", "TxDP_ADC" }, + { "IF2 ADC Mux", "VAD_ADC", "VAD_ADC" }, + + { "IF2 ADC L", NULL, "IF2 ADC Mux" }, + { "IF2 ADC R", NULL, "IF2 ADC Mux" }, + + { "IF2 ADC", NULL, "I2S2" }, + { "IF2 ADC", NULL, "IF2 ADC L" }, + { "IF2 ADC", NULL, "IF2 ADC R" }, + + { "AIF1TX", NULL, "IF1 ADC" }, + { "AIF2TX", NULL, "IF2 ADC" }, + + { "IF1 DAC1", NULL, "AIF1RX" }, + { "IF1 DAC2", NULL, "AIF1RX" }, + { "IF2 DAC", NULL, "AIF2RX" }, + + { "IF1 DAC1", NULL, "I2S1" }, + { "IF1 DAC2", NULL, "I2S1" }, + { "IF2 DAC", NULL, "I2S2" }, + + { "IF1 DAC2 L", NULL, "IF1 DAC2" }, + { "IF1 DAC2 R", NULL, "IF1 DAC2" }, + { "IF1 DAC1 L", NULL, "IF1 DAC1" }, + { "IF1 DAC1 R", NULL, "IF1 DAC1" }, + { "IF2 DAC L", NULL, "IF2 DAC" }, + { "IF2 DAC R", NULL, "IF2 DAC" }, + + { "DAC1 L Mux", "IF1 DAC", "IF1 DAC1 L" }, + { "DAC1 L Mux", "IF2 DAC", "IF2 DAC L" }, + + { "DAC1 R Mux", "IF1 DAC", "IF1 DAC1 R" }, + { "DAC1 R Mux", "IF2 DAC", "IF2 DAC R" }, + + { "DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL" }, + { "DAC1 MIXL", "DAC1 Switch", "DAC1 L Mux" }, + { "DAC1 MIXL", NULL, "dac stereo1 filter" }, + { "DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR" }, + { "DAC1 MIXR", "DAC1 Switch", "DAC1 R Mux" }, + { "DAC1 MIXR", NULL, "dac stereo1 filter" }, + + { "DAC MIX", NULL, "DAC1 MIXL" }, + { "DAC MIX", NULL, "DAC1 MIXR" }, + + { "Audio DSP", NULL, "DAC1 MIXL" }, + { "Audio DSP", NULL, "DAC1 MIXR" }, + + { "DAC L2 Mux", "IF1 DAC", "IF1 DAC2 L" }, + { "DAC L2 Mux", "IF2 DAC", "IF2 DAC L" }, + { "DAC L2 Mux", "TxDC DAC", "TxDC_DAC" }, + { "DAC L2 Mux", "VAD_ADC", "VAD_ADC" }, + { "DAC L2 Volume", NULL, "DAC L2 Mux" }, + { "DAC L2 Volume", NULL, "dac mono left filter" }, + + { "DAC R2 Mux", "IF1 DAC", "IF1 DAC2 R" }, + { "DAC R2 Mux", "IF2 DAC", "IF2 DAC R" }, + { "DAC R2 Mux", "TxDC DAC", "TxDC_DAC" }, + { "DAC R2 Mux", "TxDP ADC", "TxDP_ADC" }, + { "DAC R2 Volume", NULL, "DAC R2 Mux" }, + { "DAC R2 Volume", NULL, "dac mono right filter" }, + + { "Stereo DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXL", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "Stereo DAC MIXL", NULL, "dac stereo1 filter" }, + { "Stereo DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXR", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "Stereo DAC MIXR", NULL, "dac stereo1 filter" }, + + { "Mono DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Volume" }, + { "Mono DAC MIXL", NULL, "dac mono left filter" }, + { "Mono DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, + { "Mono DAC MIXR", NULL, "dac mono right filter" }, + + { "DAC MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, + { "DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "DAC MIXL", "DAC R2 Switch", "DAC R2 Volume" }, + { "DAC MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, + { "DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, + + { "DAC L1", NULL, "DAC L1 Power" }, + { "DAC L1", NULL, "Stereo DAC MIXL" }, + { "DAC L1", NULL, "PLL1", check_sysclk1_source }, + { "DAC R1", NULL, "DAC R1 Power" }, + { "DAC R1", NULL, "Stereo DAC MIXR" }, + { "DAC R1", NULL, "PLL1", check_sysclk1_source }, + { "DAC L2", NULL, "Mono DAC MIXL" }, + { "DAC L2", NULL, "PLL1", check_sysclk1_source }, + { "DAC R2", NULL, "Mono DAC MIXR" }, + { "DAC R2", NULL, "PLL1", check_sysclk1_source }, + + { "OUT MIXL", "BST1 Switch", "BST1" }, + { "OUT MIXL", "INL Switch", "INL VOL" }, + { "OUT MIXL", "DAC L2 Switch", "DAC L2" }, + { "OUT MIXL", "DAC L1 Switch", "DAC L1" }, + + { "OUT MIXR", "BST2 Switch", "BST2" }, + { "OUT MIXR", "INR Switch", "INR VOL" }, + { "OUT MIXR", "DAC R2 Switch", "DAC R2" }, + { "OUT MIXR", "DAC R1 Switch", "DAC R1" }, + + { "HPOVOL MIXL", "DAC1 Switch", "DAC L1" }, + { "HPOVOL MIXL", "INL Switch", "INL VOL" }, + { "HPOVOL MIXR", "DAC1 Switch", "DAC R1" }, + { "HPOVOL MIXR", "INR Switch", "INR VOL" }, + + { "DAC 2", NULL, "DAC L2" }, + { "DAC 2", NULL, "DAC R2" }, + { "DAC 1", NULL, "DAC L1" }, + { "DAC 1", NULL, "DAC R1" }, + { "HPOVOL", NULL, "HPOVOL MIXL" }, + { "HPOVOL", NULL, "HPOVOL MIXR" }, + { "HPO MIX", "DAC1 Switch", "DAC 1" }, + { "HPO MIX", "HPVOL Switch", "HPOVOL" }, + + { "LOUT MIX", "DAC L1 Switch", "DAC L1" }, + { "LOUT MIX", "DAC R1 Switch", "DAC R1" }, + { "LOUT MIX", "OUTMIX L Switch", "OUT MIXL" }, + { "LOUT MIX", "OUTMIX R Switch", "OUT MIXR" }, + + { "PDM1 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM1 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM1 L Mux", NULL, "PDM1 Power" }, + { "PDM1 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM1 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM1 R Mux", NULL, "PDM1 Power" }, + { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM2 L Mux", NULL, "PDM2 Power" }, + { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM2 R Mux", NULL, "PDM2 Power" }, + + { "HP Amp", NULL, "HPO MIX" }, + { "HP Amp", NULL, "JD Power" }, + { "HP Amp", NULL, "Mic Det Power" }, + { "HPOL", NULL, "HP Amp" }, + { "HPOL", NULL, "HP L Amp" }, + { "HPOL", NULL, "Improve HP Amp Drv" }, + { "HPOR", NULL, "HP Amp" }, + { "HPOR", NULL, "HP R Amp" }, + { "HPOR", NULL, "Improve HP Amp Drv" }, + + { "LOUT Amp", NULL, "LOUT MIX" }, + { "LOUTL", NULL, "LOUT Amp" }, + { "LOUTR", NULL, "LOUT Amp" }, + { "LOUTL", NULL, "Improve HP Amp Drv" }, + { "LOUTR", NULL, "Improve HP Amp Drv" }, + + { "PDM1L", NULL, "PDM1 L Mux" }, + { "PDM1R", NULL, "PDM1 R Mux" }, + { "PDM2L", NULL, "PDM2 L Mux" }, + { "PDM2R", NULL, "PDM2 R Mux" }, +}; + +static const struct snd_soc_dapm_widget rt5670_asrc_widgets[] = { + SND_SOC_DAPM_SUPPLY("ASRC enable", SND_SOC_NOPM, 0, 0, + rt5670_asrc_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), +}; + +static const struct snd_soc_dapm_route rt5670_asrc_routes[] = { + {"I2S1", NULL, "ASRC enable"}, + {"I2S2", NULL, "ASRC enable"}, +}; + +static int get_clk_info(int sclk, int rate, bool asrc_en) +{ + int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + + if (asrc_en) + return 0; + + if (sclk <= 0 || rate <= 0) + return -EINVAL; + + rate = rate << 8; + for (i = 0; i < ARRAY_SIZE(pd); i++) + if (sclk == rate * pd[i]) + return i; + + return -EINVAL; +} + +static int rt5670_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + if (RT5670_AIF2 == dai->id) { + snd_soc_update_bits(codec, RT5670_GPIO_CTRL1, + RT5670_I2S2_PIN_MASK, RT5670_I2S2_PIN_I2S); + } + rt5670->lrck[dai->id] = params_rate(params); + pre_div = get_clk_info(rt5670->sysclk, rt5670->lrck[dai->id], + rt5670->pdata.asrc_en); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting\n"); + return -EINVAL; + } + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + bclk_ms = frame_size > 32 ? 1 : 0; + rt5670->bclk[dai->id] = rt5670->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5670->bclk[dai->id], rt5670->lrck[dai->id]); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= RT5670_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= RT5670_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + val_len |= RT5670_I2S_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5670_AIF1: + mask_clk = RT5670_I2S_BCLK_MS1_MASK | RT5670_I2S_PD1_MASK; + val_clk = bclk_ms << RT5670_I2S_BCLK_MS1_SFT | + pre_div << RT5670_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5670_I2S1_SDP, + RT5670_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5670_ADDA_CLK1, mask_clk, val_clk); + break; + case RT5670_AIF2: + mask_clk = RT5670_I2S_BCLK_MS2_MASK | RT5670_I2S_PD2_MASK; + val_clk = bclk_ms << RT5670_I2S_BCLK_MS2_SFT | + pre_div << RT5670_I2S_PD2_SFT; + snd_soc_update_bits(codec, RT5670_I2S2_SDP, + RT5670_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5670_ADDA_CLK1, mask_clk, val_clk); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5670_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + + if (RT5670_AIF2 == dai->id) { + snd_soc_update_bits(codec, RT5670_GPIO_CTRL1, + RT5670_I2S2_PIN_MASK, RT5670_I2S2_PIN_GPIO); + } + + return 0; +} + +static int rt5670_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5670->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5670_I2S_MS_S; + rt5670->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5670_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5670_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5670_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5670_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5670_AIF1: + snd_soc_update_bits(codec, RT5670_I2S1_SDP, + RT5670_I2S_MS_MASK | RT5670_I2S_BP_MASK | + RT5670_I2S_DF_MASK, reg_val); + break; + case RT5670_AIF2: + snd_soc_update_bits(codec, RT5670_I2S2_SDP, + RT5670_I2S_MS_MASK | RT5670_I2S_BP_MASK | + RT5670_I2S_DF_MASK, reg_val); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5670->sysclk && clk_id == rt5670->sysclk_src) + return 0; + + switch (clk_id) { + case RT5670_SCLK_S_MCLK: + reg_val |= RT5670_SCLK_SRC_MCLK; + break; + case RT5670_SCLK_S_PLL1: + reg_val |= RT5670_SCLK_SRC_PLL1; + break; + case RT5670_SCLK_S_RCCLK: + reg_val |= RT5670_SCLK_SRC_RCCLK; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, reg_val); + rt5670->sysclk = freq; + rt5670->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + return 0; +} + +/** + * rt5670_pll_calc - Calcualte PLL M/N/K code. + * @freq_in: external clock provided to codec. + * @freq_out: target clock which codec works on. + * @pll_code: Pointer to structure with M, N, K and bypass flag. + * + * Calcualte M/N/K code to configure PLL for codec. And K is assigned to 2 + * which make calculation more efficiently. + * + * Returns 0 for success or negative error code. + */ +static int rt5670_pll_calc(const unsigned int freq_in, + const unsigned int freq_out, struct rt5670_pll_code *pll_code) +{ + int max_n = RT5670_PLL_N_MAX, max_m = RT5670_PLL_M_MAX; + int k, n = 0, m = 0, red, n_t, m_t, pll_out, in_t; + int out_t, red_t = abs(freq_out - freq_in); + bool bypass = false; + + if (RT5670_PLL_INP_MAX < freq_in || RT5670_PLL_INP_MIN > freq_in) + return -EINVAL; + + k = 100000000 / freq_out - 2; + if (k > RT5670_PLL_K_MAX) + k = RT5670_PLL_K_MAX; + for (n_t = 0; n_t <= max_n; n_t++) { + in_t = freq_in / (k + 2); + pll_out = freq_out / (n_t + 2); + if (in_t < 0) + continue; + if (in_t == pll_out) { + bypass = true; + n = n_t; + goto code_find; + } + red = abs(in_t - pll_out); + if (red < red_t) { + bypass = true; + n = n_t; + m = m_t; + if (red == 0) + goto code_find; + red_t = red; + } + for (m_t = 0; m_t <= max_m; m_t++) { + out_t = in_t / (m_t + 2); + red = abs(out_t - pll_out); + if (red < red_t) { + bypass = false; + n = n_t; + m = m_t; + if (red == 0) + goto code_find; + red_t = red; + } + } + } + pr_debug("Only get approximation about PLL\n"); + +code_find: + + pll_code->m_bp = bypass; + pll_code->m_code = m; + pll_code->n_code = n; + pll_code->k_code = k; + return 0; +} + +static int rt5670_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + struct rt5670_pll_code *pll_code = &rt5670->pll_code; + int ret; + + if (source == rt5670->pll_src && freq_in == rt5670->pll_in && + freq_out == rt5670->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5670->pll_in = 0; + rt5670->pll_out = 0; + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5670_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_MCLK); + break; + case RT5670_PLL1_S_BCLK1: + case RT5670_PLL1_S_BCLK2: + case RT5670_PLL1_S_BCLK3: + case RT5670_PLL1_S_BCLK4: + switch (dai->id) { + case RT5670_AIF1: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_BCLK1); + break; + case RT5670_AIF2: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_BCLK2); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rt5670_pll_calc(freq_in, freq_out, pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=%d\n", pll_code->m_bp, + (pll_code->m_bp ? 0 : pll_code->m_code), pll_code->n_code, + pll_code->k_code); + + snd_soc_write(codec, RT5670_PLL_CTRL1, + pll_code->n_code << RT5670_PLL_N_SFT | pll_code->k_code); + snd_soc_write(codec, RT5670_PLL_CTRL2, + (pll_code->m_bp ? 0 : pll_code->m_code) << RT5670_PLL_M_SFT | + pll_code->m_bp << RT5670_PLL_M_BP_SFT); + + rt5670->pll_in = freq_in; + rt5670->pll_out = freq_out; + rt5670->pll_src = source; + + return 0; +} + +static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + + if (rx_mask || tx_mask) + val |= (1 << 14); + + switch (slots) { + case 4: + val |= (1 << 12); + break; + case 6: + val |= (2 << 12); + break; + case 8: + val |= (3 << 12); + break; + case 2: + default: + break; + } + + switch (slot_width) { + case 20: + val |= (1 << 10); + break; + case 24: + val |= (2 << 10); + break; + case 32: + val |= (3 << 10); + break; + case 16: + default: + break; + } + + snd_soc_update_bits(codec, RT5670_TDM_CTRL_1, 0x7c00, val); + + return 0; +} + +static int rt5670_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2); + mdelay(10); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_FV1 | RT5670_PWR_FV2, + RT5670_PWR_FV1 | RT5670_PWR_FV2); + snd_soc_update_bits(codec, RT5670_CHARGE_PUMP, + RT5670_OSW_L_MASK | RT5670_OSW_R_MASK, + RT5670_OSW_L_DIS | RT5670_OSW_R_DIS); + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x1); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_LDO_SEL_MASK, 0x3); + } + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, RT5670_PWR_DIG1, 0x0000); + snd_soc_write(codec, RT5670_PWR_DIG2, 0x0001); + snd_soc_write(codec, RT5670_PWR_VOL, 0x0000); + snd_soc_write(codec, RT5670_PWR_MIXER, 0x0001); + snd_soc_write(codec, RT5670_PWR_ANLG1, 0x2800); + snd_soc_write(codec, RT5670_PWR_ANLG2, 0x0004); + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_LDO_SEL_MASK, 0x0); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt5670_probe(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + int ret; + + rt5670->codec = codec; + codec->control_data = rt5670->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + if (rt5670->pdata.asrc_en) { + snd_soc_dapm_new_controls(&codec->dapm, rt5670_asrc_widgets, + ARRAY_SIZE(rt5670_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, rt5670_asrc_routes, + ARRAY_SIZE(rt5670_dapm_routes)); + } + + regmap_write(rt5670->regmap, RT5670_DIG_MISC, 0xc019); + regmap_write(rt5670->regmap, RT5670_ADDA_CLK1, 0x0000); + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL2, 0x0004); + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL1, 0xc900); + regmap_write(rt5670->regmap, RT5670_JD_CTRL3, 0x0088); + + rt5670->dsp_sw = RT5670_DSP_NS; + rt5670_dsp_probe(codec); + + if (rt5670->pdata.jd1_en) { + regmap_update_bits(rt5670->regmap, + RT5670_IRQ_CTRL2, 0x200, 0x200); + snd_soc_dapm_force_enable_pin(&codec->dapm, "JD Power"); + snd_soc_dapm_sync(&codec->dapm); + } + + return 0; +} + +static int rt5670_remove(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + return 0; +} + +#ifdef CONFIG_PM +static int rt5670_suspend(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5670->regmap, true); + regcache_mark_dirty(rt5670->regmap); + return 0; +} + +static int rt5670_resume(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5670->regmap, false); + regcache_sync(rt5670->regmap); + + return 0; +} +#else +#define rt5670_suspend NULL +#define rt5670_resume NULL +#endif + +#define RT5670_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5670_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt5670_aif_dai_ops = { + .hw_params = rt5670_hw_params, + .hw_free = rt5670_hw_free, + .set_fmt = rt5670_set_dai_fmt, + .set_sysclk = rt5670_set_dai_sysclk, + .set_tdm_slot = rt5670_set_tdm_slot, + .set_pll = rt5670_set_dai_pll, +}; + +struct snd_soc_dai_driver rt5670_dai[] = { + { + .name = "rt5670-aif1", + .id = RT5670_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .ops = &rt5670_aif_dai_ops, + }, + { + .name = "rt5670-aif2", + .id = RT5670_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .ops = &rt5670_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5670 = { + .probe = rt5670_probe, + .remove = rt5670_remove, + .suspend = rt5670_suspend, + .resume = rt5670_resume, + .set_bias_level = rt5670_set_bias_level, + .idle_bias_off = true, + .controls = rt5670_snd_controls, + .num_controls = ARRAY_SIZE(rt5670_snd_controls), + .dapm_widgets = rt5670_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5670_dapm_widgets), + .dapm_routes = rt5670_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5670_dapm_routes), +}; + +static const struct regmap_config rt5670_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = RT5670_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5670_ranges) * + RT5670_PR_SPACING), + .volatile_reg = rt5670_volatile_register, + .readable_reg = rt5670_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5670_reg, + .num_reg_defaults = ARRAY_SIZE(rt5670_reg), + .ranges = rt5670_ranges, + .num_ranges = ARRAY_SIZE(rt5670_ranges), +}; + +static const struct i2c_device_id rt5670_i2c_id[] = { + { "rt5670", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id); + +static int rt5670_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5670_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5670_priv *rt5670; + int ret; + unsigned int val; + + rt5670 = devm_kzalloc(&i2c->dev, + sizeof(struct rt5670_priv), + GFP_KERNEL); + if (NULL == rt5670) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5670); + + if (pdata) + rt5670->pdata = *pdata; + + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); + if (IS_ERR(rt5670->regmap)) { + ret = PTR_ERR(rt5670->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5670->regmap, RT5670_VENDOR_ID2, &val); + if ((val != RT5670_DEVICE_ID)) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt5670/72\n", val); + return -ENODEV; + } + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, + RT5670_PWR_HP_L | RT5670_PWR_HP_R | + RT5670_PWR_VREF2, RT5670_PWR_VREF2); + msleep(1000); + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + + ret = regmap_register_patch(rt5670->regmap, init_list, + ARRAY_SIZE(init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5670->pdata.in2_diff) + regmap_update_bits(rt5670->regmap, RT5670_IN2, + RT5670_IN_DF2, RT5670_IN_DF2); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670, + rt5670_dai, ARRAY_SIZE(rt5670_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt5670_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +void rt5670_i2c_shutdown(struct i2c_client *client) +{ + struct rt5670_priv *rt5670 = i2c_get_clientdata(client); + struct snd_soc_codec *codec = rt5670->codec; + + if (codec != NULL) + rt5670_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +struct i2c_driver rt5670_i2c_driver = { + .driver = { + .name = "rt5670", + .owner = THIS_MODULE, + }, + .probe = rt5670_i2c_probe, + .remove = rt5670_i2c_remove, + .shutdown = rt5670_i2c_shutdown, + .id_table = rt5670_i2c_id, +}; + +static int __init rt5670_modinit(void) +{ + return i2c_add_driver(&rt5670_i2c_driver); +} +module_init(rt5670_modinit); + +static void __exit rt5670_modexit(void) +{ + i2c_del_driver(&rt5670_i2c_driver); +} +module_exit(rt5670_modexit); + +MODULE_DESCRIPTION("ASoC RT5670 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h new file mode 100644 index 0000000..3e9b79b --- /dev/null +++ b/sound/soc/codecs/rt5670.h @@ -0,0 +1,1966 @@ +/* + * rt5670.h -- RT5670 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5670_H__ +#define __RT5670_H__ + +#include <sound/rt5670.h> + +/* Info */ +#define RT5670_RESET 0x00 +#define RT5670_VENDOR_ID 0xfd +#define RT5670_VENDOR_ID1 0xfe +#define RT5670_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5670_HP_VOL 0x02 +#define RT5670_LOUT1 0x03 +/* I/O - Input */ +#define RT5670_CJ_CTRL1 0x0a +#define RT5670_CJ_CTRL2 0x0b +#define RT5670_CJ_CTRL3 0x0c +#define RT5670_IN2 0x0e +#define RT5670_INL1_INR1_VOL 0x0f +/* I/O - ADC/DAC/DMIC */ +#define RT5670_DAC1_DIG_VOL 0x19 +#define RT5670_DAC2_DIG_VOL 0x1a +#define RT5670_DAC_CTRL 0x1b +#define RT5670_STO1_ADC_DIG_VOL 0x1c +#define RT5670_MONO_ADC_DIG_VOL 0x1d +#define RT5670_ADC_BST_VOL1 0x1e +#define RT5670_STO2_ADC_DIG_VOL 0x1f +/* Mixer - D-D */ +#define RT5670_ADC_BST_VOL2 0x20 +#define RT5670_STO2_ADC_MIXER 0x26 +#define RT5670_STO1_ADC_MIXER 0x27 +#define RT5670_MONO_ADC_MIXER 0x28 +#define RT5670_AD_DA_MIXER 0x29 +#define RT5670_STO_DAC_MIXER 0x2a +#define RT5670_DD_MIXER 0x2b +#define RT5670_DIG_MIXER 0x2c +#define RT5670_DSP_PATH1 0x2d +#define RT5670_DSP_PATH2 0x2e +#define RT5670_DIG_INF1_DATA 0x2f +#define RT5670_DIG_INF2_DATA 0x30 +/* Mixer - PDM */ +#define RT5670_PDM_OUT_CTRL 0x31 +#define RT5670_PDM_DATA_CTRL1 0x32 +#define RT5670_PDM1_DATA_CTRL2 0x33 +#define RT5670_PDM1_DATA_CTRL3 0x34 +#define RT5670_PDM1_DATA_CTRL4 0x35 +#define RT5670_PDM2_DATA_CTRL2 0x36 +#define RT5670_PDM2_DATA_CTRL3 0x37 +#define RT5670_PDM2_DATA_CTRL4 0x38 +/* Mixer - ADC */ +#define RT5670_REC_L1_MIXER 0x3b +#define RT5670_REC_L2_MIXER 0x3c +#define RT5670_REC_R1_MIXER 0x3d +#define RT5670_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5670_HPO_MIXER 0x45 +#define RT5670_MONO_MIXER 0x4c +#define RT5670_OUT_L1_MIXER 0x4f +#define RT5670_OUT_R1_MIXER 0x52 +#define RT5670_LOUT_MIXER 0x53 +/* Power */ +#define RT5670_PWR_DIG1 0x61 +#define RT5670_PWR_DIG2 0x62 +#define RT5670_PWR_ANLG1 0x63 +#define RT5670_PWR_ANLG2 0x64 +#define RT5670_PWR_MIXER 0x65 +#define RT5670_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5670_PRIV_INDEX 0x6a +#define RT5670_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5670_I2S4_SDP 0x6f +#define RT5670_I2S1_SDP 0x70 +#define RT5670_I2S2_SDP 0x71 +#define RT5670_I2S3_SDP 0x72 +#define RT5670_ADDA_CLK1 0x73 +#define RT5670_ADDA_CLK2 0x74 +#define RT5670_DMIC_CTRL1 0x75 +#define RT5670_DMIC_CTRL2 0x76 +/* Format - TDM Control */ +#define RT5670_TDM_CTRL_1 0x77 +#define RT5670_TDM_CTRL_2 0x78 +#define RT5670_TDM_CTRL_3 0x79 + +/* Function - Analog */ +#define RT5670_DSP_CLK 0x7f +#define RT5670_GLB_CLK 0x80 +#define RT5670_PLL_CTRL1 0x81 +#define RT5670_PLL_CTRL2 0x82 +#define RT5670_ASRC_1 0x83 +#define RT5670_ASRC_2 0x84 +#define RT5670_ASRC_3 0x85 +#define RT5670_ASRC_4 0x86 +#define RT5670_ASRC_5 0x87 +#define RT5670_ASRC_7 0x89 +#define RT5670_ASRC_8 0x8a +#define RT5670_ASRC_9 0x8b +#define RT5670_ASRC_10 0x8c +#define RT5670_ASRC_11 0x8d +#define RT5670_DEPOP_M1 0x8e +#define RT5670_DEPOP_M2 0x8f +#define RT5670_DEPOP_M3 0x90 +#define RT5670_CHARGE_PUMP 0x91 +#define RT5670_MICBIAS 0x93 +#define RT5670_A_JD_CTRL1 0x94 +#define RT5670_A_JD_CTRL2 0x95 +#define RT5670_ASRC_12 0x97 +#define RT5670_ASRC_13 0x98 +#define RT5670_ASRC_14 0x99 +#define RT5670_VAD_CTRL1 0x9a +#define RT5670_VAD_CTRL2 0x9b +#define RT5670_VAD_CTRL3 0x9c +#define RT5670_VAD_CTRL4 0x9d +#define RT5670_VAD_CTRL5 0x9e +/* Function - Digital */ +#define RT5670_ADC_EQ_CTRL1 0xae +#define RT5670_ADC_EQ_CTRL2 0xaf +#define RT5670_EQ_CTRL1 0xb0 +#define RT5670_EQ_CTRL2 0xb1 +#define RT5670_ALC_DRC_CTRL1 0xb2 +#define RT5670_ALC_DRC_CTRL2 0xb3 +#define RT5670_ALC_CTRL_1 0xb4 +#define RT5670_ALC_CTRL_2 0xb5 +#define RT5670_ALC_CTRL_3 0xb6 +#define RT5670_ALC_CTRL_4 0xb7 +#define RT5670_JD_CTRL 0xbb +#define RT5670_IRQ_CTRL1 0xbc +#define RT5670_IRQ_CTRL2 0xbd +#define RT5670_IRQ_CTRL3 0xbe +#define RT5670_INT_IRQ_ST 0xbf +#define RT5670_GPIO_CTRL1 0xc0 +#define RT5670_GPIO_CTRL2 0xc1 +#define RT5670_GPIO_CTRL3 0xc2 +#define RT5670_SCRABBLE_FUN 0xcd +#define RT5670_SCRABBLE_CTRL 0xce +#define RT5670_BASE_BACK 0xcf +#define RT5670_MP3_PLUS1 0xd0 +#define RT5670_MP3_PLUS2 0xd1 +#define RT5670_ADJ_HPF1 0xd3 +#define RT5670_ADJ_HPF2 0xd4 +#define RT5670_HP_CALIB_AMP_DET 0xd6 +#define RT5670_SV_ZCD1 0xd9 +#define RT5670_SV_ZCD2 0xda +#define RT5670_IL_CMD 0xdb +#define RT5670_IL_CMD2 0xdc +#define RT5670_IL_CMD3 0xdd +#define RT5670_DRC_HL_CTRL1 0xe6 +#define RT5670_DRC_HL_CTRL2 0xe7 +#define RT5670_ADC_MONO_HP_CTRL1 0xec +#define RT5670_ADC_MONO_HP_CTRL2 0xed +#define RT5670_ADC_STO2_HP_CTRL1 0xee +#define RT5670_ADC_STO2_HP_CTRL2 0xef +#define RT5670_JD_CTRL3 0xf8 +#define RT5670_JD_CTRL4 0xf9 +/* General Control */ +#define RT5670_DIG_MISC 0xfa +#define RT5670_GEN_CTRL2 0xfb +#define RT5670_GEN_CTRL3 0xfc + + +/* Index of Codec Private Register definition */ +#define RT5670_DIG_VOL 0x00 +#define RT5670_PR_ALC_CTRL_1 0x01 +#define RT5670_PR_ALC_CTRL_2 0x02 +#define RT5670_PR_ALC_CTRL_3 0x03 +#define RT5670_PR_ALC_CTRL_4 0x04 +#define RT5670_PR_ALC_CTRL_5 0x05 +#define RT5670_PR_ALC_CTRL_6 0x06 +#define RT5670_BIAS_CUR1 0x12 +#define RT5670_BIAS_CUR3 0x14 +#define RT5670_CLSD_INT_REG1 0x1c +#define RT5670_MAMP_INT_REG2 0x37 +#define RT5670_CHOP_DAC_ADC 0x3d +#define RT5670_MIXER_INT_REG 0x3f +#define RT5670_3D_SPK 0x63 +#define RT5670_WND_1 0x6c +#define RT5670_WND_2 0x6d +#define RT5670_WND_3 0x6e +#define RT5670_WND_4 0x6f +#define RT5670_WND_5 0x70 +#define RT5670_WND_8 0x73 +#define RT5670_DIP_SPK_INF 0x75 +#define RT5670_HP_DCC_INT1 0x77 +#define RT5670_EQ_BW_LOP 0xa0 +#define RT5670_EQ_GN_LOP 0xa1 +#define RT5670_EQ_FC_BP1 0xa2 +#define RT5670_EQ_BW_BP1 0xa3 +#define RT5670_EQ_GN_BP1 0xa4 +#define RT5670_EQ_FC_BP2 0xa5 +#define RT5670_EQ_BW_BP2 0xa6 +#define RT5670_EQ_GN_BP2 0xa7 +#define RT5670_EQ_FC_BP3 0xa8 +#define RT5670_EQ_BW_BP3 0xa9 +#define RT5670_EQ_GN_BP3 0xaa +#define RT5670_EQ_FC_BP4 0xab +#define RT5670_EQ_BW_BP4 0xac +#define RT5670_EQ_GN_BP4 0xad +#define RT5670_EQ_FC_HIP1 0xae +#define RT5670_EQ_GN_HIP1 0xaf +#define RT5670_EQ_FC_HIP2 0xb0 +#define RT5670_EQ_BW_HIP2 0xb1 +#define RT5670_EQ_GN_HIP2 0xb2 +#define RT5670_EQ_PRE_VOL 0xb3 +#define RT5670_EQ_PST_VOL 0xb4 + + +/* global definition */ +#define RT5670_L_MUTE (0x1 << 15) +#define RT5670_L_MUTE_SFT 15 +#define RT5670_VOL_L_MUTE (0x1 << 14) +#define RT5670_VOL_L_SFT 14 +#define RT5670_R_MUTE (0x1 << 7) +#define RT5670_R_MUTE_SFT 7 +#define RT5670_VOL_R_MUTE (0x1 << 6) +#define RT5670_VOL_R_SFT 6 +#define RT5670_L_VOL_MASK (0x3f << 8) +#define RT5670_L_VOL_SFT 8 +#define RT5670_R_VOL_MASK (0x3f) +#define RT5670_R_VOL_SFT 0 + +/* Combo Jack Control 1 (0x0a) */ +#define RT5670_CBJ_BST1_MASK (0xf << 12) +#define RT5670_CBJ_BST1_SFT (12) +#define RT5670_CBJ_JD_HP_EN (0x1 << 9) +#define RT5670_CBJ_JD_MIC_EN (0x1 << 8) +#define RT5670_CBJ_BST1_EN (0x1 << 2) + +/* Combo Jack Control 1 (0x0b) */ +#define RT5670_CBJ_MN_JD (0x1 << 12) +#define RT5670_CAPLESS_EN (0x1 << 11) +#define RT5670_CBJ_DET_MODE (0x1 << 7) + +/* IN2 Control (0x0e) */ +#define RT5670_BST_MASK1 (0xf<<12) +#define RT5670_BST_SFT1 12 +#define RT5670_BST_MASK2 (0xf<<8) +#define RT5670_BST_SFT2 8 +#define RT5670_IN_DF1 (0x1 << 7) +#define RT5670_IN_SFT1 7 +#define RT5670_IN_DF2 (0x1 << 6) +#define RT5670_IN_SFT2 6 + +/* INL and INR Volume Control (0x0f) */ +#define RT5670_INL_SEL_MASK (0x1 << 15) +#define RT5670_INL_SEL_SFT 15 +#define RT5670_INL_SEL_IN4P (0x0 << 15) +#define RT5670_INL_SEL_MONOP (0x1 << 15) +#define RT5670_INL_VOL_MASK (0x1f << 8) +#define RT5670_INL_VOL_SFT 8 +#define RT5670_INR_SEL_MASK (0x1 << 7) +#define RT5670_INR_SEL_SFT 7 +#define RT5670_INR_SEL_IN4N (0x0 << 7) +#define RT5670_INR_SEL_MONON (0x1 << 7) +#define RT5670_INR_VOL_MASK (0x1f) +#define RT5670_INR_VOL_SFT 0 + +/* Sidetone Control (0x18) */ +#define RT5670_ST_SEL_MASK (0x7 << 9) +#define RT5670_ST_SEL_SFT 9 +#define RT5670_M_ST_DACR2 (0x1 << 8) +#define RT5670_M_ST_DACR2_SFT 8 +#define RT5670_M_ST_DACL2 (0x1 << 7) +#define RT5670_M_ST_DACL2_SFT 7 +#define RT5670_ST_EN (0x1 << 6) +#define RT5670_ST_EN_SFT 6 + +/* DAC1 Digital Volume (0x19) */ +#define RT5670_DAC_L1_VOL_MASK (0xff << 8) +#define RT5670_DAC_L1_VOL_SFT 8 +#define RT5670_DAC_R1_VOL_MASK (0xff) +#define RT5670_DAC_R1_VOL_SFT 0 + +/* DAC2 Digital Volume (0x1a) */ +#define RT5670_DAC_L2_VOL_MASK (0xff << 8) +#define RT5670_DAC_L2_VOL_SFT 8 +#define RT5670_DAC_R2_VOL_MASK (0xff) +#define RT5670_DAC_R2_VOL_SFT 0 + +/* DAC2 Control (0x1b) */ +#define RT5670_M_DAC_L2_VOL (0x1 << 13) +#define RT5670_M_DAC_L2_VOL_SFT 13 +#define RT5670_M_DAC_R2_VOL (0x1 << 12) +#define RT5670_M_DAC_R2_VOL_SFT 12 +#define RT5670_DAC2_L_SEL_MASK (0x7 << 4) +#define RT5670_DAC2_L_SEL_SFT 4 +#define RT5670_DAC2_R_SEL_MASK (0x7 << 0) +#define RT5670_DAC2_R_SEL_SFT 0 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5670_ADC_L_VOL_MASK (0x7f << 8) +#define RT5670_ADC_L_VOL_SFT 8 +#define RT5670_ADC_R_VOL_MASK (0x7f) +#define RT5670_ADC_R_VOL_SFT 0 + +/* Mono ADC Digital Volume Control (0x1d) */ +#define RT5670_MONO_ADC_L_VOL_MASK (0x7f << 8) +#define RT5670_MONO_ADC_L_VOL_SFT 8 +#define RT5670_MONO_ADC_R_VOL_MASK (0x7f) +#define RT5670_MONO_ADC_R_VOL_SFT 0 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5670_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5670_STO1_ADC_L_BST_SFT 14 +#define RT5670_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5670_STO1_ADC_R_BST_SFT 12 +#define RT5670_STO1_ADC_COMP_MASK (0x3 << 10) +#define RT5670_STO1_ADC_COMP_SFT 10 +#define RT5670_STO2_ADC_L_BST_MASK (0x3 << 8) +#define RT5670_STO2_ADC_L_BST_SFT 8 +#define RT5670_STO2_ADC_R_BST_MASK (0x3 << 6) +#define RT5670_STO2_ADC_R_BST_SFT 6 +#define RT5670_STO2_ADC_COMP_MASK (0x3 << 4) +#define RT5670_STO2_ADC_COMP_SFT 4 + +/* Stereo2 ADC Mixer Control (0x26) */ +#define RT5670_STO2_ADC_SRC_MASK (0x1 << 15) +#define RT5670_STO2_ADC_SRC_SFT 15 + +/* Stereo ADC Mixer Control (0x26 0x27) */ +#define RT5670_M_ADC_L1 (0x1 << 14) +#define RT5670_M_ADC_L1_SFT 14 +#define RT5670_M_ADC_L2 (0x1 << 13) +#define RT5670_M_ADC_L2_SFT 13 +#define RT5670_ADC_1_SRC_MASK (0x1 << 12) +#define RT5670_ADC_1_SRC_SFT 12 +#define RT5670_ADC_1_SRC_ADC (0x1 << 12) +#define RT5670_ADC_1_SRC_DACMIX (0x0 << 12) +#define RT5670_ADC_2_SRC_MASK (0x1 << 11) +#define RT5670_ADC_2_SRC_SFT 11 +#define RT5670_ADC_SRC_MASK (0x1 << 10) +#define RT5670_ADC_SRC_SFT 10 +#define RT5670_DMIC_SRC_MASK (0x3 << 8) +#define RT5670_DMIC_SRC_SFT 8 +#define RT5670_M_ADC_R1 (0x1 << 6) +#define RT5670_M_ADC_R1_SFT 6 +#define RT5670_M_ADC_R2 (0x1 << 5) +#define RT5670_M_ADC_R2_SFT 5 +#define RT5670_DMIC3_SRC_MASK (0x1 << 1) +#define RT5670_DMIC3_SRC_SFT 0 + +/* Mono ADC Mixer Control (0x28) */ +#define RT5670_M_MONO_ADC_L1 (0x1 << 14) +#define RT5670_M_MONO_ADC_L1_SFT 14 +#define RT5670_M_MONO_ADC_L2 (0x1 << 13) +#define RT5670_M_MONO_ADC_L2_SFT 13 +#define RT5670_MONO_ADC_L1_SRC_MASK (0x1 << 12) +#define RT5670_MONO_ADC_L1_SRC_SFT 12 +#define RT5670_MONO_ADC_L1_SRC_DACMIXL (0x0 << 12) +#define RT5670_MONO_ADC_L1_SRC_ADCL (0x1 << 12) +#define RT5670_MONO_ADC_L2_SRC_MASK (0x1 << 11) +#define RT5670_MONO_ADC_L2_SRC_SFT 11 +#define RT5670_MONO_ADC_L_SRC_MASK (0x1 << 10) +#define RT5670_MONO_ADC_L_SRC_SFT 10 +#define RT5670_MONO_DMIC_L_SRC_MASK (0x3 << 8) +#define RT5670_MONO_DMIC_L_SRC_SFT 8 +#define RT5670_M_MONO_ADC_R1 (0x1 << 6) +#define RT5670_M_MONO_ADC_R1_SFT 6 +#define RT5670_M_MONO_ADC_R2 (0x1 << 5) +#define RT5670_M_MONO_ADC_R2_SFT 5 +#define RT5670_MONO_ADC_R1_SRC_MASK (0x1 << 4) +#define RT5670_MONO_ADC_R1_SRC_SFT 4 +#define RT5670_MONO_ADC_R1_SRC_ADCR (0x1 << 4) +#define RT5670_MONO_ADC_R1_SRC_DACMIXR (0x0 << 4) +#define RT5670_MONO_ADC_R2_SRC_MASK (0x1 << 3) +#define RT5670_MONO_ADC_R2_SRC_SFT 3 +#define RT5670_MONO_DMIC_R_SRC_MASK (0x3) +#define RT5670_MONO_DMIC_R_SRC_SFT 0 + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5670_M_ADCMIX_L (0x1 << 15) +#define RT5670_M_ADCMIX_L_SFT 15 +#define RT5670_M_DAC1_L (0x1 << 14) +#define RT5670_M_DAC1_L_SFT 14 +#define RT5670_DAC1_R_SEL_MASK (0x3 << 10) +#define RT5670_DAC1_R_SEL_SFT 10 +#define RT5670_DAC1_R_SEL_IF1 (0x0 << 10) +#define RT5670_DAC1_R_SEL_IF2 (0x1 << 10) +#define RT5670_DAC1_R_SEL_IF3 (0x2 << 10) +#define RT5670_DAC1_R_SEL_IF4 (0x3 << 10) +#define RT5670_DAC1_L_SEL_MASK (0x3 << 8) +#define RT5670_DAC1_L_SEL_SFT 8 +#define RT5670_DAC1_L_SEL_IF1 (0x0 << 8) +#define RT5670_DAC1_L_SEL_IF2 (0x1 << 8) +#define RT5670_DAC1_L_SEL_IF3 (0x2 << 8) +#define RT5670_DAC1_L_SEL_IF4 (0x3 << 8) +#define RT5670_M_ADCMIX_R (0x1 << 7) +#define RT5670_M_ADCMIX_R_SFT 7 +#define RT5670_M_DAC1_R (0x1 << 6) +#define RT5670_M_DAC1_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5670_M_DAC_L1 (0x1 << 14) +#define RT5670_M_DAC_L1_SFT 14 +#define RT5670_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5670_DAC_L1_STO_L_VOL_SFT 13 +#define RT5670_M_DAC_L2 (0x1 << 12) +#define RT5670_M_DAC_L2_SFT 12 +#define RT5670_DAC_L2_STO_L_VOL_MASK (0x1 << 11) +#define RT5670_DAC_L2_STO_L_VOL_SFT 11 +#define RT5670_M_DAC_R1_STO_L (0x1 << 9) +#define RT5670_M_DAC_R1_STO_L_SFT 9 +#define RT5670_DAC_R1_STO_L_VOL_MASK (0x1 << 8) +#define RT5670_DAC_R1_STO_L_VOL_SFT 8 +#define RT5670_M_DAC_R1 (0x1 << 6) +#define RT5670_M_DAC_R1_SFT 6 +#define RT5670_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5670_DAC_R1_STO_R_VOL_SFT 5 +#define RT5670_M_DAC_R2 (0x1 << 4) +#define RT5670_M_DAC_R2_SFT 4 +#define RT5670_DAC_R2_STO_R_VOL_MASK (0x1 << 3) +#define RT5670_DAC_R2_STO_R_VOL_SFT 3 +#define RT5670_M_DAC_L1_STO_R (0x1 << 1) +#define RT5670_M_DAC_L1_STO_R_SFT 1 +#define RT5670_DAC_L1_STO_R_VOL_MASK (0x1) +#define RT5670_DAC_L1_STO_R_VOL_SFT 0 + +/* Mono DAC Mixer Control (0x2b) */ +#define RT5670_M_DAC_L1_MONO_L (0x1 << 14) +#define RT5670_M_DAC_L1_MONO_L_SFT 14 +#define RT5670_DAC_L1_MONO_L_VOL_MASK (0x1 << 13) +#define RT5670_DAC_L1_MONO_L_VOL_SFT 13 +#define RT5670_M_DAC_L2_MONO_L (0x1 << 12) +#define RT5670_M_DAC_L2_MONO_L_SFT 12 +#define RT5670_DAC_L2_MONO_L_VOL_MASK (0x1 << 11) +#define RT5670_DAC_L2_MONO_L_VOL_SFT 11 +#define RT5670_M_DAC_R2_MONO_L (0x1 << 10) +#define RT5670_M_DAC_R2_MONO_L_SFT 10 +#define RT5670_DAC_R2_MONO_L_VOL_MASK (0x1 << 9) +#define RT5670_DAC_R2_MONO_L_VOL_SFT 9 +#define RT5670_M_DAC_R1_MONO_R (0x1 << 6) +#define RT5670_M_DAC_R1_MONO_R_SFT 6 +#define RT5670_DAC_R1_MONO_R_VOL_MASK (0x1 << 5) +#define RT5670_DAC_R1_MONO_R_VOL_SFT 5 +#define RT5670_M_DAC_R2_MONO_R (0x1 << 4) +#define RT5670_M_DAC_R2_MONO_R_SFT 4 +#define RT5670_DAC_R2_MONO_R_VOL_MASK (0x1 << 3) +#define RT5670_DAC_R2_MONO_R_VOL_SFT 3 +#define RT5670_M_DAC_L2_MONO_R (0x1 << 2) +#define RT5670_M_DAC_L2_MONO_R_SFT 2 +#define RT5670_DAC_L2_MONO_R_VOL_MASK (0x1 << 1) +#define RT5670_DAC_L2_MONO_R_VOL_SFT 1 + +/* Digital Mixer Control (0x2c) */ +#define RT5670_M_STO_L_DAC_L (0x1 << 15) +#define RT5670_M_STO_L_DAC_L_SFT 15 +#define RT5670_STO_L_DAC_L_VOL_MASK (0x1 << 14) +#define RT5670_STO_L_DAC_L_VOL_SFT 14 +#define RT5670_M_DAC_L2_DAC_L (0x1 << 13) +#define RT5670_M_DAC_L2_DAC_L_SFT 13 +#define RT5670_DAC_L2_DAC_L_VOL_MASK (0x1 << 12) +#define RT5670_DAC_L2_DAC_L_VOL_SFT 12 +#define RT5670_M_STO_R_DAC_R (0x1 << 11) +#define RT5670_M_STO_R_DAC_R_SFT 11 +#define RT5670_STO_R_DAC_R_VOL_MASK (0x1 << 10) +#define RT5670_STO_R_DAC_R_VOL_SFT 10 +#define RT5670_M_DAC_R2_DAC_R (0x1 << 9) +#define RT5670_M_DAC_R2_DAC_R_SFT 9 +#define RT5670_DAC_R2_DAC_R_VOL_MASK (0x1 << 8) +#define RT5670_DAC_R2_DAC_R_VOL_SFT 8 +#define RT5670_M_DAC_R2_DAC_L (0x1 << 7) +#define RT5670_M_DAC_R2_DAC_L_SFT 7 +#define RT5670_DAC_R2_DAC_L_VOL_MASK (0x1 << 6) +#define RT5670_DAC_R2_DAC_L_VOL_SFT 6 +#define RT5670_M_DAC_L2_DAC_R (0x1 << 5) +#define RT5670_M_DAC_L2_DAC_R_SFT 5 +#define RT5670_DAC_L2_DAC_R_VOL_MASK (0x1 << 4) +#define RT5670_DAC_L2_DAC_R_VOL_SFT 4 + +/* DSP Path Control 1 (0x2d) */ +#define RT5670_RXDP_SEL_MASK (0x7 << 13) +#define RT5670_RXDP_SEL_SFT 13 +#define RT5670_RXDP_SRC_MASK (0x3 << 11) +#define RT5670_RXDP_SRC_SFT 11 +#define RT5670_RXDP_SRC_NOR (0x0 << 11) +#define RT5670_RXDP_SRC_DIV2 (0x1 << 11) +#define RT5670_RXDP_SRC_DIV3 (0x2 << 11) +#define RT5670_TXDP_SRC_MASK (0x3 << 4) +#define RT5670_TXDP_SRC_SFT 4 +#define RT5670_TXDP_SRC_NOR (0x0 << 4) +#define RT5670_TXDP_SRC_DIV2 (0x1 << 4) +#define RT5670_TXDP_SRC_DIV3 (0x2 << 4) +#define RT5670_DSP_UL_SEL (0x1 << 1) +#define RT5670_DSP_UL_SFT 1 +#define RT5670_DSP_DL_SEL 0x1 +#define RT5670_DSP_DL_SFT 0 + +/* DSP Path Control 2 (0x2e) */ +#define RT5670_TXDP_L_VOL_MASK (0x7f << 8) +#define RT5670_TXDP_L_VOL_SFT 8 +#define RT5670_TXDP_R_VOL_MASK (0x7f) +#define RT5670_TXDP_R_VOL_SFT 0 + +/* Digital Interface Data Control (0x2f) */ +#define RT5670_IF1_ADC2_IN_SEL (0x1 << 15) +#define RT5670_IF1_ADC2_IN_SFT 15 +#define RT5670_IF2_ADC_IN_MASK (0x7 << 12) +#define RT5670_IF2_ADC_IN_SFT 12 +#define RT5670_IF2_DAC_SEL_MASK (0x3 << 10) +#define RT5670_IF2_DAC_SEL_SFT 10 +#define RT5670_IF2_ADC_SEL_MASK (0x3 << 8) +#define RT5670_IF2_ADC_SEL_SFT 8 + +/* Digital Interface Data Control (0x30) */ +#define RT5670_IF4_ADC_IN_MASK (0x3 << 4) +#define RT5670_IF4_ADC_IN_SFT 4 + +/* PDM Output Control (0x31) */ +#define RT5670_PDM1_L_MASK (0x1 << 15) +#define RT5670_PDM1_L_SFT 15 +#define RT5670_M_PDM1_L (0x1 << 14) +#define RT5670_M_PDM1_L_SFT 14 +#define RT5670_PDM1_R_MASK (0x1 << 13) +#define RT5670_PDM1_R_SFT 13 +#define RT5670_M_PDM1_R (0x1 << 12) +#define RT5670_M_PDM1_R_SFT 12 +#define RT5670_PDM2_L_MASK (0x1 << 11) +#define RT5670_PDM2_L_SFT 11 +#define RT5670_M_PDM2_L (0x1 << 10) +#define RT5670_M_PDM2_L_SFT 10 +#define RT5670_PDM2_R_MASK (0x1 << 9) +#define RT5670_PDM2_R_SFT 9 +#define RT5670_M_PDM2_R (0x1 << 8) +#define RT5670_M_PDM2_R_SFT 8 +#define RT5670_PDM2_BUSY (0x1 << 7) +#define RT5670_PDM1_BUSY (0x1 << 6) +#define RT5670_PDM_PATTERN (0x1 << 5) +#define RT5670_PDM_GAIN (0x1 << 4) +#define RT5670_PDM_DIV_MASK (0x3) + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5670_G_HP_L_RM_L_MASK (0x7 << 13) +#define RT5670_G_HP_L_RM_L_SFT 13 +#define RT5670_G_IN_L_RM_L_MASK (0x7 << 10) +#define RT5670_G_IN_L_RM_L_SFT 10 +#define RT5670_G_BST4_RM_L_MASK (0x7 << 7) +#define RT5670_G_BST4_RM_L_SFT 7 +#define RT5670_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5670_G_BST3_RM_L_SFT 4 +#define RT5670_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5670_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5670_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5670_G_BST1_RM_L_SFT 13 +#define RT5670_M_IN_L_RM_L (0x1 << 5) +#define RT5670_M_IN_L_RM_L_SFT 5 +#define RT5670_M_BST2_RM_L (0x1 << 3) +#define RT5670_M_BST2_RM_L_SFT 3 +#define RT5670_M_BST1_RM_L (0x1 << 1) +#define RT5670_M_BST1_RM_L_SFT 1 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5670_G_HP_R_RM_R_MASK (0x7 << 13) +#define RT5670_G_HP_R_RM_R_SFT 13 +#define RT5670_G_IN_R_RM_R_MASK (0x7 << 10) +#define RT5670_G_IN_R_RM_R_SFT 10 +#define RT5670_G_BST4_RM_R_MASK (0x7 << 7) +#define RT5670_G_BST4_RM_R_SFT 7 +#define RT5670_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5670_G_BST3_RM_R_SFT 4 +#define RT5670_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5670_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5670_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5670_G_BST1_RM_R_SFT 13 +#define RT5670_M_IN_R_RM_R (0x1 << 5) +#define RT5670_M_IN_R_RM_R_SFT 5 +#define RT5670_M_BST2_RM_R (0x1 << 3) +#define RT5670_M_BST2_RM_R_SFT 3 +#define RT5670_M_BST1_RM_R (0x1 << 1) +#define RT5670_M_BST1_RM_R_SFT 1 + +/* HPMIX Control (0x45) */ +#define RT5670_M_DAC2_HM (0x1 << 15) +#define RT5670_M_DAC2_HM_SFT 15 +#define RT5670_M_HPVOL_HM (0x1 << 14) +#define RT5670_M_HPVOL_HM_SFT 14 +#define RT5670_M_DAC1_HM (0x1 << 13) +#define RT5670_M_DAC1_HM_SFT 13 +#define RT5670_G_HPOMIX_MASK (0x1 << 12) +#define RT5670_G_HPOMIX_SFT 12 +#define RT5670_M_INR1_HMR (0x1 << 3) +#define RT5670_M_INR1_HMR_SFT 3 +#define RT5670_M_DACR1_HMR (0x1 << 2) +#define RT5670_M_DACR1_HMR_SFT 2 +#define RT5670_M_INL1_HML (0x1 << 1) +#define RT5670_M_INL1_HML_SFT 1 +#define RT5670_M_DACL1_HML (0x1) +#define RT5670_M_DACL1_HML_SFT 0 + +/* Mono Output Mixer Control (0x4c) */ +#define RT5670_M_DAC_R2_MA (0x1 << 15) +#define RT5670_M_DAC_R2_MA_SFT 15 +#define RT5670_M_DAC_L2_MA (0x1 << 14) +#define RT5670_M_DAC_L2_MA_SFT 14 +#define RT5670_M_OV_R_MM (0x1 << 13) +#define RT5670_M_OV_R_MM_SFT 13 +#define RT5670_M_OV_L_MM (0x1 << 12) +#define RT5670_M_OV_L_MM_SFT 12 +#define RT5670_G_MONOMIX_MASK (0x1 << 10) +#define RT5670_G_MONOMIX_SFT 10 +#define RT5670_M_DAC_R2_MM (0x1 << 9) +#define RT5670_M_DAC_R2_MM_SFT 9 +#define RT5670_M_DAC_L2_MM (0x1 << 8) +#define RT5670_M_DAC_L2_MM_SFT 8 +#define RT5670_M_BST4_MM (0x1 << 7) +#define RT5670_M_BST4_MM_SFT 7 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5670_G_BST3_OM_L_MASK (0x7 << 13) +#define RT5670_G_BST3_OM_L_SFT 13 +#define RT5670_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5670_G_BST2_OM_L_SFT 10 +#define RT5670_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5670_G_BST1_OM_L_SFT 7 +#define RT5670_G_IN_L_OM_L_MASK (0x7 << 4) +#define RT5670_G_IN_L_OM_L_SFT 4 +#define RT5670_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5670_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5670_G_DAC_R2_OM_L_MASK (0x7 << 13) +#define RT5670_G_DAC_R2_OM_L_SFT 13 +#define RT5670_G_DAC_L2_OM_L_MASK (0x7 << 10) +#define RT5670_G_DAC_L2_OM_L_SFT 10 +#define RT5670_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5670_G_DAC_L1_OM_L_SFT 7 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5670_M_BST1_OM_L (0x1 << 5) +#define RT5670_M_BST1_OM_L_SFT 5 +#define RT5670_M_IN_L_OM_L (0x1 << 4) +#define RT5670_M_IN_L_OM_L_SFT 4 +#define RT5670_M_DAC_L2_OM_L (0x1 << 1) +#define RT5670_M_DAC_L2_OM_L_SFT 1 +#define RT5670_M_DAC_L1_OM_L (0x1) +#define RT5670_M_DAC_L1_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5670_G_BST4_OM_R_MASK (0x7 << 13) +#define RT5670_G_BST4_OM_R_SFT 13 +#define RT5670_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5670_G_BST2_OM_R_SFT 10 +#define RT5670_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5670_G_BST1_OM_R_SFT 7 +#define RT5670_G_IN_R_OM_R_MASK (0x7 << 4) +#define RT5670_G_IN_R_OM_R_SFT 4 +#define RT5670_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5670_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5670_G_DAC_L2_OM_R_MASK (0x7 << 13) +#define RT5670_G_DAC_L2_OM_R_SFT 13 +#define RT5670_G_DAC_R2_OM_R_MASK (0x7 << 10) +#define RT5670_G_DAC_R2_OM_R_SFT 10 +#define RT5670_G_DAC_R1_OM_R_MASK (0x7 << 7) +#define RT5670_G_DAC_R1_OM_R_SFT 7 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5670_M_BST2_OM_R (0x1 << 6) +#define RT5670_M_BST2_OM_R_SFT 6 +#define RT5670_M_IN_R_OM_R (0x1 << 4) +#define RT5670_M_IN_R_OM_R_SFT 4 +#define RT5670_M_DAC_R2_OM_R (0x1 << 1) +#define RT5670_M_DAC_R2_OM_R_SFT 1 +#define RT5670_M_DAC_R1_OM_R (0x1) +#define RT5670_M_DAC_R1_OM_R_SFT 0 + +/* LOUT Mixer Control (0x53) */ +#define RT5670_M_DAC_L1_LM (0x1 << 15) +#define RT5670_M_DAC_L1_LM_SFT 15 +#define RT5670_M_DAC_R1_LM (0x1 << 14) +#define RT5670_M_DAC_R1_LM_SFT 14 +#define RT5670_M_OV_L_LM (0x1 << 13) +#define RT5670_M_OV_L_LM_SFT 13 +#define RT5670_M_OV_R_LM (0x1 << 12) +#define RT5670_M_OV_R_LM_SFT 12 +#define RT5670_G_LOUTMIX_MASK (0x1 << 11) +#define RT5670_G_LOUTMIX_SFT 11 + +/* Power Management for Digital 1 (0x61) */ +#define RT5670_PWR_I2S1 (0x1 << 15) +#define RT5670_PWR_I2S1_BIT 15 +#define RT5670_PWR_I2S2 (0x1 << 14) +#define RT5670_PWR_I2S2_BIT 14 +#define RT5670_PWR_DAC_L1 (0x1 << 12) +#define RT5670_PWR_DAC_L1_BIT 12 +#define RT5670_PWR_DAC_R1 (0x1 << 11) +#define RT5670_PWR_DAC_R1_BIT 11 +#define RT5670_PWR_DAC_L2 (0x1 << 7) +#define RT5670_PWR_DAC_L2_BIT 7 +#define RT5670_PWR_DAC_R2 (0x1 << 6) +#define RT5670_PWR_DAC_R2_BIT 6 +#define RT5670_PWR_ADC_L (0x1 << 2) +#define RT5670_PWR_ADC_L_BIT 2 +#define RT5670_PWR_ADC_R (0x1 << 1) +#define RT5670_PWR_ADC_R_BIT 1 +#define RT5670_PWR_CLS_D (0x1) +#define RT5670_PWR_CLS_D_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5670_PWR_ADC_S1F (0x1 << 15) +#define RT5670_PWR_ADC_S1F_BIT 15 +#define RT5670_PWR_ADC_MF_L (0x1 << 14) +#define RT5670_PWR_ADC_MF_L_BIT 14 +#define RT5670_PWR_ADC_MF_R (0x1 << 13) +#define RT5670_PWR_ADC_MF_R_BIT 13 +#define RT5670_PWR_I2S_DSP (0x1 << 12) +#define RT5670_PWR_I2S_DSP_BIT 12 +#define RT5670_PWR_DAC_S1F (0x1 << 11) +#define RT5670_PWR_DAC_S1F_BIT 11 +#define RT5670_PWR_DAC_MF_L (0x1 << 10) +#define RT5670_PWR_DAC_MF_L_BIT 10 +#define RT5670_PWR_DAC_MF_R (0x1 << 9) +#define RT5670_PWR_DAC_MF_R_BIT 9 +#define RT5670_PWR_ADC_S2F (0x1 << 8) +#define RT5670_PWR_ADC_S2F_BIT 8 +#define RT5670_PWR_PDM1 (0x1 << 7) +#define RT5670_PWR_PDM1_BIT 7 +#define RT5670_PWR_PDM2 (0x1 << 6) +#define RT5670_PWR_PDM2_BIT 6 + +/* Power Management for Analog 1 (0x63) */ +#define RT5670_PWR_VREF1 (0x1 << 15) +#define RT5670_PWR_VREF1_BIT 15 +#define RT5670_PWR_FV1 (0x1 << 14) +#define RT5670_PWR_FV1_BIT 14 +#define RT5670_PWR_MB (0x1 << 13) +#define RT5670_PWR_MB_BIT 13 +#define RT5670_PWR_LM (0x1 << 12) +#define RT5670_PWR_LM_BIT 12 +#define RT5670_PWR_BG (0x1 << 11) +#define RT5670_PWR_BG_BIT 11 +#define RT5670_PWR_HP_L (0x1 << 7) +#define RT5670_PWR_HP_L_BIT 7 +#define RT5670_PWR_HP_R (0x1 << 6) +#define RT5670_PWR_HP_R_BIT 6 +#define RT5670_PWR_HA (0x1 << 5) +#define RT5670_PWR_HA_BIT 5 +#define RT5670_PWR_VREF2 (0x1 << 4) +#define RT5670_PWR_VREF2_BIT 4 +#define RT5670_PWR_FV2 (0x1 << 3) +#define RT5670_PWR_FV2_BIT 3 +#define RT5670_LDO_SEL_MASK (0x3) +#define RT5670_LDO_SEL_SFT 0 + +/* Power Management for Analog 2 (0x64) */ +#define RT5670_PWR_BST1 (0x1 << 15) +#define RT5670_PWR_BST1_BIT 15 +#define RT5670_PWR_BST2 (0x1 << 13) +#define RT5670_PWR_BST2_BIT 13 +#define RT5670_PWR_MB1 (0x1 << 11) +#define RT5670_PWR_MB1_BIT 11 +#define RT5670_PWR_MB2 (0x1 << 10) +#define RT5670_PWR_MB2_BIT 10 +#define RT5670_PWR_PLL (0x1 << 9) +#define RT5670_PWR_PLL_BIT 9 +#define RT5670_PWR_BST1_P (0x1 << 6) +#define RT5670_PWR_BST1_P_BIT 6 +#define RT5670_PWR_BST2_P (0x1 << 4) +#define RT5670_PWR_BST2_P_BIT 4 +#define RT5670_PWR_JD1 (0x1 << 2) +#define RT5670_PWR_JD1_BIT 2 +#define RT5670_PWR_JD (0x1 << 1) +#define RT5670_PWR_JD_BIT 1 + +/* Power Management for Mixer (0x65) */ +#define RT5670_PWR_OM_L (0x1 << 15) +#define RT5670_PWR_OM_L_BIT 15 +#define RT5670_PWR_OM_R (0x1 << 14) +#define RT5670_PWR_OM_R_BIT 14 +#define RT5670_PWR_RM_L (0x1 << 11) +#define RT5670_PWR_RM_L_BIT 11 +#define RT5670_PWR_RM_R (0x1 << 10) +#define RT5670_PWR_RM_R_BIT 10 + +/* Power Management for Volume (0x66) */ +#define RT5670_PWR_HV_L (0x1 << 11) +#define RT5670_PWR_HV_L_BIT 11 +#define RT5670_PWR_HV_R (0x1 << 10) +#define RT5670_PWR_HV_R_BIT 10 +#define RT5670_PWR_IN_L (0x1 << 9) +#define RT5670_PWR_IN_L_BIT 9 +#define RT5670_PWR_IN_R (0x1 << 8) +#define RT5670_PWR_IN_R_BIT 8 +#define RT5670_PWR_MIC_DET (0x1 << 5) +#define RT5670_PWR_MIC_DET_BIT 5 + +/* I2S1/2/3 Audio Serial Data Port Control (0x70 0x71 0x72) */ +#define RT5670_I2S_MS_MASK (0x1 << 15) +#define RT5670_I2S_MS_SFT 15 +#define RT5670_I2S_MS_M (0x0 << 15) +#define RT5670_I2S_MS_S (0x1 << 15) +#define RT5670_I2S_IF_MASK (0x7 << 12) +#define RT5670_I2S_IF_SFT 12 +#define RT5670_I2S_O_CP_MASK (0x3 << 10) +#define RT5670_I2S_O_CP_SFT 10 +#define RT5670_I2S_O_CP_OFF (0x0 << 10) +#define RT5670_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5670_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5670_I2S_I_CP_MASK (0x3 << 8) +#define RT5670_I2S_I_CP_SFT 8 +#define RT5670_I2S_I_CP_OFF (0x0 << 8) +#define RT5670_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5670_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5670_I2S_BP_MASK (0x1 << 7) +#define RT5670_I2S_BP_SFT 7 +#define RT5670_I2S_BP_NOR (0x0 << 7) +#define RT5670_I2S_BP_INV (0x1 << 7) +#define RT5670_I2S_DL_MASK (0x3 << 2) +#define RT5670_I2S_DL_SFT 2 +#define RT5670_I2S_DL_16 (0x0 << 2) +#define RT5670_I2S_DL_20 (0x1 << 2) +#define RT5670_I2S_DL_24 (0x2 << 2) +#define RT5670_I2S_DL_8 (0x3 << 2) +#define RT5670_I2S_DF_MASK (0x3) +#define RT5670_I2S_DF_SFT 0 +#define RT5670_I2S_DF_I2S (0x0) +#define RT5670_I2S_DF_LEFT (0x1) +#define RT5670_I2S_DF_PCM_A (0x2) +#define RT5670_I2S_DF_PCM_B (0x3) + +/* I2S2 Audio Serial Data Port Control (0x71) */ +#define RT5670_I2S2_SDI_MASK (0x1 << 6) +#define RT5670_I2S2_SDI_SFT 6 +#define RT5670_I2S2_SDI_I2S1 (0x0 << 6) +#define RT5670_I2S2_SDI_I2S2 (0x1 << 6) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5670_I2S_BCLK_MS1_MASK (0x1 << 15) +#define RT5670_I2S_BCLK_MS1_SFT 15 +#define RT5670_I2S_BCLK_MS1_32 (0x0 << 15) +#define RT5670_I2S_BCLK_MS1_64 (0x1 << 15) +#define RT5670_I2S_PD1_MASK (0x7 << 12) +#define RT5670_I2S_PD1_SFT 12 +#define RT5670_I2S_PD1_1 (0x0 << 12) +#define RT5670_I2S_PD1_2 (0x1 << 12) +#define RT5670_I2S_PD1_3 (0x2 << 12) +#define RT5670_I2S_PD1_4 (0x3 << 12) +#define RT5670_I2S_PD1_6 (0x4 << 12) +#define RT5670_I2S_PD1_8 (0x5 << 12) +#define RT5670_I2S_PD1_12 (0x6 << 12) +#define RT5670_I2S_PD1_16 (0x7 << 12) +#define RT5670_I2S_BCLK_MS2_MASK (0x1 << 11) +#define RT5670_I2S_BCLK_MS2_SFT 11 +#define RT5670_I2S_BCLK_MS2_32 (0x0 << 11) +#define RT5670_I2S_BCLK_MS2_64 (0x1 << 11) +#define RT5670_I2S_PD2_MASK (0x7 << 8) +#define RT5670_I2S_PD2_SFT 8 +#define RT5670_I2S_PD2_1 (0x0 << 8) +#define RT5670_I2S_PD2_2 (0x1 << 8) +#define RT5670_I2S_PD2_3 (0x2 << 8) +#define RT5670_I2S_PD2_4 (0x3 << 8) +#define RT5670_I2S_PD2_6 (0x4 << 8) +#define RT5670_I2S_PD2_8 (0x5 << 8) +#define RT5670_I2S_PD2_12 (0x6 << 8) +#define RT5670_I2S_PD2_16 (0x7 << 8) +#define RT5670_I2S_BCLK_MS3_MASK (0x1 << 7) +#define RT5670_I2S_BCLK_MS3_SFT 7 +#define RT5670_I2S_BCLK_MS3_32 (0x0 << 7) +#define RT5670_I2S_BCLK_MS3_64 (0x1 << 7) +#define RT5670_I2S_PD3_MASK (0x7 << 4) +#define RT5670_I2S_PD3_SFT 4 +#define RT5670_I2S_PD3_1 (0x0 << 4) +#define RT5670_I2S_PD3_2 (0x1 << 4) +#define RT5670_I2S_PD3_3 (0x2 << 4) +#define RT5670_I2S_PD3_4 (0x3 << 4) +#define RT5670_I2S_PD3_6 (0x4 << 4) +#define RT5670_I2S_PD3_8 (0x5 << 4) +#define RT5670_I2S_PD3_12 (0x6 << 4) +#define RT5670_I2S_PD3_16 (0x7 << 4) +#define RT5670_DAC_OSR_MASK (0x3 << 2) +#define RT5670_DAC_OSR_SFT 2 +#define RT5670_DAC_OSR_128 (0x0 << 2) +#define RT5670_DAC_OSR_64 (0x1 << 2) +#define RT5670_DAC_OSR_32 (0x2 << 2) +#define RT5670_DAC_OSR_16 (0x3 << 2) +#define RT5670_ADC_OSR_MASK (0x3) +#define RT5670_ADC_OSR_SFT 0 +#define RT5670_ADC_OSR_128 (0x0) +#define RT5670_ADC_OSR_64 (0x1) +#define RT5670_ADC_OSR_32 (0x2) +#define RT5670_ADC_OSR_16 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5670_DAC_L_OSR_MASK (0x3 << 14) +#define RT5670_DAC_L_OSR_SFT 14 +#define RT5670_DAC_L_OSR_128 (0x0 << 14) +#define RT5670_DAC_L_OSR_64 (0x1 << 14) +#define RT5670_DAC_L_OSR_32 (0x2 << 14) +#define RT5670_DAC_L_OSR_16 (0x3 << 14) +#define RT5670_ADC_R_OSR_MASK (0x3 << 12) +#define RT5670_ADC_R_OSR_SFT 12 +#define RT5670_ADC_R_OSR_128 (0x0 << 12) +#define RT5670_ADC_R_OSR_64 (0x1 << 12) +#define RT5670_ADC_R_OSR_32 (0x2 << 12) +#define RT5670_ADC_R_OSR_16 (0x3 << 12) +#define RT5670_DAHPF_EN (0x1 << 11) +#define RT5670_DAHPF_EN_SFT 11 +#define RT5670_ADHPF_EN (0x1 << 10) +#define RT5670_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5670_DMIC_1_EN_MASK (0x1 << 15) +#define RT5670_DMIC_1_EN_SFT 15 +#define RT5670_DMIC_1_DIS (0x0 << 15) +#define RT5670_DMIC_1_EN (0x1 << 15) +#define RT5670_DMIC_2_EN_MASK (0x1 << 14) +#define RT5670_DMIC_2_EN_SFT 14 +#define RT5670_DMIC_2_DIS (0x0 << 14) +#define RT5670_DMIC_2_EN (0x1 << 14) +#define RT5670_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5670_DMIC_1L_LH_SFT 13 +#define RT5670_DMIC_1L_LH_FALLING (0x0 << 13) +#define RT5670_DMIC_1L_LH_RISING (0x1 << 13) +#define RT5670_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5670_DMIC_1R_LH_SFT 12 +#define RT5670_DMIC_1R_LH_FALLING (0x0 << 12) +#define RT5670_DMIC_1R_LH_RISING (0x1 << 12) +#define RT5670_DMIC_2_DP_MASK (0x1 << 10) +#define RT5670_DMIC_2_DP_SFT 10 +#define RT5670_DMIC_2_DP_GPIO4 (0x0 << 10) +#define RT5670_DMIC_2_DP_IN1N (0x1 << 10) +#define RT5670_DMIC_2L_LH_MASK (0x1 << 9) +#define RT5670_DMIC_2L_LH_SFT 9 +#define RT5670_DMIC_2L_LH_FALLING (0x0 << 9) +#define RT5670_DMIC_2L_LH_RISING (0x1 << 9) +#define RT5670_DMIC_2R_LH_MASK (0x1 << 8) +#define RT5670_DMIC_2R_LH_SFT 8 +#define RT5670_DMIC_2R_LH_FALLING (0x0 << 8) +#define RT5670_DMIC_2R_LH_RISING (0x1 << 8) +#define RT5670_DMIC_CLK_MASK (0x7 << 5) +#define RT5670_DMIC_CLK_SFT 5 +#define RT5670_DMIC_3_EN_MASK (0x1 << 4) +#define RT5670_DMIC_3_EN_SFT 4 +#define RT5670_DMIC_3_DIS (0x0 << 4) +#define RT5670_DMIC_3_EN (0x1 << 4) +#define RT5670_DMIC_1_DP_MASK (0x3 << 0) +#define RT5670_DMIC_1_DP_SFT 0 +#define RT5670_DMIC_1_DP_GPIO6 (0x0 << 0) +#define RT5670_DMIC_1_DP_IN2P (0x1 << 0) +#define RT5670_DMIC_1_DP_GPIO7 (0x2 << 0) + +/* Global Clock Control (0x80) */ +#define RT5670_SCLK_SRC_MASK (0x3 << 14) +#define RT5670_SCLK_SRC_SFT 14 +#define RT5670_SCLK_SRC_MCLK (0x0 << 14) +#define RT5670_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5670_SCLK_SRC_RCCLK (0x2 << 14) /* 15MHz */ +#define RT5670_PLL1_SRC_MASK (0x3 << 12) +#define RT5670_PLL1_SRC_SFT 12 +#define RT5670_PLL1_SRC_MCLK (0x0 << 12) +#define RT5670_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5670_PLL1_SRC_BCLK2 (0x2 << 12) +#define RT5670_PLL1_SRC_BCLK3 (0x3 << 12) +#define RT5670_PLL1_PD_MASK (0x1 << 3) +#define RT5670_PLL1_PD_SFT 3 +#define RT5670_PLL1_PD_1 (0x0 << 3) +#define RT5670_PLL1_PD_2 (0x1 << 3) + +#define RT5670_PLL_INP_MAX 40000000 +#define RT5670_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5670_PLL_N_MAX 0x1ff +#define RT5670_PLL_N_MASK (RT5670_PLL_N_MAX << 7) +#define RT5670_PLL_N_SFT 7 +#define RT5670_PLL_K_MAX 0x1f +#define RT5670_PLL_K_MASK (RT5670_PLL_K_MAX) +#define RT5670_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5670_PLL_M_MAX 0xf +#define RT5670_PLL_M_MASK (RT5670_PLL_M_MAX << 12) +#define RT5670_PLL_M_SFT 12 +#define RT5670_PLL_M_BP (0x1 << 11) +#define RT5670_PLL_M_BP_SFT 11 + +/* ASRC Control 1 (0x83) */ +#define RT5670_STO_T_MASK (0x1 << 15) +#define RT5670_STO_T_SFT 15 +#define RT5670_STO_T_SCLK (0x0 << 15) +#define RT5670_STO_T_LRCK1 (0x1 << 15) +#define RT5670_M1_T_MASK (0x1 << 14) +#define RT5670_M1_T_SFT 14 +#define RT5670_M1_T_I2S2 (0x0 << 14) +#define RT5670_M1_T_I2S2_D3 (0x1 << 14) +#define RT5670_I2S2_F_MASK (0x1 << 12) +#define RT5670_I2S2_F_SFT 12 +#define RT5670_I2S2_F_I2S2_D2 (0x0 << 12) +#define RT5670_I2S2_F_I2S1_TCLK (0x1 << 12) +#define RT5670_DMIC_1_M_MASK (0x1 << 9) +#define RT5670_DMIC_1_M_SFT 9 +#define RT5670_DMIC_1_M_NOR (0x0 << 9) +#define RT5670_DMIC_1_M_ASYN (0x1 << 9) +#define RT5670_DMIC_2_M_MASK (0x1 << 8) +#define RT5670_DMIC_2_M_SFT 8 +#define RT5670_DMIC_2_M_NOR (0x0 << 8) +#define RT5670_DMIC_2_M_ASYN (0x1 << 8) + +/* ASRC Control 2 (0x84) */ +#define RT5670_MDA_L_M_MASK (0x1 << 15) +#define RT5670_MDA_L_M_SFT 15 +#define RT5670_MDA_L_M_NOR (0x0 << 15) +#define RT5670_MDA_L_M_ASYN (0x1 << 15) +#define RT5670_MDA_R_M_MASK (0x1 << 14) +#define RT5670_MDA_R_M_SFT 14 +#define RT5670_MDA_R_M_NOR (0x0 << 14) +#define RT5670_MDA_R_M_ASYN (0x1 << 14) +#define RT5670_MAD_L_M_MASK (0x1 << 13) +#define RT5670_MAD_L_M_SFT 13 +#define RT5670_MAD_L_M_NOR (0x0 << 13) +#define RT5670_MAD_L_M_ASYN (0x1 << 13) +#define RT5670_MAD_R_M_MASK (0x1 << 12) +#define RT5670_MAD_R_M_SFT 12 +#define RT5670_MAD_R_M_NOR (0x0 << 12) +#define RT5670_MAD_R_M_ASYN (0x1 << 12) +#define RT5670_ADC_M_MASK (0x1 << 11) +#define RT5670_ADC_M_SFT 11 +#define RT5670_ADC_M_NOR (0x0 << 11) +#define RT5670_ADC_M_ASYN (0x1 << 11) +#define RT5670_STO_DAC_M_MASK (0x1 << 5) +#define RT5670_STO_DAC_M_SFT 5 +#define RT5670_STO_DAC_M_NOR (0x0 << 5) +#define RT5670_STO_DAC_M_ASYN (0x1 << 5) +#define RT5670_I2S1_R_D_MASK (0x1 << 4) +#define RT5670_I2S1_R_D_SFT 4 +#define RT5670_I2S1_R_D_DIS (0x0 << 4) +#define RT5670_I2S1_R_D_EN (0x1 << 4) +#define RT5670_I2S2_R_D_MASK (0x1 << 3) +#define RT5670_I2S2_R_D_SFT 3 +#define RT5670_I2S2_R_D_DIS (0x0 << 3) +#define RT5670_I2S2_R_D_EN (0x1 << 3) +#define RT5670_PRE_SCLK_MASK (0x3) +#define RT5670_PRE_SCLK_SFT 0 +#define RT5670_PRE_SCLK_512 (0x0) +#define RT5670_PRE_SCLK_1024 (0x1) +#define RT5670_PRE_SCLK_2048 (0x2) + +/* ASRC Control 3 (0x85) */ +#define RT5670_I2S1_RATE_MASK (0xf << 12) +#define RT5670_I2S1_RATE_SFT 12 +#define RT5670_I2S2_RATE_MASK (0xf << 8) +#define RT5670_I2S2_RATE_SFT 8 + +/* ASRC Control 4 (0x89) */ +#define RT5670_I2S1_PD_MASK (0x7 << 12) +#define RT5670_I2S1_PD_SFT 12 +#define RT5670_I2S2_PD_MASK (0x7 << 8) +#define RT5670_I2S2_PD_SFT 8 + +/* HPOUT Over Current Detection (0x8b) */ +#define RT5670_HP_OVCD_MASK (0x1 << 10) +#define RT5670_HP_OVCD_SFT 10 +#define RT5670_HP_OVCD_DIS (0x0 << 10) +#define RT5670_HP_OVCD_EN (0x1 << 10) +#define RT5670_HP_OC_TH_MASK (0x3 << 8) +#define RT5670_HP_OC_TH_SFT 8 +#define RT5670_HP_OC_TH_90 (0x0 << 8) +#define RT5670_HP_OC_TH_105 (0x1 << 8) +#define RT5670_HP_OC_TH_120 (0x2 << 8) +#define RT5670_HP_OC_TH_135 (0x3 << 8) + +/* Class D Over Current Control (0x8c) */ +#define RT5670_CLSD_OC_MASK (0x1 << 9) +#define RT5670_CLSD_OC_SFT 9 +#define RT5670_CLSD_OC_PU (0x0 << 9) +#define RT5670_CLSD_OC_PD (0x1 << 9) +#define RT5670_AUTO_PD_MASK (0x1 << 8) +#define RT5670_AUTO_PD_SFT 8 +#define RT5670_AUTO_PD_DIS (0x0 << 8) +#define RT5670_AUTO_PD_EN (0x1 << 8) +#define RT5670_CLSD_OC_TH_MASK (0x3f) +#define RT5670_CLSD_OC_TH_SFT 0 + +/* Class D Output Control (0x8d) */ +#define RT5670_CLSD_RATIO_MASK (0xf << 12) +#define RT5670_CLSD_RATIO_SFT 12 +#define RT5670_CLSD_OM_MASK (0x1 << 11) +#define RT5670_CLSD_OM_SFT 11 +#define RT5670_CLSD_OM_MONO (0x0 << 11) +#define RT5670_CLSD_OM_STO (0x1 << 11) +#define RT5670_CLSD_SCH_MASK (0x1 << 10) +#define RT5670_CLSD_SCH_SFT 10 +#define RT5670_CLSD_SCH_L (0x0 << 10) +#define RT5670_CLSD_SCH_S (0x1 << 10) + +/* Depop Mode Control 1 (0x8e) */ +#define RT5670_SMT_TRIG_MASK (0x1 << 15) +#define RT5670_SMT_TRIG_SFT 15 +#define RT5670_SMT_TRIG_DIS (0x0 << 15) +#define RT5670_SMT_TRIG_EN (0x1 << 15) +#define RT5670_HP_L_SMT_MASK (0x1 << 9) +#define RT5670_HP_L_SMT_SFT 9 +#define RT5670_HP_L_SMT_DIS (0x0 << 9) +#define RT5670_HP_L_SMT_EN (0x1 << 9) +#define RT5670_HP_R_SMT_MASK (0x1 << 8) +#define RT5670_HP_R_SMT_SFT 8 +#define RT5670_HP_R_SMT_DIS (0x0 << 8) +#define RT5670_HP_R_SMT_EN (0x1 << 8) +#define RT5670_HP_CD_PD_MASK (0x1 << 7) +#define RT5670_HP_CD_PD_SFT 7 +#define RT5670_HP_CD_PD_DIS (0x0 << 7) +#define RT5670_HP_CD_PD_EN (0x1 << 7) +#define RT5670_RSTN_MASK (0x1 << 6) +#define RT5670_RSTN_SFT 6 +#define RT5670_RSTN_DIS (0x0 << 6) +#define RT5670_RSTN_EN (0x1 << 6) +#define RT5670_RSTP_MASK (0x1 << 5) +#define RT5670_RSTP_SFT 5 +#define RT5670_RSTP_DIS (0x0 << 5) +#define RT5670_RSTP_EN (0x1 << 5) +#define RT5670_HP_CO_MASK (0x1 << 4) +#define RT5670_HP_CO_SFT 4 +#define RT5670_HP_CO_DIS (0x0 << 4) +#define RT5670_HP_CO_EN (0x1 << 4) +#define RT5670_HP_CP_MASK (0x1 << 3) +#define RT5670_HP_CP_SFT 3 +#define RT5670_HP_CP_PD (0x0 << 3) +#define RT5670_HP_CP_PU (0x1 << 3) +#define RT5670_HP_SG_MASK (0x1 << 2) +#define RT5670_HP_SG_SFT 2 +#define RT5670_HP_SG_DIS (0x0 << 2) +#define RT5670_HP_SG_EN (0x1 << 2) +#define RT5670_HP_DP_MASK (0x1 << 1) +#define RT5670_HP_DP_SFT 1 +#define RT5670_HP_DP_PD (0x0 << 1) +#define RT5670_HP_DP_PU (0x1 << 1) +#define RT5670_HP_CB_MASK (0x1) +#define RT5670_HP_CB_SFT 0 +#define RT5670_HP_CB_PD (0x0) +#define RT5670_HP_CB_PU (0x1) + +/* Depop Mode Control 2 (0x8f) */ +#define RT5670_DEPOP_MASK (0x1 << 13) +#define RT5670_DEPOP_SFT 13 +#define RT5670_DEPOP_AUTO (0x0 << 13) +#define RT5670_DEPOP_MAN (0x1 << 13) +#define RT5670_RAMP_MASK (0x1 << 12) +#define RT5670_RAMP_SFT 12 +#define RT5670_RAMP_DIS (0x0 << 12) +#define RT5670_RAMP_EN (0x1 << 12) +#define RT5670_BPS_MASK (0x1 << 11) +#define RT5670_BPS_SFT 11 +#define RT5670_BPS_DIS (0x0 << 11) +#define RT5670_BPS_EN (0x1 << 11) +#define RT5670_FAST_UPDN_MASK (0x1 << 10) +#define RT5670_FAST_UPDN_SFT 10 +#define RT5670_FAST_UPDN_DIS (0x0 << 10) +#define RT5670_FAST_UPDN_EN (0x1 << 10) +#define RT5670_MRES_MASK (0x3 << 8) +#define RT5670_MRES_SFT 8 +#define RT5670_MRES_15MO (0x0 << 8) +#define RT5670_MRES_25MO (0x1 << 8) +#define RT5670_MRES_35MO (0x2 << 8) +#define RT5670_MRES_45MO (0x3 << 8) +#define RT5670_VLO_MASK (0x1 << 7) +#define RT5670_VLO_SFT 7 +#define RT5670_VLO_3V (0x0 << 7) +#define RT5670_VLO_32V (0x1 << 7) +#define RT5670_DIG_DP_MASK (0x1 << 6) +#define RT5670_DIG_DP_SFT 6 +#define RT5670_DIG_DP_DIS (0x0 << 6) +#define RT5670_DIG_DP_EN (0x1 << 6) +#define RT5670_DP_TH_MASK (0x3 << 4) +#define RT5670_DP_TH_SFT 4 + +/* Depop Mode Control 3 (0x90) */ +#define RT5670_CP_SYS_MASK (0x7 << 12) +#define RT5670_CP_SYS_SFT 12 +#define RT5670_CP_FQ1_MASK (0x7 << 8) +#define RT5670_CP_FQ1_SFT 8 +#define RT5670_CP_FQ2_MASK (0x7 << 4) +#define RT5670_CP_FQ2_SFT 4 +#define RT5670_CP_FQ3_MASK (0x7) +#define RT5670_CP_FQ3_SFT 0 +#define RT5670_CP_FQ_1_5_KHZ 0 +#define RT5670_CP_FQ_3_KHZ 1 +#define RT5670_CP_FQ_6_KHZ 2 +#define RT5670_CP_FQ_12_KHZ 3 +#define RT5670_CP_FQ_24_KHZ 4 +#define RT5670_CP_FQ_48_KHZ 5 +#define RT5670_CP_FQ_96_KHZ 6 +#define RT5670_CP_FQ_192_KHZ 7 + +/* HPOUT charge pump (0x91) */ +#define RT5670_OSW_L_MASK (0x1 << 11) +#define RT5670_OSW_L_SFT 11 +#define RT5670_OSW_L_DIS (0x0 << 11) +#define RT5670_OSW_L_EN (0x1 << 11) +#define RT5670_OSW_R_MASK (0x1 << 10) +#define RT5670_OSW_R_SFT 10 +#define RT5670_OSW_R_DIS (0x0 << 10) +#define RT5670_OSW_R_EN (0x1 << 10) +#define RT5670_PM_HP_MASK (0x3 << 8) +#define RT5670_PM_HP_SFT 8 +#define RT5670_PM_HP_LV (0x0 << 8) +#define RT5670_PM_HP_MV (0x1 << 8) +#define RT5670_PM_HP_HV (0x2 << 8) +#define RT5670_IB_HP_MASK (0x3 << 6) +#define RT5670_IB_HP_SFT 6 +#define RT5670_IB_HP_125IL (0x0 << 6) +#define RT5670_IB_HP_25IL (0x1 << 6) +#define RT5670_IB_HP_5IL (0x2 << 6) +#define RT5670_IB_HP_1IL (0x3 << 6) + +/* PV detection and SPK gain control (0x92) */ +#define RT5670_PVDD_DET_MASK (0x1 << 15) +#define RT5670_PVDD_DET_SFT 15 +#define RT5670_PVDD_DET_DIS (0x0 << 15) +#define RT5670_PVDD_DET_EN (0x1 << 15) +#define RT5670_SPK_AG_MASK (0x1 << 14) +#define RT5670_SPK_AG_SFT 14 +#define RT5670_SPK_AG_DIS (0x0 << 14) +#define RT5670_SPK_AG_EN (0x1 << 14) + +/* Micbias Control (0x93) */ +#define RT5670_MIC1_BS_MASK (0x1 << 15) +#define RT5670_MIC1_BS_SFT 15 +#define RT5670_MIC1_BS_9AV (0x0 << 15) +#define RT5670_MIC1_BS_75AV (0x1 << 15) +#define RT5670_MIC2_BS_MASK (0x1 << 14) +#define RT5670_MIC2_BS_SFT 14 +#define RT5670_MIC2_BS_9AV (0x0 << 14) +#define RT5670_MIC2_BS_75AV (0x1 << 14) +#define RT5670_MIC1_CLK_MASK (0x1 << 13) +#define RT5670_MIC1_CLK_SFT 13 +#define RT5670_MIC1_CLK_DIS (0x0 << 13) +#define RT5670_MIC1_CLK_EN (0x1 << 13) +#define RT5670_MIC2_CLK_MASK (0x1 << 12) +#define RT5670_MIC2_CLK_SFT 12 +#define RT5670_MIC2_CLK_DIS (0x0 << 12) +#define RT5670_MIC2_CLK_EN (0x1 << 12) +#define RT5670_MIC1_OVCD_MASK (0x1 << 11) +#define RT5670_MIC1_OVCD_SFT 11 +#define RT5670_MIC1_OVCD_DIS (0x0 << 11) +#define RT5670_MIC1_OVCD_EN (0x1 << 11) +#define RT5670_MIC1_OVTH_MASK (0x3 << 9) +#define RT5670_MIC1_OVTH_SFT 9 +#define RT5670_MIC1_OVTH_600UA (0x0 << 9) +#define RT5670_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5670_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5670_MIC2_OVCD_MASK (0x1 << 8) +#define RT5670_MIC2_OVCD_SFT 8 +#define RT5670_MIC2_OVCD_DIS (0x0 << 8) +#define RT5670_MIC2_OVCD_EN (0x1 << 8) +#define RT5670_MIC2_OVTH_MASK (0x3 << 6) +#define RT5670_MIC2_OVTH_SFT 6 +#define RT5670_MIC2_OVTH_600UA (0x0 << 6) +#define RT5670_MIC2_OVTH_1500UA (0x1 << 6) +#define RT5670_MIC2_OVTH_2000UA (0x2 << 6) +#define RT5670_PWR_MB_MASK (0x1 << 5) +#define RT5670_PWR_MB_SFT 5 +#define RT5670_PWR_MB_PD (0x0 << 5) +#define RT5670_PWR_MB_PU (0x1 << 5) +#define RT5670_PWR_CLK25M_MASK (0x1 << 4) +#define RT5670_PWR_CLK25M_SFT 4 +#define RT5670_PWR_CLK25M_PD (0x0 << 4) +#define RT5670_PWR_CLK25M_PU (0x1 << 4) + +/* VAD Control 4 (0x9d) */ +#define RT5670_VAD_SEL_MASK (0x3 << 8) +#define RT5670_VAD_SEL_SFT 8 + +/* EQ Control 1 (0xb0) */ +#define RT5670_EQ_SRC_MASK (0x1 << 15) +#define RT5670_EQ_SRC_SFT 15 +#define RT5670_EQ_SRC_DAC (0x0 << 15) +#define RT5670_EQ_SRC_ADC (0x1 << 15) +#define RT5670_EQ_UPD (0x1 << 14) +#define RT5670_EQ_UPD_BIT 14 +#define RT5670_EQ_CD_MASK (0x1 << 13) +#define RT5670_EQ_CD_SFT 13 +#define RT5670_EQ_CD_DIS (0x0 << 13) +#define RT5670_EQ_CD_EN (0x1 << 13) +#define RT5670_EQ_DITH_MASK (0x3 << 8) +#define RT5670_EQ_DITH_SFT 8 +#define RT5670_EQ_DITH_NOR (0x0 << 8) +#define RT5670_EQ_DITH_LSB (0x1 << 8) +#define RT5670_EQ_DITH_LSB_1 (0x2 << 8) +#define RT5670_EQ_DITH_LSB_2 (0x3 << 8) + +/* EQ Control 2 (0xb1) */ +#define RT5670_EQ_HPF1_M_MASK (0x1 << 8) +#define RT5670_EQ_HPF1_M_SFT 8 +#define RT5670_EQ_HPF1_M_HI (0x0 << 8) +#define RT5670_EQ_HPF1_M_1ST (0x1 << 8) +#define RT5670_EQ_LPF1_M_MASK (0x1 << 7) +#define RT5670_EQ_LPF1_M_SFT 7 +#define RT5670_EQ_LPF1_M_LO (0x0 << 7) +#define RT5670_EQ_LPF1_M_1ST (0x1 << 7) +#define RT5670_EQ_HPF2_MASK (0x1 << 6) +#define RT5670_EQ_HPF2_SFT 6 +#define RT5670_EQ_HPF2_DIS (0x0 << 6) +#define RT5670_EQ_HPF2_EN (0x1 << 6) +#define RT5670_EQ_HPF1_MASK (0x1 << 5) +#define RT5670_EQ_HPF1_SFT 5 +#define RT5670_EQ_HPF1_DIS (0x0 << 5) +#define RT5670_EQ_HPF1_EN (0x1 << 5) +#define RT5670_EQ_BPF4_MASK (0x1 << 4) +#define RT5670_EQ_BPF4_SFT 4 +#define RT5670_EQ_BPF4_DIS (0x0 << 4) +#define RT5670_EQ_BPF4_EN (0x1 << 4) +#define RT5670_EQ_BPF3_MASK (0x1 << 3) +#define RT5670_EQ_BPF3_SFT 3 +#define RT5670_EQ_BPF3_DIS (0x0 << 3) +#define RT5670_EQ_BPF3_EN (0x1 << 3) +#define RT5670_EQ_BPF2_MASK (0x1 << 2) +#define RT5670_EQ_BPF2_SFT 2 +#define RT5670_EQ_BPF2_DIS (0x0 << 2) +#define RT5670_EQ_BPF2_EN (0x1 << 2) +#define RT5670_EQ_BPF1_MASK (0x1 << 1) +#define RT5670_EQ_BPF1_SFT 1 +#define RT5670_EQ_BPF1_DIS (0x0 << 1) +#define RT5670_EQ_BPF1_EN (0x1 << 1) +#define RT5670_EQ_LPF_MASK (0x1) +#define RT5670_EQ_LPF_SFT 0 +#define RT5670_EQ_LPF_DIS (0x0) +#define RT5670_EQ_LPF_EN (0x1) +#define RT5670_EQ_CTRL_MASK (0x7f) + +/* Memory Test (0xb2) */ +#define RT5670_MT_MASK (0x1 << 15) +#define RT5670_MT_SFT 15 +#define RT5670_MT_DIS (0x0 << 15) +#define RT5670_MT_EN (0x1 << 15) + +/* DRC/AGC Control 1 (0xb4) */ +#define RT5670_DRC_AGC_P_MASK (0x1 << 15) +#define RT5670_DRC_AGC_P_SFT 15 +#define RT5670_DRC_AGC_P_DAC (0x0 << 15) +#define RT5670_DRC_AGC_P_ADC (0x1 << 15) +#define RT5670_DRC_AGC_MASK (0x1 << 14) +#define RT5670_DRC_AGC_SFT 14 +#define RT5670_DRC_AGC_DIS (0x0 << 14) +#define RT5670_DRC_AGC_EN (0x1 << 14) +#define RT5670_DRC_AGC_UPD (0x1 << 13) +#define RT5670_DRC_AGC_UPD_BIT 13 +#define RT5670_DRC_AGC_AR_MASK (0x1f << 8) +#define RT5670_DRC_AGC_AR_SFT 8 +#define RT5670_DRC_AGC_R_MASK (0x7 << 5) +#define RT5670_DRC_AGC_R_SFT 5 +#define RT5670_DRC_AGC_R_48K (0x1 << 5) +#define RT5670_DRC_AGC_R_96K (0x2 << 5) +#define RT5670_DRC_AGC_R_192K (0x3 << 5) +#define RT5670_DRC_AGC_R_441K (0x5 << 5) +#define RT5670_DRC_AGC_R_882K (0x6 << 5) +#define RT5670_DRC_AGC_R_1764K (0x7 << 5) +#define RT5670_DRC_AGC_RC_MASK (0x1f) +#define RT5670_DRC_AGC_RC_SFT 0 + +/* DRC/AGC Control 2 (0xb5) */ +#define RT5670_DRC_AGC_POB_MASK (0x3f << 8) +#define RT5670_DRC_AGC_POB_SFT 8 +#define RT5670_DRC_AGC_CP_MASK (0x1 << 7) +#define RT5670_DRC_AGC_CP_SFT 7 +#define RT5670_DRC_AGC_CP_DIS (0x0 << 7) +#define RT5670_DRC_AGC_CP_EN (0x1 << 7) +#define RT5670_DRC_AGC_CPR_MASK (0x3 << 5) +#define RT5670_DRC_AGC_CPR_SFT 5 +#define RT5670_DRC_AGC_CPR_1_1 (0x0 << 5) +#define RT5670_DRC_AGC_CPR_1_2 (0x1 << 5) +#define RT5670_DRC_AGC_CPR_1_3 (0x2 << 5) +#define RT5670_DRC_AGC_CPR_1_4 (0x3 << 5) +#define RT5670_DRC_AGC_PRB_MASK (0x1f) +#define RT5670_DRC_AGC_PRB_SFT 0 + +/* DRC/AGC Control 3 (0xb6) */ +#define RT5670_DRC_AGC_NGB_MASK (0xf << 12) +#define RT5670_DRC_AGC_NGB_SFT 12 +#define RT5670_DRC_AGC_TAR_MASK (0x1f << 7) +#define RT5670_DRC_AGC_TAR_SFT 7 +#define RT5670_DRC_AGC_NG_MASK (0x1 << 6) +#define RT5670_DRC_AGC_NG_SFT 6 +#define RT5670_DRC_AGC_NG_DIS (0x0 << 6) +#define RT5670_DRC_AGC_NG_EN (0x1 << 6) +#define RT5670_DRC_AGC_NGH_MASK (0x1 << 5) +#define RT5670_DRC_AGC_NGH_SFT 5 +#define RT5670_DRC_AGC_NGH_DIS (0x0 << 5) +#define RT5670_DRC_AGC_NGH_EN (0x1 << 5) +#define RT5670_DRC_AGC_NGT_MASK (0x1f) +#define RT5670_DRC_AGC_NGT_SFT 0 + +/* Jack Detect Control (0xbb) */ +#define RT5670_JD_MASK (0x7 << 13) +#define RT5670_JD_SFT 13 +#define RT5670_JD_DIS (0x0 << 13) +#define RT5670_JD_GPIO1 (0x1 << 13) +#define RT5670_JD_JD1_IN4P (0x2 << 13) +#define RT5670_JD_JD2_IN4N (0x3 << 13) +#define RT5670_JD_GPIO2 (0x4 << 13) +#define RT5670_JD_GPIO3 (0x5 << 13) +#define RT5670_JD_GPIO4 (0x6 << 13) +#define RT5670_JD_HP_MASK (0x1 << 11) +#define RT5670_JD_HP_SFT 11 +#define RT5670_JD_HP_DIS (0x0 << 11) +#define RT5670_JD_HP_EN (0x1 << 11) +#define RT5670_JD_HP_TRG_MASK (0x1 << 10) +#define RT5670_JD_HP_TRG_SFT 10 +#define RT5670_JD_HP_TRG_LO (0x0 << 10) +#define RT5670_JD_HP_TRG_HI (0x1 << 10) +#define RT5670_JD_SPL_MASK (0x1 << 9) +#define RT5670_JD_SPL_SFT 9 +#define RT5670_JD_SPL_DIS (0x0 << 9) +#define RT5670_JD_SPL_EN (0x1 << 9) +#define RT5670_JD_SPL_TRG_MASK (0x1 << 8) +#define RT5670_JD_SPL_TRG_SFT 8 +#define RT5670_JD_SPL_TRG_LO (0x0 << 8) +#define RT5670_JD_SPL_TRG_HI (0x1 << 8) +#define RT5670_JD_SPR_MASK (0x1 << 7) +#define RT5670_JD_SPR_SFT 7 +#define RT5670_JD_SPR_DIS (0x0 << 7) +#define RT5670_JD_SPR_EN (0x1 << 7) +#define RT5670_JD_SPR_TRG_MASK (0x1 << 6) +#define RT5670_JD_SPR_TRG_SFT 6 +#define RT5670_JD_SPR_TRG_LO (0x0 << 6) +#define RT5670_JD_SPR_TRG_HI (0x1 << 6) +#define RT5670_JD_MO_MASK (0x1 << 5) +#define RT5670_JD_MO_SFT 5 +#define RT5670_JD_MO_DIS (0x0 << 5) +#define RT5670_JD_MO_EN (0x1 << 5) +#define RT5670_JD_MO_TRG_MASK (0x1 << 4) +#define RT5670_JD_MO_TRG_SFT 4 +#define RT5670_JD_MO_TRG_LO (0x0 << 4) +#define RT5670_JD_MO_TRG_HI (0x1 << 4) +#define RT5670_JD_LO_MASK (0x1 << 3) +#define RT5670_JD_LO_SFT 3 +#define RT5670_JD_LO_DIS (0x0 << 3) +#define RT5670_JD_LO_EN (0x1 << 3) +#define RT5670_JD_LO_TRG_MASK (0x1 << 2) +#define RT5670_JD_LO_TRG_SFT 2 +#define RT5670_JD_LO_TRG_LO (0x0 << 2) +#define RT5670_JD_LO_TRG_HI (0x1 << 2) +#define RT5670_JD1_IN4P_MASK (0x1 << 1) +#define RT5670_JD1_IN4P_SFT 1 +#define RT5670_JD1_IN4P_DIS (0x0 << 1) +#define RT5670_JD1_IN4P_EN (0x1 << 1) +#define RT5670_JD2_IN4N_MASK (0x1) +#define RT5670_JD2_IN4N_SFT 0 +#define RT5670_JD2_IN4N_DIS (0x0) +#define RT5670_JD2_IN4N_EN (0x1) + +/* IRQ Control 1 (0xbd) */ +#define RT5670_IRQ_JD_MASK (0x1 << 15) +#define RT5670_IRQ_JD_SFT 15 +#define RT5670_IRQ_JD_BP (0x0 << 15) +#define RT5670_IRQ_JD_NOR (0x1 << 15) +#define RT5670_IRQ_OT_MASK (0x1 << 14) +#define RT5670_IRQ_OT_SFT 14 +#define RT5670_IRQ_OT_BP (0x0 << 14) +#define RT5670_IRQ_OT_NOR (0x1 << 14) +#define RT5670_JD_STKY_MASK (0x1 << 13) +#define RT5670_JD_STKY_SFT 13 +#define RT5670_JD_STKY_DIS (0x0 << 13) +#define RT5670_JD_STKY_EN (0x1 << 13) +#define RT5670_OT_STKY_MASK (0x1 << 12) +#define RT5670_OT_STKY_SFT 12 +#define RT5670_OT_STKY_DIS (0x0 << 12) +#define RT5670_OT_STKY_EN (0x1 << 12) +#define RT5670_JD_P_MASK (0x1 << 11) +#define RT5670_JD_P_SFT 11 +#define RT5670_JD_P_NOR (0x0 << 11) +#define RT5670_JD_P_INV (0x1 << 11) +#define RT5670_OT_P_MASK (0x1 << 10) +#define RT5670_OT_P_SFT 10 +#define RT5670_OT_P_NOR (0x0 << 10) +#define RT5670_OT_P_INV (0x1 << 10) + +/* IRQ Control 2 (0xbe) */ +#define RT5670_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5670_IRQ_MB1_OC_SFT 15 +#define RT5670_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5670_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5670_IRQ_MB2_OC_MASK (0x1 << 14) +#define RT5670_IRQ_MB2_OC_SFT 14 +#define RT5670_IRQ_MB2_OC_BP (0x0 << 14) +#define RT5670_IRQ_MB2_OC_NOR (0x1 << 14) +#define RT5670_MB1_OC_STKY_MASK (0x1 << 11) +#define RT5670_MB1_OC_STKY_SFT 11 +#define RT5670_MB1_OC_STKY_DIS (0x0 << 11) +#define RT5670_MB1_OC_STKY_EN (0x1 << 11) +#define RT5670_MB2_OC_STKY_MASK (0x1 << 10) +#define RT5670_MB2_OC_STKY_SFT 10 +#define RT5670_MB2_OC_STKY_DIS (0x0 << 10) +#define RT5670_MB2_OC_STKY_EN (0x1 << 10) +#define RT5670_MB1_OC_P_MASK (0x1 << 7) +#define RT5670_MB1_OC_P_SFT 7 +#define RT5670_MB1_OC_P_NOR (0x0 << 7) +#define RT5670_MB1_OC_P_INV (0x1 << 7) +#define RT5670_MB2_OC_P_MASK (0x1 << 6) +#define RT5670_MB2_OC_P_SFT 6 +#define RT5670_MB2_OC_P_NOR (0x0 << 6) +#define RT5670_MB2_OC_P_INV (0x1 << 6) +#define RT5670_MB1_OC_CLR (0x1 << 3) +#define RT5670_MB1_OC_CLR_SFT 3 +#define RT5670_MB2_OC_CLR (0x1 << 2) +#define RT5670_MB2_OC_CLR_SFT 2 + +/* GPIO Control 1 (0xc0) */ +#define RT5670_GP1_PIN_MASK (0x1 << 15) +#define RT5670_GP1_PIN_SFT 15 +#define RT5670_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5670_GP1_PIN_IRQ (0x1 << 15) +#define RT5670_GP2_PIN_MASK (0x1 << 14) +#define RT5670_GP2_PIN_SFT 14 +#define RT5670_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5670_GP2_PIN_DMIC1_SCL (0x1 << 14) +#define RT5670_GP3_PIN_MASK (0x3 << 12) +#define RT5670_GP3_PIN_SFT 12 +#define RT5670_GP3_PIN_GPIO3 (0x0 << 12) +#define RT5670_GP3_PIN_DMIC1_SDA (0x1 << 12) +#define RT5670_GP3_PIN_IRQ (0x2 << 12) +#define RT5670_GP4_PIN_MASK (0x1 << 11) +#define RT5670_GP4_PIN_SFT 11 +#define RT5670_GP4_PIN_GPIO4 (0x0 << 11) +#define RT5670_GP4_PIN_DMIC2_SDA (0x1 << 11) +#define RT5670_DP_SIG_MASK (0x1 << 10) +#define RT5670_DP_SIG_SFT 10 +#define RT5670_DP_SIG_TEST (0x0 << 10) +#define RT5670_DP_SIG_AP (0x1 << 10) +#define RT5670_GPIO_M_MASK (0x1 << 9) +#define RT5670_GPIO_M_SFT 9 +#define RT5670_GPIO_M_FLT (0x0 << 9) +#define RT5670_GPIO_M_PH (0x1 << 9) +#define RT5670_I2S2_PIN_MASK (0x1 << 8) +#define RT5670_I2S2_PIN_SFT 8 +#define RT5670_I2S2_PIN_I2S (0x0 << 8) +#define RT5670_I2S2_PIN_GPIO (0x1 << 8) +#define RT5670_GP5_PIN_MASK (0x1 << 7) +#define RT5670_GP5_PIN_SFT 7 +#define RT5670_GP5_PIN_GPIO5 (0x0 << 7) +#define RT5670_GP5_PIN_DMIC3_SCL (0x1 << 7) +#define RT5670_GP6_PIN_MASK (0x1 << 6) +#define RT5670_GP6_PIN_SFT 6 +#define RT5670_GP6_PIN_GPIO6 (0x0 << 6) +#define RT5670_GP6_PIN_DMIC1_SDA (0x1 << 6) +#define RT5670_GP7_PIN_MASK (0x3 << 4) +#define RT5670_GP7_PIN_SFT 4 +#define RT5670_GP7_PIN_GPIO7 (0x0 << 4) +#define RT5670_GP7_PIN_DMIC1_SDA (0x1 << 4) +#define RT5670_GP7_PIN_PDM_SCL2 (0x2 << 4) +#define RT5670_GP8_PIN_MASK (0x1 << 3) +#define RT5670_GP8_PIN_SFT 3 +#define RT5670_GP8_PIN_GPIO8 (0x0 << 3) +#define RT5670_GP8_PIN_DMIC2_SDA (0x1 << 3) +#define RT5670_GP9_PIN_MASK (0x1 << 2) +#define RT5670_GP9_PIN_SFT 2 +#define RT5670_GP9_PIN_GPIO9 (0x0 << 2) +#define RT5670_GP9_PIN_DMIC3_SDA (0x1 << 2) +#define RT5670_GP10_PIN_MASK (0x3) +#define RT5670_GP10_PIN_SFT 0 +#define RT5670_GP10_PIN_GPIO9 (0x0) +#define RT5670_GP10_PIN_DMIC3_SDA (0x1) +#define RT5670_GP10_PIN_PDM_ADT2 (0x2) + +/* GPIO Control 2 (0xc1) */ +#define RT5670_GP4_PF_MASK (0x1 << 11) +#define RT5670_GP4_PF_SFT 11 +#define RT5670_GP4_PF_IN (0x0 << 11) +#define RT5670_GP4_PF_OUT (0x1 << 11) +#define RT5670_GP4_OUT_MASK (0x1 << 10) +#define RT5670_GP4_OUT_SFT 10 +#define RT5670_GP4_OUT_LO (0x0 << 10) +#define RT5670_GP4_OUT_HI (0x1 << 10) +#define RT5670_GP4_P_MASK (0x1 << 9) +#define RT5670_GP4_P_SFT 9 +#define RT5670_GP4_P_NOR (0x0 << 9) +#define RT5670_GP4_P_INV (0x1 << 9) +#define RT5670_GP3_PF_MASK (0x1 << 8) +#define RT5670_GP3_PF_SFT 8 +#define RT5670_GP3_PF_IN (0x0 << 8) +#define RT5670_GP3_PF_OUT (0x1 << 8) +#define RT5670_GP3_OUT_MASK (0x1 << 7) +#define RT5670_GP3_OUT_SFT 7 +#define RT5670_GP3_OUT_LO (0x0 << 7) +#define RT5670_GP3_OUT_HI (0x1 << 7) +#define RT5670_GP3_P_MASK (0x1 << 6) +#define RT5670_GP3_P_SFT 6 +#define RT5670_GP3_P_NOR (0x0 << 6) +#define RT5670_GP3_P_INV (0x1 << 6) +#define RT5670_GP2_PF_MASK (0x1 << 5) +#define RT5670_GP2_PF_SFT 5 +#define RT5670_GP2_PF_IN (0x0 << 5) +#define RT5670_GP2_PF_OUT (0x1 << 5) +#define RT5670_GP2_OUT_MASK (0x1 << 4) +#define RT5670_GP2_OUT_SFT 4 +#define RT5670_GP2_OUT_LO (0x0 << 4) +#define RT5670_GP2_OUT_HI (0x1 << 4) +#define RT5670_GP2_P_MASK (0x1 << 3) +#define RT5670_GP2_P_SFT 3 +#define RT5670_GP2_P_NOR (0x0 << 3) +#define RT5670_GP2_P_INV (0x1 << 3) +#define RT5670_GP1_PF_MASK (0x1 << 2) +#define RT5670_GP1_PF_SFT 2 +#define RT5670_GP1_PF_IN (0x0 << 2) +#define RT5670_GP1_PF_OUT (0x1 << 2) +#define RT5670_GP1_OUT_MASK (0x1 << 1) +#define RT5670_GP1_OUT_SFT 1 +#define RT5670_GP1_OUT_LO (0x0 << 1) +#define RT5670_GP1_OUT_HI (0x1 << 1) +#define RT5670_GP1_P_MASK (0x1) +#define RT5670_GP1_P_SFT 0 +#define RT5670_GP1_P_NOR (0x0) +#define RT5670_GP1_P_INV (0x1) + +/* Scramble Function (0xcd) */ +#define RT5670_SCB_KEY_MASK (0xff) +#define RT5670_SCB_KEY_SFT 0 + +/* Scramble Control (0xce) */ +#define RT5670_SCB_SWAP_MASK (0x1 << 15) +#define RT5670_SCB_SWAP_SFT 15 +#define RT5670_SCB_SWAP_DIS (0x0 << 15) +#define RT5670_SCB_SWAP_EN (0x1 << 15) +#define RT5670_SCB_MASK (0x1 << 14) +#define RT5670_SCB_SFT 14 +#define RT5670_SCB_DIS (0x0 << 14) +#define RT5670_SCB_EN (0x1 << 14) + +/* Baseback Control (0xcf) */ +#define RT5670_BB_MASK (0x1 << 15) +#define RT5670_BB_SFT 15 +#define RT5670_BB_DIS (0x0 << 15) +#define RT5670_BB_EN (0x1 << 15) +#define RT5670_BB_CT_MASK (0x7 << 12) +#define RT5670_BB_CT_SFT 12 +#define RT5670_BB_CT_A (0x0 << 12) +#define RT5670_BB_CT_B (0x1 << 12) +#define RT5670_BB_CT_C (0x2 << 12) +#define RT5670_BB_CT_D (0x3 << 12) +#define RT5670_M_BB_L_MASK (0x1 << 9) +#define RT5670_M_BB_L_SFT 9 +#define RT5670_M_BB_R_MASK (0x1 << 8) +#define RT5670_M_BB_R_SFT 8 +#define RT5670_M_BB_HPF_L_MASK (0x1 << 7) +#define RT5670_M_BB_HPF_L_SFT 7 +#define RT5670_M_BB_HPF_R_MASK (0x1 << 6) +#define RT5670_M_BB_HPF_R_SFT 6 +#define RT5670_G_BB_BST_MASK (0x3f) +#define RT5670_G_BB_BST_SFT 0 + +/* MP3 Plus Control 1 (0xd0) */ +#define RT5670_M_MP3_L_MASK (0x1 << 15) +#define RT5670_M_MP3_L_SFT 15 +#define RT5670_M_MP3_R_MASK (0x1 << 14) +#define RT5670_M_MP3_R_SFT 14 +#define RT5670_M_MP3_MASK (0x1 << 13) +#define RT5670_M_MP3_SFT 13 +#define RT5670_M_MP3_DIS (0x0 << 13) +#define RT5670_M_MP3_EN (0x1 << 13) +#define RT5670_EG_MP3_MASK (0x1f << 8) +#define RT5670_EG_MP3_SFT 8 +#define RT5670_MP3_HLP_MASK (0x1 << 7) +#define RT5670_MP3_HLP_SFT 7 +#define RT5670_MP3_HLP_DIS (0x0 << 7) +#define RT5670_MP3_HLP_EN (0x1 << 7) +#define RT5670_M_MP3_ORG_L_MASK (0x1 << 6) +#define RT5670_M_MP3_ORG_L_SFT 6 +#define RT5670_M_MP3_ORG_R_MASK (0x1 << 5) +#define RT5670_M_MP3_ORG_R_SFT 5 + +/* MP3 Plus Control 2 (0xd1) */ +#define RT5670_MP3_WT_MASK (0x1 << 13) +#define RT5670_MP3_WT_SFT 13 +#define RT5670_MP3_WT_1_4 (0x0 << 13) +#define RT5670_MP3_WT_1_2 (0x1 << 13) +#define RT5670_OG_MP3_MASK (0x1f << 8) +#define RT5670_OG_MP3_SFT 8 +#define RT5670_HG_MP3_MASK (0x3f) +#define RT5670_HG_MP3_SFT 0 + +/* 3D HP Control 1 (0xd2) */ +#define RT5670_3D_CF_MASK (0x1 << 15) +#define RT5670_3D_CF_SFT 15 +#define RT5670_3D_CF_DIS (0x0 << 15) +#define RT5670_3D_CF_EN (0x1 << 15) +#define RT5670_3D_HP_MASK (0x1 << 14) +#define RT5670_3D_HP_SFT 14 +#define RT5670_3D_HP_DIS (0x0 << 14) +#define RT5670_3D_HP_EN (0x1 << 14) +#define RT5670_3D_BT_MASK (0x1 << 13) +#define RT5670_3D_BT_SFT 13 +#define RT5670_3D_BT_DIS (0x0 << 13) +#define RT5670_3D_BT_EN (0x1 << 13) +#define RT5670_3D_1F_MIX_MASK (0x3 << 11) +#define RT5670_3D_1F_MIX_SFT 11 +#define RT5670_3D_HP_M_MASK (0x1 << 10) +#define RT5670_3D_HP_M_SFT 10 +#define RT5670_3D_HP_M_SUR (0x0 << 10) +#define RT5670_3D_HP_M_FRO (0x1 << 10) +#define RT5670_M_3D_HRTF_MASK (0x1 << 9) +#define RT5670_M_3D_HRTF_SFT 9 +#define RT5670_M_3D_D2H_MASK (0x1 << 8) +#define RT5670_M_3D_D2H_SFT 8 +#define RT5670_M_3D_D2R_MASK (0x1 << 7) +#define RT5670_M_3D_D2R_SFT 7 +#define RT5670_M_3D_REVB_MASK (0x1 << 6) +#define RT5670_M_3D_REVB_SFT 6 + +/* Adjustable high pass filter control 1 (0xd3) */ +#define RT5670_2ND_HPF_MASK (0x1 << 15) +#define RT5670_2ND_HPF_SFT 15 +#define RT5670_2ND_HPF_DIS (0x0 << 15) +#define RT5670_2ND_HPF_EN (0x1 << 15) +#define RT5670_HPF_CF_L_MASK (0x7 << 12) +#define RT5670_HPF_CF_L_SFT 12 +#define RT5670_1ST_HPF_MASK (0x1 << 11) +#define RT5670_1ST_HPF_SFT 11 +#define RT5670_1ST_HPF_DIS (0x0 << 11) +#define RT5670_1ST_HPF_EN (0x1 << 11) +#define RT5670_HPF_CF_R_MASK (0x7 << 8) +#define RT5670_HPF_CF_R_SFT 8 +#define RT5670_ZD_T_MASK (0x3 << 6) +#define RT5670_ZD_T_SFT 6 +#define RT5670_ZD_F_MASK (0x3 << 4) +#define RT5670_ZD_F_SFT 4 +#define RT5670_ZD_F_IM (0x0 << 4) +#define RT5670_ZD_F_ZC_IM (0x1 << 4) +#define RT5670_ZD_F_ZC_IOD (0x2 << 4) +#define RT5670_ZD_F_UN (0x3 << 4) + +/* HP calibration control and Amp detection (0xd6) */ +#define RT5670_SI_DAC_MASK (0x1 << 11) +#define RT5670_SI_DAC_SFT 11 +#define RT5670_SI_DAC_AUTO (0x0 << 11) +#define RT5670_SI_DAC_TEST (0x1 << 11) +#define RT5670_DC_CAL_M_MASK (0x1 << 10) +#define RT5670_DC_CAL_M_SFT 10 +#define RT5670_DC_CAL_M_CAL (0x0 << 10) +#define RT5670_DC_CAL_M_NOR (0x1 << 10) +#define RT5670_DC_CAL_MASK (0x1 << 9) +#define RT5670_DC_CAL_SFT 9 +#define RT5670_DC_CAL_DIS (0x0 << 9) +#define RT5670_DC_CAL_EN (0x1 << 9) +#define RT5670_HPD_RCV_MASK (0x7 << 6) +#define RT5670_HPD_RCV_SFT 6 +#define RT5670_HPD_PS_MASK (0x1 << 5) +#define RT5670_HPD_PS_SFT 5 +#define RT5670_HPD_PS_DIS (0x0 << 5) +#define RT5670_HPD_PS_EN (0x1 << 5) +#define RT5670_CAL_M_MASK (0x1 << 4) +#define RT5670_CAL_M_SFT 4 +#define RT5670_CAL_M_DEP (0x0 << 4) +#define RT5670_CAL_M_CAL (0x1 << 4) +#define RT5670_CAL_MASK (0x1 << 3) +#define RT5670_CAL_SFT 3 +#define RT5670_CAL_DIS (0x0 << 3) +#define RT5670_CAL_EN (0x1 << 3) +#define RT5670_CAL_TEST_MASK (0x1 << 2) +#define RT5670_CAL_TEST_SFT 2 +#define RT5670_CAL_TEST_DIS (0x0 << 2) +#define RT5670_CAL_TEST_EN (0x1 << 2) +#define RT5670_CAL_P_MASK (0x3) +#define RT5670_CAL_P_SFT 0 +#define RT5670_CAL_P_NONE (0x0) +#define RT5670_CAL_P_CAL (0x1) +#define RT5670_CAL_P_DAC_CAL (0x2) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5670_SV_MASK (0x1 << 15) +#define RT5670_SV_SFT 15 +#define RT5670_SV_DIS (0x0 << 15) +#define RT5670_SV_EN (0x1 << 15) +#define RT5670_SPO_SV_MASK (0x1 << 14) +#define RT5670_SPO_SV_SFT 14 +#define RT5670_SPO_SV_DIS (0x0 << 14) +#define RT5670_SPO_SV_EN (0x1 << 14) +#define RT5670_OUT_SV_MASK (0x1 << 13) +#define RT5670_OUT_SV_SFT 13 +#define RT5670_OUT_SV_DIS (0x0 << 13) +#define RT5670_OUT_SV_EN (0x1 << 13) +#define RT5670_HP_SV_MASK (0x1 << 12) +#define RT5670_HP_SV_SFT 12 +#define RT5670_HP_SV_DIS (0x0 << 12) +#define RT5670_HP_SV_EN (0x1 << 12) +#define RT5670_ZCD_DIG_MASK (0x1 << 11) +#define RT5670_ZCD_DIG_SFT 11 +#define RT5670_ZCD_DIG_DIS (0x0 << 11) +#define RT5670_ZCD_DIG_EN (0x1 << 11) +#define RT5670_ZCD_MASK (0x1 << 10) +#define RT5670_ZCD_SFT 10 +#define RT5670_ZCD_PD (0x0 << 10) +#define RT5670_ZCD_PU (0x1 << 10) +#define RT5670_M_ZCD_MASK (0x3f << 4) +#define RT5670_M_ZCD_SFT 4 +#define RT5670_M_ZCD_RM_L (0x1 << 9) +#define RT5670_M_ZCD_RM_R (0x1 << 8) +#define RT5670_M_ZCD_SM_L (0x1 << 7) +#define RT5670_M_ZCD_SM_R (0x1 << 6) +#define RT5670_M_ZCD_OM_L (0x1 << 5) +#define RT5670_M_ZCD_OM_R (0x1 << 4) +#define RT5670_SV_DLY_MASK (0xf) +#define RT5670_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5670_ZCD_HP_MASK (0x1 << 15) +#define RT5670_ZCD_HP_SFT 15 +#define RT5670_ZCD_HP_DIS (0x0 << 15) +#define RT5670_ZCD_HP_EN (0x1 << 15) + + +/* Codec Private Register definition */ +/* 3D Speaker Control (0x63) */ +#define RT5670_3D_SPK_MASK (0x1 << 15) +#define RT5670_3D_SPK_SFT 15 +#define RT5670_3D_SPK_DIS (0x0 << 15) +#define RT5670_3D_SPK_EN (0x1 << 15) +#define RT5670_3D_SPK_M_MASK (0x3 << 13) +#define RT5670_3D_SPK_M_SFT 13 +#define RT5670_3D_SPK_CG_MASK (0x1f << 8) +#define RT5670_3D_SPK_CG_SFT 8 +#define RT5670_3D_SPK_SG_MASK (0x1f) +#define RT5670_3D_SPK_SG_SFT 0 + +/* Wind Noise Detection Control 1 (0x6c) */ +#define RT5670_WND_MASK (0x1 << 15) +#define RT5670_WND_SFT 15 +#define RT5670_WND_DIS (0x0 << 15) +#define RT5670_WND_EN (0x1 << 15) + +/* Wind Noise Detection Control 2 (0x6d) */ +#define RT5670_WND_FC_NW_MASK (0x3f << 10) +#define RT5670_WND_FC_NW_SFT 10 +#define RT5670_WND_FC_WK_MASK (0x3f << 4) +#define RT5670_WND_FC_WK_SFT 4 + +/* Wind Noise Detection Control 3 (0x6e) */ +#define RT5670_HPF_FC_MASK (0x3f << 6) +#define RT5670_HPF_FC_SFT 6 +#define RT5670_WND_FC_ST_MASK (0x3f) +#define RT5670_WND_FC_ST_SFT 0 + +/* Wind Noise Detection Control 4 (0x6f) */ +#define RT5670_WND_TH_LO_MASK (0x3ff) +#define RT5670_WND_TH_LO_SFT 0 + +/* Wind Noise Detection Control 5 (0x70) */ +#define RT5670_WND_TH_HI_MASK (0x3ff) +#define RT5670_WND_TH_HI_SFT 0 + +/* Wind Noise Detection Control 8 (0x73) */ +#define RT5670_WND_WIND_MASK (0x1 << 13) /* Read-Only */ +#define RT5670_WND_WIND_SFT 13 +#define RT5670_WND_STRONG_MASK (0x1 << 12) /* Read-Only */ +#define RT5670_WND_STRONG_SFT 12 +enum { + RT5670_NO_WIND, + RT5670_BREEZE, + RT5670_STORM, +}; + +/* Dipole Speaker Interface (0x75) */ +#define RT5670_DP_ATT_MASK (0x3 << 14) +#define RT5670_DP_ATT_SFT 14 +#define RT5670_DP_SPK_MASK (0x1 << 10) +#define RT5670_DP_SPK_SFT 10 +#define RT5670_DP_SPK_DIS (0x0 << 10) +#define RT5670_DP_SPK_EN (0x1 << 10) + +/* EQ Pre Volume Control (0xb3) */ +#define RT5670_EQ_PRE_VOL_MASK (0xffff) +#define RT5670_EQ_PRE_VOL_SFT 0 + +/* EQ Post Volume Control (0xb4) */ +#define RT5670_EQ_PST_VOL_MASK (0xffff) +#define RT5670_EQ_PST_VOL_SFT 0 + +/* Jack Detect Control 3 (0xf8) */ +#define RT5670_CMP_MIC_IN_DET_MASK (0x7 << 12) +#define RT5670_JD_CBJ_EN (0x1 << 7) +#define RT5670_JD_CBJ_POL (0x1 << 6) +#define RT5670_JD_TRI_CBJ_SEL_MASK (0x7 << 3) +#define RT5670_JD_TRI_CBJ_SEL_SFT (3) +#define RT5670_JD_TRI_HPO_SEL_MASK (0x7) +#define RT5670_JD_TRI_HPO_SEL_SFT (0) +#define RT5670_JD_F_GPIO_JD1 (0x0) +#define RT5670_JD_F_JD1_1 (0x1) +#define RT5670_JD_F_JD1_2 (0x2) +#define RT5670_JD_F_JD2 (0x3) +#define RT5670_JD_F_JD3 (0x4) +#define RT5670_JD_F_GPIO_JD2 (0x5) +#define RT5670_JD_F_MX0B_12 (0x6) + +/* Digital Misc Control (0xfa) */ +#define RT5670_RST_DSP (0x1 << 13) +#define RT5670_IF1_ADC1_IN1_SEL (0x1 << 12) +#define RT5670_IF1_ADC1_IN1_SFT 12 +#define RT5670_IF1_ADC1_IN2_SEL (0x1 << 11) +#define RT5670_IF1_ADC1_IN2_SFT 11 +#define RT5670_IF1_ADC2_IN1_SEL (0x1 << 10) +#define RT5670_IF1_ADC2_IN1_SFT 10 + +/* General Control2 (0xfb) */ +#define RT5670_RXDC_SRC_MASK (0x1 << 7) +#define RT5670_RXDC_SRC_STO (0x0 << 7) +#define RT5670_RXDC_SRC_MONO (0x1 << 7) +#define RT5670_RXDC_SRC_SFT (7) +#define RT5670_RXDP2_SEL_MASK (0x1 << 3) +#define RT5670_RXDP2_SEL_IF2 (0x0 << 3) +#define RT5670_RXDP2_SEL_ADC (0x1 << 3) +#define RT5670_RXDP2_SEL_SFT (3) + +/* System Clock Source */ +enum { + RT5670_SCLK_S_MCLK, + RT5670_SCLK_S_PLL1, + RT5670_SCLK_S_RCCLK, +}; + +/* PLL1 Source */ +enum { + RT5670_PLL1_S_MCLK, + RT5670_PLL1_S_BCLK1, + RT5670_PLL1_S_BCLK2, + RT5670_PLL1_S_BCLK3, + RT5670_PLL1_S_BCLK4, +}; + +enum { + RT5670_AIF1, + RT5670_AIF2, + RT5670_AIF3, + RT5670_AIF4, + RT5670_AIFS, +}; + +struct rt5670_pll_code { + bool m_bp; /* Indicates bypass m code or not. */ + int m_code; + int n_code; + int k_code; +}; + +struct rt5670_priv { + struct snd_soc_codec *codec; + struct rt5670_platform_data pdata; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck[RT5670_AIFS]; + int bclk[RT5670_AIFS]; + int master[RT5670_AIFS]; + + struct rt5670_pll_code pll_code; + int pll_src; + int pll_in; + int pll_out; + + int dsp_sw; /* expected parameter setting */ + int dsp_rate; + int jack_type; +}; + +#endif /* __RT5670_H__ */ -- 1.8.1.1.439.g50a6b54

3 3

[alsa-devel] Fw: Isochronous transfer error on USB3
by Mauro Carvalho Chehab 09 Jan '14

09 Jan '14

Hi Hans/Takashi, I'm getting an weird behavior with em28xx, especially when the device is connected into an audio port. I'm using, on my tests, an em28xx HVR-950 device, using this tree: http://git.linuxtv.org/mchehab/experimental.git/shortlog/refs/heads/em28xx-… Where the alsa driver is at: http://git.linuxtv.org/mchehab/experimental.git/blob/refs/heads/em28xx-v4l2… I'm testing it with xawtv3 (http://git.linuxtv.org/xawtv3.git). The ALSA userspace code there is at: http://git.linuxtv.org/xawtv3.git/blob/HEAD:/common/alsa_stream.c What happens is that, when I require xawtv3 to use any latency lower than 65 ms, the audio doesn't work, as it gets lots of underruns per second. FYI, em28xx works at a 48000 KHz sampling rate, and its PM capture Hw is described as: static struct snd_pcm_hardware snd_em28xx_hw_capture = { .info = SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_MMAP_VALID, .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_KNOT, .rate_min = 48000, .rate_max = 48000, .channels_min = 2, .channels_max = 2, .buffer_bytes_max = 62720 * 8, /* just about the value in usbaudio.c */ .period_bytes_min = 64, /* 12544/2, */ .period_bytes_max = 12544, .periods_min = 2, .periods_max = 98, /* 12544, */ }; On my tests, I experimentally discovered that the minimal latency to avoid ALSA library underruns is: - 65ms when using xHCI; - 25ms when using EHCI. Any latency lower than that causes lots of overruns. Very high latency also causes overruns (but on a lower rate, as the period is bigger). I'm wandering if is there anything that could be done either at Kernel side or at userspace side to automatically get some configuration that works as-is, without requiring the user to play with the latency parameter by hand. The alsa-info data is enclosed. Thank you! Mauro PS.: I'm still trying to understand why the minimal allowed latency is different when using xHCI, but I suspect that it is because it uses a different urb->interval than EHCI. Forwarded message: Date: Wed, 08 Jan 2014 16:03:51 -0200 From: Mauro Carvalho Chehab <m.chehab(a)samsung.com> To: Sarah Sharp <sarah.a.sharp(a)linux.intel.com> Cc: jean-philippe francois <jp.francois(a)cynove.com>, linux-usb(a)vger.kernel.org, LMML <linux-media(a)vger.kernel.org>, Shuah Khan <shuah.kh(a)samsung.com> Subject: Re: Isochronous transfer error on USB3 Em Wed, 08 Jan 2014 15:05:12 -0200 Mauro Carvalho Chehab <m.chehab(a)samsung.com> escreveu: > Em Wed, 08 Jan 2014 14:31:28 -0200 > Mauro Carvalho Chehab <m.chehab(a)samsung.com> escreveu: > > > Em Thu, 02 Jan 2014 14:07:22 -0800 > > Sarah Sharp <sarah.a.sharp(a)linux.intel.com> escreveu: > > > > > On Sun, Dec 29, 2013 at 02:54:40AM -0200, Mauro Carvalho Chehab wrote: > > > > It seems that usb_unlink_urb() is causing troubles with xHCI: the > > > > endpoint stops streaming, but, after that, it doesn't start again, > > > > and lots of debug messages are produced. I emailed you the full log > > > > after start streaming in priv (too big for vger), but basically, > > > > it produces: > > > > > > > > [ 1635.754546] xhci_hcd 0000:00:14.0: Endpoint 0x81 not halted, refusing to reset. > > > > [ 1635.754562] xhci_hcd 0000:00:14.0: Endpoint 0x82 not halted, refusing to reset. > > > > [ 1635.754577] xhci_hcd 0000:00:14.0: Endpoint 0x83 not halted, refusing to reset. > > > > [ 1635.754594] xhci_hcd 0000:00:14.0: Endpoint 0x84 not halted, refusing to reset. > > > > > > I think that's due to the driver (or userspace) attempting to reset the > > > endpoint when it didn't actually receive a stall (-EPIPE) status from an > > > URB. When that happens, the xHCI host controller endpoint "toggle" bits > > > get out of sync with the device toggle bits, and the result is that all > > > transfers will fail to the endpoint from then on until you switch > > > alternate interface settings or unplug/replug the device. > > > > > > Try this patch: > > > > > > http://marc.info/?l=linux-usb&m=138116117104619&w=2 > > > > > > It's still under RFC, and I know it has race conditions, but it will let > > > you quickly test whether this fixes your issue. > > > > Didn't work fine, or at least it didn't solve all the problems. Also, it > > started to cause OOPSes due to the race conditions. > > > > > > > > This has been a long-standing xHCI driver bug. I asked my OPW intern to > > > work on the patch to fix it, but she may be a bit busy with her new job > > > to finish up the RFC. I'll probably have to take over finishing the > > > patch, if this turns out to be your issue. > > > > > > > (Not sure why it is trying to stop all endpoints - as just one endpoint was > > > > requested to restart). > > > > > > Something is calling into usb_clear_halt() with all the endpoints. > > > Userspace, perhaps? > > > > No, userspace is not doing it. The userspace doesn't even know that this > > device is USB (and were written at the time that all media drivers were > > PCI only - so it doesn't have any USB specific call on it). > > > > > You could add WARN() calls to usb_clear_halt() to > > > see what code is resetting the endpoints. In any case, it's not part of > > > the USB core code to change configuration or alt settings, since I don't > > > see any xHCI driver output from the endpoint bandwidth code in this > > > chunk of the dmesg you sent: > > > > The em28xx-audio.c driver may need to call usb_set_interface() while > > the video is still streaming, in order to unmute the audio. That happens > > when the audio device is opened. > > > > With EHCI, this works properly. > > > > > [ 1649.640783] xhci_hcd 0000:00:14.0: Removing canceled TD starting at 0xb41e8580 (dma). > > > [ 1649.640784] xhci_hcd 0000:00:14.0: TRB to noop at offset 0xb41e8580 > > > [ 1649.643159] xhci_hcd 0000:00:14.0: Endpoint 0x81 not halted, refusing to reset. > > > [ 1649.643188] xhci_hcd 0000:00:14.0: Endpoint 0x82 not halted, refusing to reset. > > > [ 1649.643215] xhci_hcd 0000:00:14.0: Endpoint 0x83 not halted, refusing to reset. > > > [ 1649.643239] xhci_hcd 0000:00:14.0: Endpoint 0x84 not halted, refusing to reset. > > > [ 1649.735539] xhci_hcd 0000:00:14.0: ERROR no room on ep ring, try ring expansion > > > > > > Sarah Sharp > > > > Btw, sometimes, I get such logs: > > > > [ 646.192273] xhci_hcd 0000:00:14.0: Miss service interval error, set skip flag > > [ 646.192292] xhci_hcd 0000:00:14.0: Miss service interval error, set skip flag > > [ 646.192311] xhci_hcd 0000:00:14.0: Miss service interval error, set skip flag > > [ 646.192329] xhci_hcd 0000:00:14.0: Miss service interval error, set skip flag > > [ 646.192351] xhci_hcd 0000:00:14.0: Miss service interval error, set skip flag > > [ 646.192376] xhci_hcd 0000:00:14.0: Miss service interval error, set skip flag > > > > After adding some debug at em28xx-audio, triggering alsa trigger start > > events, I'm getting those: > > > > [ 3078.971224] snd_em28xx_capture_trigger: start capture > > [ 3078.971284] xhci_hcd 0000:00:14.0: ERROR no room on ep ring, try ring expansion > > [ 3078.971311] xhci_hcd 0000:00:14.0: ring expansion succeed, now has 4 segments > > [ 3078.971350] xhci_hcd 0000:00:14.0: ERROR no room on ep ring, try ring expansion > > [ 3078.971387] xhci_hcd 0000:00:14.0: ring expansion succeed, now has 8 segments > > [ 3079.034626] em28xx_audio_isocirq, 64 packets (first one with size 12) > > > > Here, some audio data arrives. > > > > [ 3079.034665] snd_em28xx_capture_trigger: stop capture > > > > It seems, however, that this didn't arrive in time, and causes an alsa > > buffer underrun. So, it cancels the existing URBs. > > > > PS.: Even with EHCI, it causes a few ALSA underruns before it gets steady. > > I suspect that this is due to em28xx time to synchronize audio and video > > streams. > > > > [ 3079.034736] xhci_hcd 0000:00:14.0: Cancel URB ffff880207900000, dev 4, ep 0x83, starting at offset 0x1ffb13850 > > [ 3079.034755] xhci_hcd 0000:00:14.0: // Ding dong! > > [ 3079.034783] xhci_hcd 0000:00:14.0: Stopped on Transfer TRB > > [ 3079.034790] snd_em28xx_capture_trigger: start capture > > > > While xHCI is still canceling the URBs, a new trigger happens, and it > > calls usb_submit_urb(). > > > > [ 3079.034819] xhci_hcd 0000:00:14.0: Removing canceled TD starting at 0x1ffb13850 (dma). > > [ 3079.034835] xhci_hcd 0000:00:14.0: TRB to noop at offset 0x1ffb13850 > > ... > > [ 3079.036341] xhci_hcd 0000:00:14.0: Removing canceled TD starting at 0xb624b850 (dma). > > [ 3079.036352] xhci_hcd 0000:00:14.0: TRB to noop at offset 0xb624b850 > > [ 3079.036365] em28xx_audio_isocirq, 64 packets (first one with size 0) > > > > But xHCI only finishes cancelling the first URB here... > > > > [ 3079.036382] xhci_hcd 0000:00:14.0: Cancel URB ffff880207900800, dev 4, ep 0x83, starting at offset 0x1ff937010 > > ... > > [ 3079.043158] xhci_hcd 0000:00:14.0: TRB to noop at offset 0x1ffb13840 > > [ 3079.043170] em28xx_audio_isocirq, 64 packets (first one with size 0) > > > > And only here, it finishes to cancel the entire operation. > > > > [ 3079.043231] xhci_hcd 0000:00:14.0: ERROR no room on ep ring, try ring expansion > > [ 3079.043299] xhci_hcd 0000:00:14.0: ring expansion succeed, now has 16 segments > > [ 3079.428996] em28xx_audio_isocirq, 64 packets (first one with size 64) > > > > Finally, after ~400ms after the new usb_submit_urb(), the first audio packet > > appears... > > > > [ 3079.429069] snd_em28xx_capture_trigger: stop capture > > > > However, this is not fast enough to avoid ALSA buffer underrun. So, > > the driver cancels the existing URBs... > > > > [ 3079.429204] xhci_hcd 0000:00:14.0: Cancel URB ffff880207900000, dev 4, ep 0x83, starting at offset 0xc5a7f4b0 > > [ 3079.429241] snd_em28xx_capture_trigger: start capture > > > > And submits a new set. > > > > Not sure how to fix it. > > Hmm... calling xawtv with a very high-latency alsa buffer works (the > default latency is 30ms, with works fine with an EHCI port): > > $ xawtv --alsa-latency 500 > > Of course, a half-second latency means that audio and video won't be > properly synchronized, but at least audio works. > > I'll turn off the USB logs and do more experiences with the latency, > in order to have an idea on how faster is EHCI to handle the > ISOC requests, when compared with xHCI. Ok, at lest on my quad-core 3rd gen i7core notebook, a latency of 65ms is enough for audio to work on em28xx with xHCI. However, using such latency on EHCI causes underruns. A latency of 90ms seems to work fine on both drivers. I'm starting to wander that maybe xHCI is not using the same urb->interval than EHCI, and if this could explain those issues. Regards, -- Cheers, Mauro -- To unsubscribe from this list: send the line "unsubscribe linux-media" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html -- Cheers, Mauro -- upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.62 !!################################ !!Script ran on: Wed Jan 8 18:29:18 UTC 2014 !!Linux Distribution !!------------------ Fedora release 20 (Heisenbug) Fedora release 20 (Heisenbug) NAME=Fedora ID=fedora PRETTY_NAME="Fedora 20 (Heisenbug)" CPE_NAME="cpe:/o:fedoraproject:fedora:20" HOME_URL="https://fedoraproject.org/" BUG_REPORT_URL="https://bugzilla.redhat.com/" REDHAT_BUGZILLA_PRODUCT="Fedora" REDHAT_BUGZILLA_PRODUCT_VERSION=20 REDHAT_SUPPORT_PRODUCT="Fedora" REDHAT_SUPPORT_PRODUCT_VERSION=20 Fedora release 20 (Heisenbug) Fedora release 20 (Heisenbug) !!DMI Information !!--------------- Manufacturer: SAMSUNG ELECTRONICS CO., LTD. Product Name: 550P5C/550P7C Product Version: P04ABI Firmware Version: P04ABI.013.130220.dg !!Kernel Information !!------------------ Kernel release: 3.13.0-rc1+ Operating System: GNU/Linux Architecture: x86_64 Processor: x86_64 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k3.13.0-rc1+ Library version: 1.0.27.2 Utilities version: 1.0.27.2 !!Loaded ALSA modules !!------------------- em28xx_alsa snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/bin/pulseaudio) Running - Yes !!Soundcards recognised by ALSA !!----------------------------- 0 [Em28xxAudio ]: Em28xx-Audio - Em28xx Audio Empia Em28xx Audio 1 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf7910000 irq 47 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller (rev 04) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:1b.0 0403: 8086:1e20 (rev 04) Subsystem: 144d:c0d1 !!Loaded sound module options !!--------------------------- !!Module: em28xx_alsa debug : 0 !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC269VC Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0269 Subsystem Id: 0x144dc0d1 Revision Id: 0x100202 No Modem Function Group found Default PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0 Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x4b 0x4b] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC269VC Analog", type="Audio", device=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC269VC Analog", type="Audio", device=0 Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 Amp-In vals: [0x27 0x27] Converter: stream=4, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 Amp-In vals: [0x97 0x97] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x22 Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Internal Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Internal Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 5 0x18 0x19 0x1a 0x1b 0x1d Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] [0x00] Connection: 2 0x02 0x0b Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00010014: OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x15 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c 0x0d* Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x17 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Control: name="Bass Speaker Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Speaker Surround Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00] Pincap 0x00000010: OUT Pin Default 0x90170120: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x0f Node 0x18 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x03a11830: [Jack] Mic at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x19 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x90a70940: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Front Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1d [Pin Complex] wcaps 0x400400: Mono Pincap 0x00000020: IN Pin Default 0x4006a21d: [N/A] Line Out at Ext N/A Conn = Digital, Color = UNKNOWN DefAssociation = 0x1, Sequence = 0xd Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1e [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=37 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 7 0x18 0x19 0x1a 0x1b 0x1d 0x0b 0x12 Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 6 0x18 0x19 0x1a 0x1b 0x1d 0x0b Codec: Intel PantherPoint HDMI Address: 3 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862806 Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=3 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="ELD", index=0, device=3 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x02 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x80] Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x58560020: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x03 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x80] Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x58560030: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x04 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 3 Jan 8 16:21 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 9 Jan 8 15:41 /dev/snd/controlC1 crw-rw----+ 1 root audio 116, 8 Jan 8 15:41 /dev/snd/hwC1D0 crw-rw----+ 1 root audio 116, 7 Jan 8 15:41 /dev/snd/hwC1D3 crw-rw----+ 1 root audio 116, 2 Jan 8 16:21 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 6 Jan 8 15:51 /dev/snd/pcmC1D0c crw-rw----+ 1 root audio 116, 5 Jan 8 16:21 /dev/snd/pcmC1D0p crw-rw----+ 1 root audio 116, 4 Jan 8 15:51 /dev/snd/pcmC1D3p crw-rw----+ 1 root audio 116, 1 Jan 8 15:41 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Jan 8 15:41 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x. 2 root root 80 Jan 8 16:21 . drwxr-xr-x. 3 root root 260 Jan 8 16:21 .. lrwxrwxrwx. 1 root root 12 Jan 8 16:21 pci-0000:00:1a.0-usb-0:1.2 -> ../controlC0 lrwxrwxrwx. 1 root root 12 Jan 8 15:41 pci-0000:00:1b.0 -> ../controlC1 !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) # # Place your global alsa-lib configuration here... # !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 1: PCH [HDA Intel PCH], device 0: ALC269VC Analog [ALC269VC Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Em28xxAudio [Em28xx Audio], device 0: Em28xx Audio [Empia 28xx Capture] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 0: ALC269VC Analog [ALC269VC Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Em28xxAudio] Card hw:0 'Em28xxAudio'/'Empia Em28xx Audio' Mixer name : '' Components : '' Controls : 24 Simple ctrls : 12 Simple mixer control 'Master',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] Simple mixer control 'PCM',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] Simple mixer control 'Surround',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [on] Front Right: 31 [100%] Playback [on] Simple mixer control 'LFE',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [on] Front Right: 31 [100%] Playback [on] Simple mixer control 'Line',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] Simple mixer control 'Line In',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] Simple mixer control 'CD',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] Simple mixer control 'Microphone',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] Simple mixer control 'Video',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 23 [74%] Playback [on] Front Right: 31 [100%] Playback [on] Simple mixer control 'Phone',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] Simple mixer control 'Mono',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] Simple mixer control 'AUX',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] !!-------Mixer controls for card 1 [PCH] Card hw:1 'PCH'/'HDA Intel PCH at 0xf7910000 irq 47' Mixer name : 'Intel PantherPoint HDMI' Components : 'HDA:10ec0269,144dc0d1,00100202 HDA:80862806,80860101,00100000' Controls : 31 Simple ctrls : 12 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 75 [86%] [-9.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 0 [0%] [-65.25dB] [off] Front Right: Playback 0 [0%] [-65.25dB] [off] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 87 [100%] [0.00dB] [on] Front Right: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'Bass Speaker',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 254 [100%] [-0.20dB] Front Right: Playback 254 [100%] [-0.20dB] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 63 Front Left: Capture 39 [62%] [12.00dB] [on] Front Right: Capture 39 [62%] [12.00dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Enabled' Simple mixer control 'Internal Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Internal Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] !!Alsactl output !!-------------- --startcollapse-- state.Em28xxAudio { control.1 { iface MIXER name 'Video Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.2 { iface MIXER name 'Video Volume' value.0 23 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.3 { iface MIXER name 'Line In Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.4 { iface MIXER name 'Line In Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.5 { iface MIXER name 'Phone Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.6 { iface MIXER name 'Phone Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.7 { iface MIXER name 'Microphone Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.8 { iface MIXER name 'Microphone Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.9 { iface MIXER name 'CD Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.10 { iface MIXER name 'CD Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.11 { iface MIXER name 'AUX Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.12 { iface MIXER name 'AUX Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.13 { iface MIXER name 'PCM Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.14 { iface MIXER name 'PCM Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.15 { iface MIXER name 'Master Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.16 { iface MIXER name 'Master Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.17 { iface MIXER name 'Line Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.18 { iface MIXER name 'Line Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.19 { iface MIXER name 'Mono Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.20 { iface MIXER name 'Mono Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.21 { iface MIXER name 'LFE Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.22 { iface MIXER name 'LFE Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.23 { iface MIXER name 'Surround Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.24 { iface MIXER name 'Surround Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } } state.PCH { control.1 { iface MIXER name 'Headphone Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 87' dbmin -6525 dbmax 0 dbvalue.0 -6525 dbvalue.1 -6525 } } control.2 { iface MIXER name 'Headphone Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.3 { iface MIXER name 'Speaker Playback Volume' value.0 87 value.1 87 comment { access 'read write' type INTEGER count 2 range '0 - 87' dbmin -6525 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.4 { iface MIXER name 'Speaker Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.5 { iface MIXER name 'Bass Speaker Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.6 { iface MIXER name 'Internal Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.7 { iface MIXER name 'Internal Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.8 { iface MIXER name 'Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.9 { iface MIXER name 'Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.10 { iface MIXER name 'Auto-Mute Mode' value Enabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.11 { iface MIXER name 'Capture Volume' value.0 39 value.1 39 comment { access 'read write' type INTEGER count 2 range '0 - 63' dbmin -1725 dbmax 3000 dbvalue.0 1200 dbvalue.1 1200 } } control.12 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.13 { iface MIXER name 'Internal Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.14 { iface MIXER name 'Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.15 { iface MIXER name 'Master Playback Volume' value 75 comment { access 'read write' type INTEGER count 1 range '0 - 87' dbmin -6525 dbmax 0 dbvalue.0 -900 } } control.16 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.17 { iface CARD name 'Internal Mic Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.18 { iface CARD name 'Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.19 { iface CARD name 'Headphone Jack' value false comment { access read type BOOLEAN count 1 } } control.20 { iface CARD name 'Speaker Front Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.21 { iface CARD name 'Speaker Surround Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.22 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 comment { access read type INTEGER count 4 range '0 - 36' } } control.23 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.24 { iface CARD name 'HDMI/DP,pcm=3 Jack' value false comment { access read type BOOLEAN count 1 } } control.25 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.26 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.27 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.28 { iface MIXER name 'IEC958 Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.29 { iface PCM device 3 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.30 { iface PCM device 3 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.31 { iface MIXER name 'PCM Playback Volume' value.0 254 value.1 254 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 -20 dbvalue.1 -20 } } } --endcollapse-- !!All Loaded Modules !!------------------ Module fuse ccm nf_conntrack_netbios_ns nf_conntrack_broadcast ipt_MASQUERADE ip6t_REJECT xt_conntrack ebtable_nat ebtable_broute bridge stp llc ebtable_filter ebtables ip6table_nat nf_conntrack_ipv6 nf_defrag_ipv6 nf_nat_ipv6 ip6table_mangle ip6table_security ip6table_raw ip6table_filter ip6_tables iptable_nat nf_conntrack_ipv4 nf_defrag_ipv4 nf_nat_ipv4 nf_nat nf_conntrack iptable_mangle iptable_security iptable_raw bnep vfat fat arc4 iwldvm mac80211 iwlwifi cfg80211 x86_pkg_temp_thermal coretemp kvm_intel kvm crc32_pclmul crc32c_intel ghash_clmulni_intel netconsole rc_hauppauge em28xx_rc rc_core lgdt330x snd_hda_codec_hdmi snd_hda_codec_realtek em28xx_dvb dvb_core snd_hda_intel snd_hda_codec em28xx_alsa snd_hwdep snd_seq snd_seq_device snd_pcm snd_page_alloc snd_timer snd nfsd auth_rpcgss tuner_xc2028 tuner tvp5150 em28xx_v4l uvcvideo em28xx videobuf2_vmalloc tveeprom videobuf2_memops v4l2_common videobuf2_core videodev media btusb bluetooth joydev serio_raw iTCO_wdt iTCO_vendor_support rfkill soundcore r8169 mii shpchp mei_me mei microcode i2c_i801 lpc_ich mfd_core nfs_acl lockd sunrpc nouveau i915 ttm i2c_algo_bit drm_kms_helper drm i2c_core mxm_wmi wmi video !!Sysfs Files !!----------- /sys/class/sound/hwC1D0/init_pin_configs: 0x12 0x411111f0 0x14 0x411111f0 0x15 0x0321101f 0x17 0x90170120 0x18 0x03a11830 0x19 0x90a70940 0x1a 0x411111f0 0x1b 0x90170110 0x1d 0x4006a21d 0x1e 0x411111f0 /sys/class/sound/hwC1D0/driver_pin_configs: /sys/class/sound/hwC1D0/user_pin_configs: /sys/class/sound/hwC1D0/init_verbs: /sys/class/sound/hwC1D0/hints: /sys/class/sound/hwC1D3/init_pin_configs: 0x05 0x18560010 0x06 0x58560020 0x07 0x58560030 /sys/class/sound/hwC1D3/driver_pin_configs: /sys/class/sound/hwC1D3/user_pin_configs: /sys/class/sound/hwC1D3/init_verbs: /sys/class/sound/hwC1D3/hints: !!ALSA/HDA dmesg !!-------------- [ 24.204127] em28xx: Registered (Em28xx Audio Extension) extension [ 24.207614] snd_hda_intel 0000:00:1b.0: enabling device (0000 -> 0002) [ 24.209216] snd_hda_intel 0000:00:1b.0: irq 47 for MSI/MSI-X [ 24.213022] em2882/3 #0: Binding DVB extension [ 24.222898] ALSA sound/pci/hda/patch_realtek.c:419 SKU: Nid=0x1d sku_cfg=0x4006a21d [ 24.222901] ALSA sound/pci/hda/patch_realtek.c:421 SKU: port_connectivity=0x1 [ 24.222902] ALSA sound/pci/hda/patch_realtek.c:422 SKU: enable_pcbeep=0x0 [ 24.222903] ALSA sound/pci/hda/patch_realtek.c:423 SKU: check_sum=0x00000006 [ 24.222904] ALSA sound/pci/hda/patch_realtek.c:424 SKU: customization=0x000000a2 [ 24.222905] ALSA sound/pci/hda/patch_realtek.c:425 SKU: external_amp=0x3 [ 24.222906] ALSA sound/pci/hda/patch_realtek.c:426 SKU: platform_type=0x1 [ 24.222908] ALSA sound/pci/hda/patch_realtek.c:427 SKU: swap=0x0 [ 24.222909] ALSA sound/pci/hda/patch_realtek.c:428 SKU: override=0x1 [ 24.223221] ALSA sound/pci/hda/hda_auto_parser.c:393 autoconfig: line_outs=2 (0x1b/0x17/0x0/0x0/0x0) type:speaker [ 24.223223] ALSA sound/pci/hda/hda_auto_parser.c:397 speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 24.223224] ALSA sound/pci/hda/hda_auto_parser.c:401 hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 24.223225] ALSA sound/pci/hda/hda_auto_parser.c:402 mono: mono_out=0x0 [ 24.223226] ALSA sound/pci/hda/hda_auto_parser.c:406 inputs: [ 24.223228] ALSA sound/pci/hda/hda_auto_parser.c:410 Internal Mic=0x19 [ 24.223230] ALSA sound/pci/hda/hda_auto_parser.c:410 Mic=0x18 [ 24.223232] ALSA sound/pci/hda/patch_realtek.c:491 realtek: No valid SSID, checking pincfg 0x4006a21d for NID 0x1d [ 24.223234] ALSA sound/pci/hda/patch_realtek.c:507 realtek: Enabling init ASM_ID=0xa21d CODEC_ID=10ec0269 [ 24.236439] input: HDA Intel PCH HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1b.0/sound/card1/input11 [ 24.237093] input: HDA Intel PCH Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card1/input10 [ 24.237688] input: HDA Intel PCH Mic as /devices/pci0000:00/0000:00:1b.0/sound/card1/input9 [ 24.257693] xc2028: Xcv2028/3028 init called! -- [ 873.141600] URB ffff88022219f800 submitted while active [ 873.141619] Modules linked in: fuse ccm nf_conntrack_netbios_ns nf_conntrack_broadcast ipt_MASQUERADE ip6t_REJECT xt_conntrack ebtable_nat ebtable_broute bridge stp llc ebtable_filter ebtables ip6table_nat nf_conntrack_ipv6 nf_defrag_ipv6 nf_nat_ipv6 ip6table_mangle ip6table_security ip6table_raw ip6table_filter ip6_tables iptable_nat nf_conntrack_ipv4 nf_defrag_ipv4 nf_nat_ipv4 nf_nat nf_conntrack iptable_mangle iptable_security iptable_raw bnep vfat fat arc4 iwldvm mac80211 iwlwifi cfg80211 x86_pkg_temp_thermal coretemp kvm_intel kvm crc32_pclmul crc32c_intel ghash_clmulni_intel netconsole rc_hauppauge em28xx_rc rc_core lgdt330x snd_hda_codec_hdmi snd_hda_codec_realtek em28xx_dvb dvb_core snd_hda_intel snd_hda_codec em28xx_alsa snd_hwdep snd_seq snd_seq_device snd_pcm snd_page_alloc snd_timer snd [ 873.143618] nfsd auth_rpcgss tuner_xc2028 tuner tvp5150 em28xx_v4l uvcvideo em28xx videobuf2_vmalloc tveeprom videobuf2_memops v4l2_common videobuf2_core videodev media btusb bluetooth joydev serio_raw iTCO_wdt iTCO_vendor_support rfkill soundcore r8169 mii shpchp mei_me mei microcode i2c_i801 lpc_ich mfd_core nfs_acl lockd sunrpc nouveau i915 ttm i2c_algo_bit drm_kms_helper drm i2c_core mxm_wmi wmi video [ 873.145388] CPU: 4 PID: 2103 Comm: kworker/4:0 Not tainted 3.13.0-rc1+ #22

3 5

[alsa-devel] [PATCH] ASoC: fsl_ssi: Set default slot number for common cases
by Nicolin Chen 09 Jan '14

09 Jan '14

For those platforms using DAI master mode like I2S, it's better to pre-set a default slot number so that there's no need for these common cases to set the slot number from its machine driver any more. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 94dedcb..57ab45b 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -711,6 +711,16 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) if (ssi_private->imx_ac97) fsl_ssi_setup_ac97(ssi_private); + /* Set a default slot number so that there is no need for those common + * cases like I2S mode to call the extra set_tdm_slot() any more. + */ + if (!ssi_private->imx_ac97) { + write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(2)); + write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(2)); + } + return 0; } -- 1.8.4

2 3

[alsa-devel] [PATCH 1/2] ALSA: Remove superfluous header inclusions in memalloc.c
by Takashi Iwai 09 Jan '14

09 Jan '14

After cutting off the proc and page reservation codes, we don't need many headers any longer. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index be1544ac4613..e568c14ca88d 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -22,17 +22,13 @@ */ #include <linux/module.h> -#include <linux/proc_fs.h> #include <linux/init.h> -#include <linux/pci.h> #include <linux/slab.h> #include <linux/mm.h> -#include <linux/seq_file.h> #include <asm/uaccess.h> #include <linux/dma-mapping.h> #include <linux/genalloc.h> #include <linux/moduleparam.h> -#include <linux/mutex.h> #include <sound/memalloc.h> -- 1.8.5.2

1 1

[alsa-devel] [PATCH] Support for Yamaha MOXF
by Michael Slade 09 Jan '14

09 Jan '14

My apologies if I'm doing this wrong. I have managed to get a Yamaha MOXF8 working with the attached patch. Presumably also works with a MOXF6. The new Yamaha MOXF6/8 are successors to the MOX6/8. Apart from things like new voices, a flash expansion port, and a different colour at the ends, they are pretty much identical in operation. The MOXF has a USB ID of 0499:150f. So I needed to add a new quirks-table.h stanza for it, by copy-pasting the MOX one and changing he ID. I haven't tested it to death or anything, but I am having no trouble recording and playing audio and midi. Let me know if this patch is acceptable or of there is more that is needed from me. Note, this patch is against Ubuntu Raring's 3.8.0 kernel, but it's a trivial patch. Mick.

3 3

[alsa-devel] [PATCH 1/4] ASoC: simple-card: add simple audio card widgets support
by Xiubo Li 09 Jan '14

09 Jan '14

For many audio cards there need to add some off-CODEC widgets, and this is hard to be supported by dts only, because maybe some widgets need one or more call backs, such as wevent. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- .../devicetree/bindings/sound/simple-card.txt | 1 + include/sound/simple_card.h | 14 ++++ sound/soc/generic/Makefile | 3 +- sound/soc/generic/simple-card-widgets.c | 79 ++++++++++++++++++++++ 4 files changed, 96 insertions(+), 1 deletion(-) create mode 100644 sound/soc/generic/simple-card-widgets.c diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index e9e20ec..e626302 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -15,6 +15,7 @@ Optional properties: Each entry is a pair of strings, the first being the connection's sink, the second being the connection's source. +- simple-sudio-card,widgets : The name of the audio card off-CODEC widgets. Required subnodes: diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 6c74527..75574f8 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -12,8 +12,16 @@ #ifndef __SIMPLE_CARD_H #define __SIMPLE_CARD_H +#include <linux/of.h> #include <sound/soc.h> +struct asoc_simple_card_widgets { + const char *name; + struct list_head list; + const struct snd_soc_dapm_widget *widgets; + unsigned int widgets_cnt; +}; + struct asoc_simple_dai { const char *name; unsigned int fmt; @@ -35,4 +43,10 @@ struct asoc_simple_card_info { struct snd_soc_card snd_card; }; +struct asoc_simple_card_widgets * +asoc_simple_card_get_widgets(struct device_node *); +int asoc_simple_card_widgets_register(struct asoc_simple_card_widgets *); +void +asoc_simple_card_widgets_unregister(struct asoc_simple_card_widgets *); + #endif /* __SIMPLE_CARD_H */ diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 9c3b246..dbacd12 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -1,3 +1,4 @@ -snd-soc-simple-card-objs := simple-card.o +snd-soc-simple-card-objs := simple-card-widgets.o +snd-soc-simple-card-objs += simple-card.o obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o diff --git a/sound/soc/generic/simple-card-widgets.c b/sound/soc/generic/simple-card-widgets.c new file mode 100644 index 0000000..6eb0c8c --- /dev/null +++ b/sound/soc/generic/simple-card-widgets.c @@ -0,0 +1,79 @@ +/* + * ASoC simple sound card widgets support + * + * Copyright 2013 Freescale Semiconductor, Inc. + * Xiubo Li <Li.Xiubo(a)freescale.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <sound/simple_card.h> + +static DEFINE_MUTEX(simple_widgets_mutex); +static LIST_HEAD(simple_widgets_list); + +void +asoc_simple_card_widgets_unregister(struct asoc_simple_card_widgets *comp) +{ + mutex_lock(&simple_widgets_mutex); + list_del(&comp->list); + mutex_unlock(&simple_widgets_mutex); +} +EXPORT_SYMBOL_GPL(asoc_simple_card_widgets_unregister); + +int +asoc_simple_card_widgets_register(struct asoc_simple_card_widgets *comp) +{ + if (!comp) { + pr_err("Simple Card: Failed to register widgets\n"); + return -EINVAL; + } + + mutex_lock(&simple_widgets_mutex); + list_add(&comp->list, &simple_widgets_list); + mutex_unlock(&simple_widgets_mutex); + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_card_widgets_register); + +struct asoc_simple_card_widgets * +asoc_simple_card_get_widgets(struct device_node *np) +{ + struct asoc_simple_card_widgets *comp; + const char *string; + int ret; + + if (!np) + return ERR_PTR(-EINVAL); + + if (!of_property_read_bool(np, "simple-audio-card,widgets")) + return NULL; + + ret = of_property_read_string(np, "simple-audio-card,widgets", + &string); + if (ret < 0) + return ERR_PTR(ret); + + mutex_lock(&simple_widgets_mutex); + list_for_each_entry(comp, &simple_widgets_list, list) { + if (!strcmp(string, comp->name)) { + mutex_unlock(&simple_widgets_mutex); + return comp; + } + + } + mutex_unlock(&simple_widgets_mutex); + + return ERR_PTR(-EPROBE_DEFER); +} +EXPORT_SYMBOL_GPL(asoc_simple_card_get_widgets); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:asoc-simple-card-widgets"); +MODULE_DESCRIPTION("ASoC Simple Sound Card Widgets"); +MODULE_AUTHOR("Xiubo Li <Li.Xiubo(a)freescale.com>"); -- 1.8.4

3 7

[alsa-devel] [PATCH 0/3] ALSA: hda - enable display audio for Intel Broadwell
by mengdong.lin＠intel.com 09 Jan '14

09 Jan '14

From: Mengdong Lin <mengdong.lin(a)intel.com> The following patches enable HDMI/DP audio for Intel Broadwell platform from audio driver side. Gfx driver just reuses Haswell audio enabling code. Mengdong Lin (3): ALSA: hda - add device ID for Broadwell display audio controller ALSA: hda - add codec ID for Broadwell display audio codec ALSA: hda/hdmi - apply all Haswell fix-ups to Broadwell display codec sound/pci/hda/hda_intel.c | 3 +++ sound/pci/hda/patch_hdmi.c | 17 +++++++++++------ 2 files changed, 14 insertions(+), 6 deletions(-) -- 1.8.1.2

2 4

[alsa-devel] [PATCH] ASoC: da9055: Partially fix device registration
by Mark Brown 08 Jan '14

08 Jan '14

From: Mark Brown <broonie(a)linaro.org> It has never been possible to even load the da9055 driver since both the MFD and CODEC drivers register the I2C device "da9055". Fix this by using the usual pattern for MFDs with multiple I2C addresses and having the CODEC driver reference an I2C dummy registered by the MFD. Since I don't know which I2C address to use for the CODEC a FIXME has been left in the MFD, this doesn't make anything any worse since the device has never been able to load in the first place. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- drivers/mfd/da9055-core.c | 3 +++ drivers/mfd/da9055-i2c.c | 2 ++ include/linux/mfd/da9055/core.h | 2 +- sound/soc/codecs/da9055.c | 49 ++++++++++++++++++++--------------------- 4 files changed, 30 insertions(+), 26 deletions(-) diff --git a/drivers/mfd/da9055-core.c b/drivers/mfd/da9055-core.c index caf8dcffd0ad..fb0284cda4a5 100644 --- a/drivers/mfd/da9055-core.c +++ b/drivers/mfd/da9055-core.c @@ -365,6 +365,9 @@ static const struct mfd_cell da9055_devs[] = { .of_compatible = "dialog,da9055-watchdog", .name = "da9055-watchdog", }, + { + .name = "da9055-codec", + }, }; static struct regmap_irq_chip da9055_regmap_irq_chip = { diff --git a/drivers/mfd/da9055-i2c.c b/drivers/mfd/da9055-i2c.c index 13af7e50021e..4f5e18750e1a 100644 --- a/drivers/mfd/da9055-i2c.c +++ b/drivers/mfd/da9055-i2c.c @@ -39,6 +39,8 @@ static int da9055_i2c_probe(struct i2c_client *i2c, da9055->dev = &i2c->dev; da9055->chip_irq = i2c->irq; + /* FIXME: Should use i2c_add_dummy() for the CODEC */ + i2c_set_clientdata(i2c, da9055); return da9055_device_init(da9055); diff --git a/include/linux/mfd/da9055/core.h b/include/linux/mfd/da9055/core.h index 956afa445998..e293797ded66 100644 --- a/include/linux/mfd/da9055/core.h +++ b/include/linux/mfd/da9055/core.h @@ -42,7 +42,7 @@ struct da9055 { struct regmap *regmap; struct regmap_irq_chip_data *irq_data; struct device *dev; - struct i2c_client *i2c_client; + struct i2c_client *i2c_client, *codec; int irq_base; int chip_irq; diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 52b79a487ac7..88c1a884aed8 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -14,10 +14,10 @@ */ #include <linux/delay.h> -#include <linux/i2c.h> #include <linux/regmap.h> #include <linux/slab.h> #include <linux/module.h> +#include <linux/mfd/da9055/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -1474,7 +1474,7 @@ static struct snd_soc_codec_driver soc_codec_dev_da9055 = { .num_dapm_routes = ARRAY_SIZE(da9055_audio_map), }; -static const struct regmap_config da9055_regmap_config = { +static const struct regmap_config da9055_codec_regmap_config = { .reg_bits = 8, .val_bits = 8, @@ -1484,14 +1484,19 @@ static const struct regmap_config da9055_regmap_config = { .cache_type = REGCACHE_RBTREE, }; -static int da9055_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +static int da9055_dev_probe(struct platform_device *pdev) { + struct da9055 *core = dev_get_drvdata(pdev->dev.parent); struct da9055_priv *da9055; - struct da9055_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct da9055_platform_data *pdata = dev_get_platdata(&pdev->dev); int ret; - da9055 = devm_kzalloc(&i2c->dev, sizeof(struct da9055_priv), + if (!core->codec) { + dev_err(&pdev->dev, "No CODEC device registered\n"); + return -ENODEV; + } + + da9055 = devm_kzalloc(&pdev->dev, sizeof(struct da9055_priv), GFP_KERNEL); if (!da9055) return -ENOMEM; @@ -1499,48 +1504,42 @@ static int da9055_i2c_probe(struct i2c_client *i2c, if (pdata) da9055->pdata = pdata; - i2c_set_clientdata(i2c, da9055); + platform_set_drvdata(pdev, da9055); - da9055->regmap = devm_regmap_init_i2c(i2c, &da9055_regmap_config); + da9055->regmap = devm_regmap_init_i2c(core->codec, + &da9055_codec_regmap_config); if (IS_ERR(da9055->regmap)) { ret = PTR_ERR(da9055->regmap); - dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + dev_err(&pdev->dev, "regmap_init() failed: %d\n", ret); return ret; } - ret = snd_soc_register_codec(&i2c->dev, + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_da9055, &da9055_dai, 1); if (ret < 0) { - dev_err(&i2c->dev, "Failed to register da9055 codec: %d\n", + dev_err(&pdev->dev, "Failed to register da9055 codec: %d\n", ret); } return ret; } -static int da9055_remove(struct i2c_client *client) +static int da9055_dev_remove(struct platform_device *pdev) { - snd_soc_unregister_codec(&client->dev); + snd_soc_unregister_codec(&pdev->dev); return 0; } -static const struct i2c_device_id da9055_i2c_id[] = { - { "da9055", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); - /* I2C codec control layer */ -static struct i2c_driver da9055_i2c_driver = { +static struct platform_driver da9055_codec_driver = { .driver = { - .name = "da9055", + .name = "da9055-codec", .owner = THIS_MODULE, }, - .probe = da9055_i2c_probe, - .remove = da9055_remove, - .id_table = da9055_i2c_id, + .probe = da9055_dev_probe, + .remove = da9055_dev_remove, }; -module_i2c_driver(da9055_i2c_driver); +module_platform_driver(da9055_codec_driver); MODULE_DESCRIPTION("ASoC DA9055 Codec driver"); MODULE_AUTHOR("David Chen, Ashish Chavan"); -- 1.8.5.2

2 1

Re: [alsa-devel] snd-usb: "delay: estimated 0, actual 352"
by Takashi Iwai 08 Jan '14

08 Jan '14

At Thu, 6 Sep 2012 15:17:58 +0200, Markus Trippelsdorf wrote: > > On 2012.09.06 at 15:08 +0200, Takashi Iwai wrote: > > At Thu, 6 Sep 2012 11:43:48 +0200, > > Markus Trippelsdorf wrote: > > > > > > On 2012.09.06 at 10:21 +0200, Takashi Iwai wrote: > > > > At Thu, 06 Sep 2012 09:35:26 +0200, > > > > Takashi Iwai wrote: > > > > > > > > In short, a patch like below may fix the issue (note: completely > > > > untested!) > > > > > > No it doesn't, unfortunately... > > > > OK, I start tracking down the problem a bit more deeply now. > > > > The issue happens when the first two URBs are passed to > > retire_playback_urb(). These are URBs filled before start_endpoints() > > are set, so they contain actually zero size. Even though these are > > a sort of dummy packets, the driver still tries to check with the > > queued delay account, and gives bogus errors. > > > > So, essentially the messages are harmless and nothing to worry too > > much, but surely it doesn't look sexy. > > > > The patch below should fix the problem. Please give it a try. > > Yes, your patch finally fixes the problem. > Thank you Takashi-san. Thanks for your quick test! If Daniel has no objection with that patch, I'm going to merge it. Takashi

3 2

[alsa-devel] [PATCH 1/4] ASoC: max98088: Use params_width() rather than memory format
by Mark Brown 08 Jan '14

08 Jan '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max98088.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 53d7dab4e054..ee660e2d3df3 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1233,12 +1233,12 @@ static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: snd_soc_update_bits(codec, M98088_REG_14_DAI1_FORMAT, M98088_DAI_WS, 0); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: snd_soc_update_bits(codec, M98088_REG_14_DAI1_FORMAT, M98088_DAI_WS, M98088_DAI_WS); break; -- 1.8.5.2

1 3

[alsa-devel] [PATCH 1/2 v2] ASoC: wm_adsp: Factor out ADSP2 boot proceedure
by Charles Keepax 08 Jan '14

08 Jan '14

Move the ADSP2 boot proceedure into a work structure in preparation for running it asynchronously with the reset of the audio path bring up. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 181 +++++++++++++++++++++++++------------------- sound/soc/codecs/wm_adsp.h | 2 + 2 files changed, 105 insertions(+), 78 deletions(-) Changes since v1: - Schedule to the system unbound work queue since it is likely to be a fairly long task. - Move final start signal to the DSP to be an out_drv widget such that the download is overlapped with the output enables. Thanks, Charles diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index fb0c678..41bbac9 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1497,107 +1497,131 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) return 0; } -int wm_adsp2_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +void wm_adsp2_boot_work(struct work_struct *work) { - struct snd_soc_codec *codec = w->codec; - struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); - struct wm_adsp *dsp = &dsps[w->shift]; - struct wm_adsp_alg_region *alg_region; - struct wm_coeff_ctl *ctl; - unsigned int val; + struct wm_adsp *dsp = container_of(work, + struct wm_adsp, + boot_work); int ret; + unsigned int val; - dsp->card = codec->card; + /* + * For simplicity set the DSP clock rate to be the + * SYSCLK rate rather than making it configurable. + */ + ret = regmap_read(dsp->regmap, ARIZONA_SYSTEM_CLOCK_1, &val); + if (ret != 0) { + adsp_err(dsp, "Failed to read SYSCLK state: %d\n", ret); + return; + } + val = (val & ARIZONA_SYSCLK_FREQ_MASK) + >> ARIZONA_SYSCLK_FREQ_SHIFT; - switch (event) { - case SND_SOC_DAPM_POST_PMU: - /* - * For simplicity set the DSP clock rate to be the - * SYSCLK rate rather than making it configurable. - */ - ret = regmap_read(dsp->regmap, ARIZONA_SYSTEM_CLOCK_1, &val); - if (ret != 0) { - adsp_err(dsp, "Failed to read SYSCLK state: %d\n", - ret); - return ret; - } - val = (val & ARIZONA_SYSCLK_FREQ_MASK) - >> ARIZONA_SYSCLK_FREQ_SHIFT; + ret = regmap_update_bits_async(dsp->regmap, + dsp->base + ADSP2_CLOCKING, + ADSP2_CLK_SEL_MASK, val); + if (ret != 0) { + adsp_err(dsp, "Failed to set clock rate: %d\n", ret); + return; + } - ret = regmap_update_bits_async(dsp->regmap, - dsp->base + ADSP2_CLOCKING, - ADSP2_CLK_SEL_MASK, val); + if (dsp->dvfs) { + ret = regmap_read(dsp->regmap, + dsp->base + ADSP2_CLOCKING, &val); if (ret != 0) { - adsp_err(dsp, "Failed to set clock rate: %d\n", - ret); - return ret; + dev_err(dsp->dev, "Failed to read clocking: %d\n", ret); + return; } - if (dsp->dvfs) { - ret = regmap_read(dsp->regmap, - dsp->base + ADSP2_CLOCKING, &val); + if ((val & ADSP2_CLK_SEL_MASK) >= 3) { + ret = regulator_enable(dsp->dvfs); if (ret != 0) { dev_err(dsp->dev, - "Failed to read clocking: %d\n", ret); - return ret; + "Failed to enable supply: %d\n", + ret); + return; } - if ((val & ADSP2_CLK_SEL_MASK) >= 3) { - ret = regulator_enable(dsp->dvfs); - if (ret != 0) { - dev_err(dsp->dev, - "Failed to enable supply: %d\n", - ret); - return ret; - } - - ret = regulator_set_voltage(dsp->dvfs, - 1800000, - 1800000); - if (ret != 0) { - dev_err(dsp->dev, - "Failed to raise supply: %d\n", - ret); - return ret; - } + ret = regulator_set_voltage(dsp->dvfs, + 1800000, + 1800000); + if (ret != 0) { + dev_err(dsp->dev, + "Failed to raise supply: %d\n", + ret); + return; } } + } - ret = wm_adsp2_ena(dsp); - if (ret != 0) - return ret; + ret = wm_adsp2_ena(dsp); + if (ret != 0) + return; - ret = wm_adsp_load(dsp); - if (ret != 0) - goto err; + ret = wm_adsp_load(dsp); + if (ret != 0) + goto err; - ret = wm_adsp_setup_algs(dsp); - if (ret != 0) - goto err; + ret = wm_adsp_setup_algs(dsp); + if (ret != 0) + goto err; - ret = wm_adsp_load_coeff(dsp); - if (ret != 0) - goto err; + ret = wm_adsp_load_coeff(dsp); + if (ret != 0) + goto err; - /* Initialize caches for enabled and unset controls */ - ret = wm_coeff_init_control_caches(dsp); - if (ret != 0) - goto err; + /* Initialize caches for enabled and unset controls */ + ret = wm_coeff_init_control_caches(dsp); + if (ret != 0) + goto err; - /* Sync set controls */ - ret = wm_coeff_sync_controls(dsp); - if (ret != 0) - goto err; + /* Sync set controls */ + ret = wm_coeff_sync_controls(dsp); + if (ret != 0) + goto err; + + ret = regmap_update_bits_async(dsp->regmap, + dsp->base + ADSP2_CONTROL, + ADSP2_CORE_ENA, + ADSP2_CORE_ENA); + if (ret != 0) + goto err; + + dsp->running = true; + + return; - ret = regmap_update_bits_async(dsp->regmap, - dsp->base + ADSP2_CONTROL, - ADSP2_CORE_ENA | ADSP2_START, - ADSP2_CORE_ENA | ADSP2_START); +err: + regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_SYS_ENA | ADSP2_CORE_ENA | ADSP2_START, 0); +} + +int wm_adsp2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); + struct wm_adsp *dsp = &dsps[w->shift]; + struct wm_adsp_alg_region *alg_region; + struct wm_coeff_ctl *ctl; + int ret; + + dsp->card = codec->card; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + queue_work(system_unbound_wq, &dsp->boot_work); + flush_work(&dsp->boot_work); + + if (!dsp->running) + return -EIO; + + ret = regmap_update_bits(dsp->regmap, + dsp->base + ADSP2_CONTROL, + ADSP2_START, + ADSP2_START); if (ret != 0) goto err; - - dsp->running = true; break; case SND_SOC_DAPM_PRE_PMD: @@ -1668,6 +1692,7 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) INIT_LIST_HEAD(&adsp->alg_regions); INIT_LIST_HEAD(&adsp->ctl_list); + INIT_WORK(&adsp->boot_work, wm_adsp2_boot_work); if (dvfs) { adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index d018dea..b172c1d 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -59,6 +59,8 @@ struct wm_adsp { struct regulator *dvfs; struct list_head ctl_list; + + struct work_struct boot_work; }; #define WM_ADSP1(wname, num) \ -- 1.7.2.5

2 2

[alsa-devel] mx6: Audio does not work on linux-next 20140107
by Fabio Estevam 08 Jan '14

08 Jan '14

Hi, Playing audio on a mx6qsabresd board running linux-next 20140107 leads to the following crash: $ aplay clarinet.wav random: nonblocking pool is initialized Playing WAVE 'clarinet.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono underrun!!! (at least -477981.433 ms long) underrun!!! (at least -477981.435 ms long) underrun!!! (at least -477981.436 ms long) underrun!!! (at least -477981.435 ms long) underrun!!! (at least -477981.436 ms long) underrun!!! (at least -477981.435 ms long) underrun!!! (at least -477981.436 ms long) underrun!!! (at least -477981.436 ms long) underrun!!! (at least -477981.436 ms long) underrun!!! (at least -477981.436 ms long) Unable to handle kernel NULL pointer dereference at virtual address 00000044 pgd = 80004000 [00000044] *pgd=00000000 Internal error: Oops: 17 [#1] SMP ARM Modules linked in: CPU: 0 PID: 0 Comm: swapper/0 Not tainted 3.13.0-rc7-next-20140107+ #607 task: 8089fef0 ti: 80894000 task.ti: 80894000 PC is at dmaengine_pcm_dma_complete+0x10/0x50 LR is at sdma_tasklet+0x128/0x130 pc : [<804aee6c>] lr : [<802e6428>] psr: a0000113 sp : 80895df8 ip : 80895e08 fp : 80895e04 r10: 808f64c0 r9 : bfaa41b4 r8 : 00000000 r7 : 8089cd24 r6 : 00000001 r5 : 00000003 r4 : bfaa40e8 r3 : 00000000 r2 : 00000020 r1 : 00000000 r0 : bf15c000 Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c53c7d Table: 4eedc04a DAC: 00000017 Process swapper/0 (pid: 0, stack limit = 0x80894240) Stack: (0x80895df8 to 0x80896000) 5de0: 80895e2c 80895e08 5e00: 802e6428 804aee68 bfaa41b0 00000000 80894000 8089cd24 00000000 808f64c0 5e20: 80895e64 80895e30 8002c4b8 802e630c 80062a6c 80890390 00000040 00000000 5e40: 80896098 80894000 80896080 00000006 00000100 00000006 80895eb4 80895e68 5e60: 8002ca38 8002c434 00000022 00000000 bf807a14 00000001 00200000 ffff915f 5e80: 0000000a 00000000 f4000100 80894000 80890ff0 80894000 00000022 00000000 5ea0: 8063350c 808f4abd 80895ecc 80895eb8 8002cf4c 8002c914 00000184 8089cd24 5ec0: 80895ef4 80895ed0 8000f35c 8002cea8 000000a0 f400010c 8089ce80 80895f18 5ee0: f4000100 80894000 80895f14 80895ef8 80008640 8000f310 8000f734 20000113 5f00: ffffffff 80895f4c 80895f6c 80895f18 800130a4 8000861c 00000001 00000001 5f20: 00000000 8089fef0 00000000 8089c99c 8089c938 808f4abd 80894000 8063350c 5f40: 808f4abd 80895f6c 80895f30 80895f60 80062ab0 8000f734 20000113 ffffffff 5f60: 80895f84 80895f70 8006ad14 8000f718 00000000 8181d140 80895fb4 80895f88 5f80: 80624260 8006acb8 00000001 00000000 806241ac 8089ca38 808f4cc0 8089ca38 5fa0: 808f4cc0 ffffffff 80895ff4 80895fb8 80843b40 806241b8 ffffffff ffffffff 5fc0: 808435d0 00000000 00000000 80883970 10c53c7d 8089c928 8088396c 808a1524 5fe0: 1000406a 00000000 00000000 80895ff8 10008074 80843840 00000000 00000000 Backtrace: [<804aee5c>] (dmaengine_pcm_dma_complete) from [<802e6428>] (sdma_tasklet+0x128) [<802e6300>] (sdma_tasklet) from [<8002c4b8>] (tasklet_action+0x90/0x148) r10:808f64c0 r8:00000000 r7:8089cd24 r6:80894000 r5:00000000 r4:bfaa41b0 [<8002c428>] (tasklet_action) from [<8002ca38>] (__do_softirq+0x130/0x278) r10:00000006 r9:00000100 r8:00000006 r7:80896080 r6:80894000 r5:80896098 r4:00000000 [<8002c908>] (__do_softirq) from [<8002cf4c>] (irq_exit+0xb0/0x100) r10:808f4abd r9:8063350c r8:00000000 r7:00000022 r6:80894000 r5:80890ff0 r4:80894000 [<8002ce9c>] (irq_exit) from [<8000f35c>] (handle_IRQ+0x58/0xb8) r4:8089cd24 r3:00000184 [<8000f304>] (handle_IRQ) from [<80008640>] (gic_handle_irq+0x30/0x64) r8:80894000 r7:f4000100 r6:80895f18 r5:8089ce80 r4:f400010c r3:000000a0 [<80008610>] (gic_handle_irq) from [<800130a4>] (__irq_svc+0x44/0x5c) Exception stack(0x80895f18 to 0x80895f60) 5f00: 00000001 00000001 5f20: 00000000 8089fef0 00000000 8089c99c 8089c938 808f4abd 80894000 8063350c 5f40: 808f4abd 80895f6c 80895f30 80895f60 80062ab0 8000f734 20000113 ffffffff r7:80895f4c r6:ffffffff r5:20000113 r4:8000f734 [<8000f70c>] (arch_cpu_idle) from [<8006ad14>] (cpu_startup_entry+0x68/0x150) [<8006acac>] (cpu_startup_entry) from [<80624260>] (rest_init+0xb4/0xdc) r7:8181d140 r3:00000000 [<806241ac>] (rest_init) from [<80843b40>] (start_kernel+0x30c/0x364) r6:ffffffff r5:808f4cc0 r4:8089ca38 [<80843834>] (start_kernel) from [<10008074>] (0x10008074) Code: e1a0c00d e92dd800 e24cb004 e59030c8 (e5931044) ---[ end trace 076f597fb036d850 ]--- Kernel panic - not syncing: Fatal exception in interrupt CPU1: stopping CPU: 1 PID: 0 Comm: swapper/1 Tainted: G D 3.13.0-rc7-next-20140107+ 7 Backtrace: [<8001233c>] (dump_backtrace) from [<800124d8>] (show_stack+0x18/0x1c) r6:80890ff0 r5:8089cd24 r4:00000000 r3:00000000 [<800124c0>] (show_stack) from [<80629fac>] (dump_stack+0x80/0x9c) [<80629f2c>] (dump_stack) from [<80015494>] (handle_IPI+0x148/0x174) r4:00000001 r3:bf87d100 [<8001534c>] (handle_IPI) from [<8000866c>] (gic_handle_irq+0x5c/0x64) r8:bf89c000 r7:f4000100 r6:bf89df70 r5:8089ce80 r4:f400010c r3:bf87d100 [<80008610>] (gic_handle_irq) from [<800130a4>] (__irq_svc+0x44/0x5c) Exception stack(0xbf89df70 to 0xbf89dfb8) df60: 8000f730 bf89df80 00000000 00000000 df80: 00000000 8089c99c 8089c938 808f4abd bf89c000 8063350c 808f4abd bf89dfc4 dfa0: bf89dfa8 bf89dfb8 80062b9c 8000f734 60000113 ffffffff r7:bf89dfa4 r6:ffffffff r5:60000113 r4:8000f734 [<8000f70c>] (arch_cpu_idle) from [<8006ad14>] (cpu_startup_entry+0x68/0x150) [<8006acac>] (cpu_startup_entry) from [<800150e0>] (secondary_start_kernel+0x12) r7:808f4fd0 r3:bf87d100 [<80014fb4>] (secondary_start_kernel) from [<10008704>] (0x10008704) r5:0000001f r4:4f88406a CPU3: stopping CPU: 3 PID: 0 Comm: swapper/3 Tainted: G D 3.13.0-rc7-next-20140107+ 7 Backtrace: [<8001233c>] (dump_backtrace) from [<800124d8>] (show_stack+0x18/0x1c) r6:80890ff0 r5:8089cd24 r4:00000000 r3:00000000 [<800124c0>] (show_stack) from [<80629fac>] (dump_stack+0x80/0x9c) [<80629f2c>] (dump_stack) from [<80015494>] (handle_IPI+0x148/0x174) r4:00000003 r3:bf87e300 [<8001534c>] (handle_IPI) from [<8000866c>] (gic_handle_irq+0x5c/0x64) r8:bf8a0000 r7:f4000100 r6:bf8a1f70 r5:8089ce80 r4:f400010c r3:bf87e300 [<80008610>] (gic_handle_irq) from [<800130a4>] (__irq_svc+0x44/0x5c) Exception stack(0xbf8a1f70 to 0xbf8a1fb8) 1f60: 8000f730 bf8a1f80 00000000 00000000 1f80: 00000000 8089c99c 8089c938 808f4abd bf8a0000 8063350c 808f4abd bf8a1fc4 1fa0: bf8a1fa8 bf8a1fb8 80062b9c 8000f734 60000013 ffffffff r7:bf8a1fa4 r6:ffffffff r5:60000013 r4:8000f734 [<8000f70c>] (arch_cpu_idle) from [<8006ad14>] (cpu_startup_entry+0x68/0x150) [<8006acac>] (cpu_startup_entry) from [<800150e0>] (secondary_start_kernel+0x12) r7:808f4fd0 r3:bf87e300 [<80014fb4>] (secondary_start_kernel) from [<10008704>] (0x10008704) r5:0000001f r4:4f88406a CPU2: stopping CPU: 2 PID: 0 Comm: swapper/2 Tainted: G D 3.13.0-rc7-next-20140107+ 7 Backtrace: [<8001233c>] (dump_backtrace) from [<800124d8>] (show_stack+0x18/0x1c) r6:80890ff0 r5:8089cd24 r4:00000000 r3:00000000 [<800124c0>] (show_stack) from [<80629fac>] (dump_stack+0x80/0x9c) [<80629f2c>] (dump_stack) from [<80015494>] (handle_IPI+0x148/0x174) r4:00000002 r3:bf87da00 [<8001534c>] (handle_IPI) from [<8000866c>] (gic_handle_irq+0x5c/0x64) r8:bf89e000 r7:f4000100 r6:bf89ff70 r5:8089ce80 r4:f400010c r3:bf87da00 [<80008610>] (gic_handle_irq) from [<800130a4>] (__irq_svc+0x44/0x5c) Exception stack(0xbf89ff70 to 0xbf89ffb8) ff60: 8000f730 bf89ff80 00000000 00000000 ff80: 00000000 8089c99c 8089c938 808f4abd bf89e000 8063350c 808f4abd bf89ffc4 ffa0: bf89ffa8 bf89ffb8 80062b9c 8000f734 60000013 ffffffff r7:bf89ffa4 r6:ffffffff r5:60000013 r4:8000f734 [<8000f70c>] (arch_cpu_idle) from [<8006ad14>] (cpu_startup_entry+0x68/0x150) [<8006acac>] (cpu_startup_entry) from [<800150e0>] (secondary_start_kernel+0x12) r7:808f4fd0 r3:bf87da00 [<80014fb4>] (secondary_start_kernel) from [<10008704>] (0x10008704) r5:0000001f r4:4f88406a drm_kms_helper: panic occurred, switching back to text console I haven't started debugging it, but if this looks familiar to someone, please let me know. Regards, Fabio Estevam

3 10

[alsa-devel] [PATCH 1/2] ALSA: Remove memory accounting in memalloc helper
by Takashi Iwai 08 Jan '14

08 Jan '14

It's almost superfluous, and doesn't help much for real uses. Let's reduce the layer size. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 31 ++----------------------------- 1 file changed, 2 insertions(+), 29 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 5e1c7bc73b29..bfaf4f04d91d 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -63,18 +63,6 @@ struct snd_mem_list { * */ -static long snd_allocated_pages; /* holding the number of allocated pages */ - -static inline void inc_snd_pages(int order) -{ - snd_allocated_pages += 1 << order; -} - -static inline void dec_snd_pages(int order) -{ - snd_allocated_pages -= 1 << order; -} - /** * snd_malloc_pages - allocate pages with the given size * @size: the size to allocate in bytes @@ -87,7 +75,6 @@ static inline void dec_snd_pages(int order) void *snd_malloc_pages(size_t size, gfp_t gfp_flags) { int pg; - void *res; if (WARN_ON(!size)) return NULL; @@ -95,9 +82,7 @@ void *snd_malloc_pages(size_t size, gfp_t gfp_flags) return NULL; gfp_flags |= __GFP_COMP; /* compound page lets parts be mapped */ pg = get_order(size); - if ((res = (void *) __get_free_pages(gfp_flags, pg)) != NULL) - inc_snd_pages(pg); - return res; + return (void *) __get_free_pages(gfp_flags, pg); } /** @@ -114,7 +99,6 @@ void snd_free_pages(void *ptr, size_t size) if (ptr == NULL) return; pg = get_order(size); - dec_snd_pages(pg); free_pages((unsigned long) ptr, pg); } @@ -129,7 +113,6 @@ void snd_free_pages(void *ptr, size_t size) static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) { int pg; - void *res; gfp_t gfp_flags; if (WARN_ON(!dma)) @@ -139,11 +122,7 @@ static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *d | __GFP_COMP /* compound page lets parts be mapped */ | __GFP_NORETRY /* don't trigger OOM-killer */ | __GFP_NOWARN; /* no stack trace print - this call is non-critical */ - res = dma_alloc_coherent(dev, PAGE_SIZE << pg, dma, gfp_flags); - if (res != NULL) - inc_snd_pages(pg); - - return res; + return dma_alloc_coherent(dev, PAGE_SIZE << pg, dma, gfp_flags); } /* free the coherent DMA pages */ @@ -155,7 +134,6 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, if (ptr == NULL) return; pg = get_order(size); - dec_snd_pages(pg); dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); } @@ -432,14 +410,11 @@ static struct proc_dir_entry *snd_mem_proc; static int snd_mem_proc_read(struct seq_file *seq, void *offset) { - long pages = snd_allocated_pages >> (PAGE_SHIFT-12); struct snd_mem_list *mem; int devno; static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG" }; mutex_lock(&list_mutex); - seq_printf(seq, "pages : %li bytes (%li pages per %likB)\n", - pages * PAGE_SIZE, pages, PAGE_SIZE / 1024); devno = 0; list_for_each_entry(mem, &mem_list_head, list) { devno++; @@ -581,8 +556,6 @@ static void __exit snd_mem_exit(void) { remove_proc_entry(SND_MEM_PROC_FILE, NULL); free_all_reserved_pages(); - if (snd_allocated_pages > 0) - printk(KERN_ERR "snd-malloc: Memory leak? pages not freed = %li\n", snd_allocated_pages); } -- 1.8.5.2

1 1

[alsa-devel] [PATCH 00/11] ASoC/MFD: twl4030 codec local register cache removal
by Peter Ujfalusi 08 Jan '14

08 Jan '14

Hi, After the twl6040 codec, now it is time for the twl4030 codec driver to loose it's local reg cache and let the MFD core's regmap to take care of the caching. We have different functions behind of the 0x49 i2c address in twl (where the audio registers can be found between 0x01 - 0x49). I mark all non audio registers as volatile for now to avoid any side effect of the caching. Gradually other functionalities can also move their caching to regmap to simplify the code. I have included couple of clean up patches after the local regcache removal to the series to freshen up the code a bit. Regards, Peter --- Peter Ujfalusi (11): MFD: twl-core: Simplify IO wrapper functions by moving common code out MFD: twl-core: API to set the regcache bypass for a given regmap in twl MFD: twl-core: Enable regcache for audio registers ASoC: twl4030: Separate write condition checking from I/O function ASoC: twl4030: Remove check defaults functionality ASoC: twl4030: Remove reset registers functionality ASoC: twl4030: Introduce local ctl register cache ASoC: twl4030: Remove local reg cache ASoC: twl4030: Parameter alignment fixes (for code consistency) ASoC: twl4030: Move the ctl cache update local to twl4030_write() function ASoC: twl4030: Pass the twl4030_priv directly to twl4030_can_write_to_chip() drivers/mfd/twl-core.c | 190 +++++++++++++++++++---- include/linux/i2c/twl.h | 5 +- sound/soc/codecs/twl4030.c | 376 ++++++++++++++++----------------------------- 3 files changed, 300 insertions(+), 271 deletions(-) -- 1.8.5.2

3 21

[alsa-devel] [PATCH] ASoC: ssm2602: add 16kHz sampling rate support
by Andreas Pretzsch 08 Jan '14

08 Jan '14

SSM260x also supports 16kHz with external master clocks of 12.000MHz, 12.288MHz and 18.432MHz. Add matching coefficients, update constraints and announced rates. Signed-off-by: Andreas Pretzsch <apr(a)cn-eng.de> --- patch against topic/ssm2602 of kernel.org/.../broonie/sound.git Values taken from SSM2603 datasheet (2602/2603/2604 identical here). Tested on real hardware with 12.288MHz master clock. Useful e.g. for wideband codecs like G.722. sound/soc/codecs/ssm2602.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index c6dd485..af76bbd 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -194,7 +194,7 @@ static const struct snd_soc_dapm_route ssm2604_routes[] = { }; static const unsigned int ssm2602_rates_12288000[] = { - 8000, 32000, 48000, 96000, + 8000, 16000, 32000, 48000, 96000, }; static struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = { @@ -231,6 +231,11 @@ static const struct ssm2602_coeff ssm2602_coeff_table[] = { {18432000, 32000, SSM2602_COEFF_SRATE(0x6, 0x1, 0x0)}, {12000000, 32000, SSM2602_COEFF_SRATE(0x6, 0x0, 0x1)}, + /* 16k */ + {12288000, 16000, SSM2602_COEFF_SRATE(0x5, 0x0, 0x0)}, + {18432000, 16000, SSM2602_COEFF_SRATE(0x5, 0x1, 0x0)}, + {12000000, 16000, SSM2602_COEFF_SRATE(0xa, 0x0, 0x1)}, + /* 8k */ {12288000, 8000, SSM2602_COEFF_SRATE(0x3, 0x0, 0x0)}, {18432000, 8000, SSM2602_COEFF_SRATE(0x3, 0x1, 0x0)}, @@ -473,9 +478,10 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_32000 |\ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ - SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000) #define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -- 1.7.10.4

3 2

[alsa-devel] [PATCH v3 0/5] ASoC: add CSR SiRFSoC sound drivers
by RongJun Ying 08 Jan '14

08 Jan '14

From: Rongjun Ying <rongjun.ying(a)csr.com> This patchset adds CSR SiRFSoC sound drivers including: 1. the platform DMA driver which will be shard by all DAI 2. the I2S CPU DAI driver 3. the USP-based PCM CPU DAI driver 4. CPU DAI and Codec driver for internal chip codec 5. the mach driver for EVB board using internal codec -v3: 1. Use devm_* API for driver 2. Remove the extcon stuff code 3. Calculated automatically the div vaule at runtime based on the sample rate 4. Automatically discovering the configuration of pcm hardware from the dmaengine driver 5. Add binding documents Rongjun Ying (5): ASoC: sirf: add sirf platform driver which provides DMA ASoC: sirf: add I2S CPU DAI driver ASoC: usp-pcm: add CPU DAI driver for PCM simulated from USP ASoC: sirf-soc-inner: add drivers for both CPU and Codec DAIs ASoC: sirf-inner: add mach driver for SiRFSoC internal codec .../bindings/sound/sirf,inner-audio-codec.txt | 41 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sirf/Kconfig | 18 + sound/soc/sirf/Makefile | 11 + sound/soc/sirf/sirf-audio.h | 268 ++++++++ sound/soc/sirf/sirf-i2s.c | 435 +++++++++++++ sound/soc/sirf/sirf-inner.c | 155 +++++ sound/soc/sirf/sirf-pcm.c | 68 ++ sound/soc/sirf/sirf-soc-inner.c | 653 ++++++++++++++++++++ sound/soc/sirf/sirf-usp.c | 463 ++++++++++++++ sound/soc/sirf/sirf-usp.h | 276 +++++++++ 12 files changed, 2390 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sirf,inner-audio-codec.txt create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio.h create mode 100644 sound/soc/sirf/sirf-i2s.c create mode 100644 sound/soc/sirf/sirf-inner.c create mode 100644 sound/soc/sirf/sirf-pcm.c create mode 100644 sound/soc/sirf/sirf-soc-inner.c create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h -- 1.7.5.4

4 14

[alsa-devel] [PATCH] ASoC: fsl-sai: Clean up the code
by Xiubo Li 08 Jan '14

08 Jan '14

Makes the code slightly shorter. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5d38a67..cdd3fa8 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -62,26 +62,25 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, reg_cr2 = FSL_SAI_RCR2; val_cr2 = sai_readl(sai, sai->base + reg_cr2); + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; + switch (clk_id) { case FSL_SAI_CLK_BUS: - val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; val_cr2 |= FSL_SAI_CR2_MSEL_BUS; break; case FSL_SAI_CLK_MAST1: - val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; val_cr2 |= FSL_SAI_CR2_MSEL_MCLK1; break; case FSL_SAI_CLK_MAST2: - val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; val_cr2 |= FSL_SAI_CR2_MSEL_MCLK2; break; case FSL_SAI_CLK_MAST3: - val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; val_cr2 |= FSL_SAI_CR2_MSEL_MCLK3; break; default: return -EINVAL; } + sai_writel(sai, val_cr2, sai->base + reg_cr2); return 0; -- 1.8.4

2 1

[alsa-devel] [PATCH -next] ASoC: ux500: Fix sparse non static symbol warning
by Wei Yongjun 08 Jan '14

08 Jan '14

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fixes the following sparse warning: sound/soc/ux500/ux500_msp_i2s.c:649:5: warning: symbol 'ux500_msp_i2s_of_init_msp' was not declared. Should it be static? Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/ux500/ux500_msp_i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 7f2a4ac..959d7b4 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -646,9 +646,9 @@ int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir) } -int ux500_msp_i2s_of_init_msp(struct platform_device *pdev, - struct ux500_msp *msp, - struct msp_i2s_platform_data **platform_data) +static int ux500_msp_i2s_of_init_msp(struct platform_device *pdev, + struct ux500_msp *msp, + struct msp_i2s_platform_data **platform_data) { struct msp_i2s_platform_data *pdata;

5 5

[alsa-devel] [PATCH] Patched to allow fine-grained control of PLL and clock dividers
by Daniel Matuschek 08 Jan '14

08 Jan '14

Changes: 1. allow the driver to accept S32_LE PCM streams 2. fine-grained control of the PLL generation WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using the "pllid" argument of the "set_pll" function is is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. When pllid=0, the behaviour of the driver does not change. --- sound/soc/codecs/wm8804.c | 25 +++++++++++++++++-------- sound/soc/codecs/wm8804.h | 7 +++++++ 2 files changed, 24 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 1704b1e..f3f73b2 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -2,6 +2,7 @@ * wm8804.c -- WM8804 S/PDIF transceiver driver * * Copyright 2010-11 Wolfson Microelectronics plc + * patched by Daniel Matuschek <info(a)crazyaudio.com> to allow fine-grained + * control of PLL and dividers * * Author: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> * @@ -277,6 +278,7 @@ static int wm8804_hw_params(struct snd_pcm_substream *substream, blen = 0x1; break; case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: blen = 0x2; break; default: @@ -318,7 +320,7 @@ static struct { #define FIXED_PLL_SIZE ((1ULL << 22) * 10) static int pll_factors(struct pll_div *pll_div, unsigned int target, - unsigned int source) + unsigned int source, int mclk_div) { u64 Kpart; unsigned long int K, Ndiv, Nmod, tmp; @@ -332,9 +334,14 @@ static int pll_factors(struct pll_div *pll_div, unsigned int target, tmp = target * post_table[i].div; if (tmp >= 90000000 && tmp <= 100000000) { pll_div->freqmode = post_table[i].freqmode; - pll_div->mclkdiv = post_table[i].mclkdiv; - target *= post_table[i].div; - break; + if ((mclk_div == WM8804_MCLKDIV_DONTCARE) || + ((post_table[i].mclkdiv == 1) && + (mclk_div == WM8804_MCLKDIV_1)) || + ((post_table[i].mclkdiv == 0) && + (mclk_div == WM8804_MCLKDIV_0))) { + pll_div->mclkdiv = post_table[i].mclkdiv; + target *= post_table[i].div; + break; + } } } @@ -388,7 +395,7 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, int ret; struct pll_div pll_div; - ret = pll_factors(&pll_div, freq_out, freq_in); + ret = pll_factors(&pll_div, freq_out, freq_in, pll_id); if (ret) return ret; @@ -641,7 +648,8 @@ static const struct snd_soc_dai_ops wm8804_dai_ops = { }; #define WM8804_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE) #define WM8804_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ @@ -674,7 +682,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .suspend = wm8804_suspend, .resume = wm8804_resume, .set_bias_level = wm8804_set_bias_level, - .idle_bias_off = true, + .idle_bias_off = false, .controls = wm8804_snd_controls, .num_controls = ARRAY_SIZE(wm8804_snd_controls), @@ -760,6 +768,7 @@ static int wm8804_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8804, &wm8804_dai, 1); + return ret; } diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index 8ec14f5..0365177 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -58,4 +58,11 @@ #define WM8804_CLKOUT_DIV 1 +#define WM8804_MCLKDIV_DONTCARE 0 +#define WM8804_MCLKDIV_0 1 +#define WM8804_MCLKDIV_1 2 +#define WM8804_PLL_MCLKDIV_DONTCARE WM8804_MCLKDIV_DONTCARE +#define WM8804_PLL_MCLKDIV_0 WM8804_MCLKDIV_0 +#define WM8804_PLL_MCLKDIV_1 WM8804_MCLKDIV_1 + #endif /* _WM8804_H */ -- 1.7.9.5

3 2

[alsa-devel] [PATCH 1/2] ASoC: utils: Add internal call to determine if DAI is dummy.
by Liam Girdwood 08 Jan '14

08 Jan '14

Provide a quick way to tell if a DAI is a dummy DAI or a regular DAI. This is for internal DAPM usage only and is used to determe whether to insert a DAI link connection into the DAPM graph. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-utils.c | 7 +++++++ 2 files changed, 9 insertions(+) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 243d3b6..71f27c4 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -123,6 +123,8 @@ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, int direction); +int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); + struct snd_soc_dai_ops { /* * DAI clocking configuration, all optional. diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 6ebdfd9..7f22ca3 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -119,6 +119,13 @@ static struct snd_soc_dai_driver dummy_dai = { }, }; +int snd_soc_dai_is_dummy(struct snd_soc_dai *dai) +{ + if (dai->driver == &dummy_dai) + return 1; + return 0; +} + static int snd_soc_dummy_probe(struct platform_device *pdev) { int ret; -- 1.8.3.2

3 4

[alsa-devel] [PATCH] alsa-utils: fix pcm_read() return value
by Olivier Langlois 08 Jan '14

08 Jan '14

Because of the way the pcm_read() functions are currently used, returning rcount or result is equivalent but I feel it is more accurate to return 'result'. Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> --- aplay/aplay.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index e0631c4..69e8bda 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -2039,7 +2039,7 @@ static ssize_t pcm_read(u_char *data, size_t rcount) data += r * bits_per_frame / 8; } } - return rcount; + return result; } static ssize_t pcm_readv(u_char **data, unsigned int channels, size_t rcount) @@ -2084,7 +2084,7 @@ static ssize_t pcm_readv(u_char **data, unsigned int channels, size_t rcount) count -= r; } } - return rcount; + return result; } /* -- 1.8.5.2

2 1

[alsa-devel] [PATCH] ALSA: hda - Split Thinkpad ACPI-related code
by Takashi Iwai 08 Jan '14

08 Jan '14

Both patch_realtek.c and patch_conexant.c contain the fairy same code snippet for supporting Thinkpad ACPI LED controls. Split them into thinkpad_helper.c and include it from both places. Although this isn't the best approach from the code size POV, the probability for coexistence of both Realtek and Conexant codecs on a single machine is pretty low, thus it'll end up with less memory footprint than splitting to yet another module. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 100 ++-------------------------------------- sound/pci/hda/patch_realtek.c | 93 ++----------------------------------- sound/pci/hda/thinkpad_helper.c | 99 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 105 insertions(+), 187 deletions(-) create mode 100644 sound/pci/hda/thinkpad_helper.c diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6df4c35b48c5..4e0ec146553d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3240,102 +3240,8 @@ enum { CXT_FIXUP_THINKPAD_ACPI, }; -#if IS_ENABLED(CONFIG_THINKPAD_ACPI) - -#include <linux/thinkpad_acpi.h> -#include <linux/acpi.h> - -static int (*led_set_func)(int, bool); - -static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, - void **rv) -{ - bool *found = context; - *found = true; - return AE_OK; -} - -static bool is_thinkpad(struct hda_codec *codec) -{ - bool found = false; - if (codec->subsystem_id >> 16 != 0x17aa) - return false; - if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) - return true; - found = false; - return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found; -} - -static void update_tpacpi_mute_led(void *private_data, int enabled) -{ - struct hda_codec *codec = private_data; - struct conexant_spec *spec = codec->spec; - - if (spec->dynamic_eapd) - cx_auto_vmaster_hook(private_data, enabled); - - if (led_set_func) - led_set_func(TPACPI_LED_MUTE, !enabled); -} - -static void update_tpacpi_micmute_led(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) -{ - if (!ucontrol || !led_set_func) - return; - if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { - /* TODO: How do I verify if it's a mono or stereo here? */ - bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]; - led_set_func(TPACPI_LED_MICMUTE, !val); - } -} - -static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - struct conexant_spec *spec = codec->spec; - - bool removefunc = false; - - if (action == HDA_FIXUP_ACT_PROBE) { - if (!is_thinkpad(codec)) - return; - if (!led_set_func) - led_set_func = symbol_request(tpacpi_led_set); - if (!led_set_func) { - snd_printk(KERN_WARNING "Failed to find thinkpad-acpi symbol tpacpi_led_set\n"); - return; - } - - removefunc = true; - if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { - spec->gen.vmaster_mute.hook = update_tpacpi_mute_led; - removefunc = false; - } - if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { - if (spec->gen.num_adc_nids > 1) - snd_printdd("Skipping micmute LED control due to several ADCs"); - else { - spec->gen.cap_sync_hook = update_tpacpi_micmute_led; - removefunc = false; - } - } - } - - if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { - symbol_put(tpacpi_led_set); - led_set_func = NULL; - } -} - -#else - -static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ -} - -#endif +/* for hda_fixup_thinkpad_acpi() */ +#include "thinkpad_helper.c" static void cxt_fixup_stereo_dmic(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -3492,7 +3398,7 @@ static const struct hda_fixup cxt_fixups[] = { }, [CXT_FIXUP_THINKPAD_ACPI] = { .type = HDA_FIXUP_FUNC, - .v.func = cxt_fixup_thinkpad_acpi, + .v.func = hda_fixup_thinkpad_acpi, }, }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 74b496b0d179..d35cf3dc6711 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3781,95 +3781,8 @@ static void alc290_fixup_mono_speakers(struct hda_codec *codec, } } -#if IS_ENABLED(CONFIG_THINKPAD_ACPI) - -#include <linux/thinkpad_acpi.h> -#include <linux/acpi.h> - -static int (*led_set_func)(int, bool); - -static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, - void **rv) -{ - bool *found = context; - *found = true; - return AE_OK; -} - -static bool is_thinkpad(struct hda_codec *codec) -{ - bool found = false; - if (codec->subsystem_id >> 16 != 0x17aa) - return false; - if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) - return true; - found = false; - return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found; -} - -static void update_tpacpi_mute_led(void *private_data, int enabled) -{ - if (led_set_func) - led_set_func(TPACPI_LED_MUTE, !enabled); -} - -static void update_tpacpi_micmute_led(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) -{ - if (!ucontrol || !led_set_func) - return; - if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { - /* TODO: How do I verify if it's a mono or stereo here? */ - bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]; - led_set_func(TPACPI_LED_MICMUTE, !val); - } -} - -static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - struct alc_spec *spec = codec->spec; - bool removefunc = false; - - if (action == HDA_FIXUP_ACT_PROBE) { - if (!is_thinkpad(codec)) - return; - if (!led_set_func) - led_set_func = symbol_request(tpacpi_led_set); - if (!led_set_func) { - snd_printk(KERN_WARNING "Failed to find thinkpad-acpi symbol tpacpi_led_set\n"); - return; - } - - removefunc = true; - if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { - spec->gen.vmaster_mute.hook = update_tpacpi_mute_led; - removefunc = false; - } - if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { - if (spec->gen.num_adc_nids > 1) - snd_printdd("Skipping micmute LED control due to several ADCs"); - else { - spec->gen.cap_sync_hook = update_tpacpi_micmute_led; - removefunc = false; - } - } - } - - if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { - symbol_put(tpacpi_led_set); - led_set_func = NULL; - } -} - -#else - -static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ -} - -#endif +/* for hda_fixup_thinkpad_acpi() */ +#include "thinkpad_helper.c" enum { ALC269_FIXUP_SONY_VAIO, @@ -4273,7 +4186,7 @@ static const struct hda_fixup alc269_fixups[] = { }, [ALC269_FIXUP_THINKPAD_ACPI] = { .type = HDA_FIXUP_FUNC, - .v.func = alc_fixup_thinkpad_acpi, + .v.func = hda_fixup_thinkpad_acpi, }, [ALC255_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c new file mode 100644 index 000000000000..8492b8a3ce47 --- /dev/null +++ b/sound/pci/hda/thinkpad_helper.c @@ -0,0 +1,99 @@ +/* Helper functions for Thinkpad LED control; + * to be included from codec driver + */ + +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) + +#include <linux/acpi.h> +#include <linux/thinkpad_acpi.h> + +static int (*led_set_func)(int, bool); + +static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, + void **rv) +{ + bool *found = context; + *found = true; + return AE_OK; +} + +static bool is_thinkpad(struct hda_codec *codec) +{ + bool found = false; + if (codec->subsystem_id >> 16 != 0x17aa) + return false; + if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) + return true; + found = false; + return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found; +} + +static void update_tpacpi_mute_led(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + struct hda_gen_spec *spec = codec->spec; + + if (spec->vmaster_mute.hook) + spec->vmaster_mute.hook(private_data, enabled); + + if (led_set_func) + led_set_func(TPACPI_LED_MUTE, !enabled); +} + +static void update_tpacpi_micmute_led(struct hda_codec *codec, + struct snd_ctl_elem_value *ucontrol) +{ + if (!ucontrol || !led_set_func) + return; + if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { + /* TODO: How do I verify if it's a mono or stereo here? */ + bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]; + led_set_func(TPACPI_LED_MICMUTE, !val); + } +} + +static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct hda_gen_spec *spec = codec->spec; + bool removefunc = false; + + if (action == HDA_FIXUP_ACT_PROBE) { + if (!is_thinkpad(codec)) + return; + if (!led_set_func) + led_set_func = symbol_request(tpacpi_led_set); + if (!led_set_func) { + snd_printk(KERN_WARNING "Failed to find thinkpad-acpi symbol tpacpi_led_set\n"); + return; + } + + removefunc = true; + if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { + spec->vmaster_mute.hook = update_tpacpi_mute_led; + removefunc = false; + } + if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { + if (spec->num_adc_nids > 1) + snd_printdd("Skipping micmute LED control due to several ADCs"); + else { + spec->cap_sync_hook = update_tpacpi_micmute_led; + removefunc = false; + } + } + } + + if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { + symbol_put(tpacpi_led_set); + led_set_func = NULL; + } +} + +#else /* CONFIG_THINKPAD_ACPI */ + +static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ +} + +#endif /* CONFIG_THINKPAD_ACPI */ -- 1.8.5.2

1 0

[alsa-devel] [PATCH] ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constant
by Lars-Peter Clausen 08 Jan '14

08 Jan '14

The driver defines ADAU1701_SEROCTL_WORD_LEN_16 as 0x10 while it should be b10, so 0x2. This patch fixes it. Reported-by: Magnus Reftel <magnus.reftel(a)lockless.no> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/adau1701.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index ebff112..adee866 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -71,7 +71,7 @@ #define ADAU1701_SEROCTL_WORD_LEN_24 0x0000 #define ADAU1701_SEROCTL_WORD_LEN_20 0x0001 -#define ADAU1701_SEROCTL_WORD_LEN_16 0x0010 +#define ADAU1701_SEROCTL_WORD_LEN_16 0x0002 #define ADAU1701_SEROCTL_WORD_LEN_MASK 0x0003 #define ADAU1701_AUXNPOW_VBPD 0x40 -- 1.8.0

1 0

[alsa-devel] [PATCH] ALSA: compress: add num_sample_rates in snd_codec_desc
by Vinod Koul 08 Jan '14

08 Jan '14

this gives ability to convey the valid values of supported rates in sample_rates array Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/uapi/sound/compress_params.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index b62b24b..165e705 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -326,6 +326,7 @@ union snd_codec_options { /** struct snd_codec_desc - description of codec capabilities * @max_ch: Maximum number of audio channels * @sample_rates: Sampling rates in Hz, use values like 48000 for this + * @num_sample_rates: Number of valid values in sample_rates array * @bit_rate: Indexed array containing supported bit rates * @num_bitrates: Number of valid values in bit_rate array * @rate_control: value is specified by SND_RATECONTROLMODE defines. @@ -348,6 +349,7 @@ union snd_codec_options { struct snd_codec_desc { __u32 max_ch; __u32 sample_rates[MAX_NUM_SAMPLE_RATES]; + __u32 num_sample_rates; __u32 bit_rate[MAX_NUM_BITRATES]; __u32 num_bitrates; __u32 rate_control; -- 1.7.0.4

3 9

[alsa-devel] [PATCH v2] ASoC: soc-core: add snd_soc_of_parse_master_clkdir() for dt
by Xiubo Li 08 Jan '14

08 Jan '14

This patch adds snd_soc_of_parse_master_clkdir() and supports below style on dt: master-clkdir-out; If this property is absent 0 will be returned, 1 otherwise. 0 : SND_SOC_CLOCK_IN 1 : SND_SOC_CLOCK_OUT Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 11 +++++++++++ 2 files changed, 12 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5a049d9..690f4f6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1170,6 +1170,7 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname); int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); +unsigned int snd_soc_of_parse_master_clkdir(struct device_node *np); unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix); int snd_soc_of_get_dai_name(struct device_node *of_node, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index be88df5..f155207 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4472,6 +4472,17 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_routing); +unsigned int snd_soc_of_parse_master_clkdir(struct device_node *np) +{ + /* + * check "master-clkdir-out" + * false means SND_SOC_CLOCK_IN + * true means SND_SOC_CLOCK_OUT + */ + return !!of_property_read_bool(np, "master-clkdir-out"); +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_master_clkdir); + unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix) { -- 1.8.4

3 5

[alsa-devel] exceprionally large value
by Johan Esvelt 07 Jan '14

07 Jan '14

Hi, My PC crashed and these are the last 3 lines in /var/log/messages: -------------------------8<---------------------- Jan 7 19:11:48 charles pulseaudio[2646]: alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 416448 bytes (2360 ms). Jan 7 19:11:48 charles pulseaudio[2646]: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_emu10k1'. Please report this issue to the ALSA developers. Jan 7 19:11:48 charles pulseaudio[2646]: alJan 7 19:20:03 charles kernel: imklog 5.8.10, log source = /proc/kmsg started. -------------------------------8<----------------- I am running Centos 6.5: Linux version 2.6.32-358.18.1.el6.x86_64 (mockbuild(a)c6b10.bsys.dev.centos.org) (gcc version 4.4.7 20120313 (Red Hat 4.4.7-3) (GCC) ) #1 SMP Wed Aug 28 17:19:38 UTC 2013 What can I do to prevent this from happening again? Best regards, Johan Esvelt

1 0

[alsa-devel] [PATCH] ASoC: Docs: Update the Overview document
by Liam Girdwood 07 Jan '14

07 Jan '14

Update the ASoC overview to bring it up to date with the current code base and include multi-component. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Documentation/sound/alsa/soc/overview.txt | 27 ++++++++++++++++++--------- 1 file changed, 18 insertions(+), 9 deletions(-) diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt index 138ac88..ff88f52 100644 --- a/Documentation/sound/alsa/soc/overview.txt +++ b/Documentation/sound/alsa/soc/overview.txt @@ -49,18 +49,23 @@ features :- * Machine specific controls: Allow machines to add controls to the sound card (e.g. volume control for speaker amplifier). -To achieve all this, ASoC basically splits an embedded audio system into 3 -components :- +To achieve all this, ASoC basically splits an embedded audio system into +multiple re-usable component drivers :- - * Codec driver: The codec driver is platform independent and contains audio - controls, audio interface capabilities, codec DAPM definition and codec IO - functions. + * Codec class drivers: The codec class driver is platform independent and + contains audio controls, audio interface capabilities, codec DAPM + definition and codec IO functions. This class extends to BT, FM and MODEM + ICs if required. Codec class drivers should be generic code that can run + on any architecture and machine. - * Platform driver: The platform driver contains the audio DMA engine and audio - interface drivers (e.g. I2S, AC97, PCM) for that platform. + * Platform class drivers: The platform class driver includes the audio DMA + engine driver, digital audio interface (DAI) drivers (e.g. I2S, AC97, PCM) + and any audio DSP drivers for that platform. - * Machine driver: The machine driver handles any machine specific controls and - audio events (e.g. turning on an amp at start of playback). + * Machine class driver: The machine driver class acts as the glue that + decribes and binds the other component drivers together to form an ALSA + "sound card device". It handles any machine specific controls and + machine level audio events (e.g. turning on an amp at start of playback). Documentation @@ -84,3 +89,7 @@ machine.txt: Machine driver internals. pop_clicks.txt: How to minimise audio artifacts. clocking.txt: ASoC clocking for best power performance. + +jack.txt: ASoC jack detection. + +DPCM.txt: Dynamic PCM - Describes DPCM with DSP examples. -- 1.8.3.2

2 3

[alsa-devel] [PATCHv10] dmaengine: Add support for BCM2835
by Florian Meier 07 Jan '14

07 Jan '14

Add support for DMA controller of BCM2835 as used in the Raspberry Pi. Currently it only supports cyclic DMA. Signed-off-by: Florian Meier <florian.meier(a)koalo.de> --- Merging and rebasing was easier than I feared and I found an additional API change (DMA_SUCCESS). Andy: I would be happy to have your Reviewed-by again, but I removed it because of these changes. Changes compared to v9: - Added commas to device tree documentation - DMA_SUCCESS -> DMA_COMPLETE - Removed #if defined(CONFIG_OF) - Use of dma_get_any_slave_channel - Use of dma_set_mask_and_coherent - Removed if (pdev->dev.of_node) in probe .../devicetree/bindings/dma/bcm2835-dma.txt | 57 ++ drivers/dma/Kconfig | 6 + drivers/dma/Makefile | 1 + drivers/dma/bcm2835-dma.c | 706 +++++++++++++++++++++ 4 files changed, 770 insertions(+) create mode 100644 Documentation/devicetree/bindings/dma/bcm2835-dma.txt create mode 100644 drivers/dma/bcm2835-dma.c diff --git a/Documentation/devicetree/bindings/dma/bcm2835-dma.txt b/Documentation/devicetree/bindings/dma/bcm2835-dma.txt new file mode 100644 index 0000000..1396078 --- /dev/null +++ b/Documentation/devicetree/bindings/dma/bcm2835-dma.txt @@ -0,0 +1,57 @@ +* BCM2835 DMA controller + +The BCM2835 DMA controller has 16 channels in total. +Only the lower 13 channels have an associated IRQ. +Some arbitrary channels are used by the firmware +(1,3,6,7 in the current firmware version). +The channels 0,2 and 3 have special functionality +and should not be used by the driver. + +Required properties: +- compatible: Should be "brcm,bcm2835-dma". +- reg: Should contain DMA registers location and length. +- interrupts: Should contain the DMA interrupts associated + to the DMA channels in ascending order. +- #dma-cells: Must be <1>, the cell in the dmas property of the + client device represents the DREQ number. +- brcm,dma-channel-mask: Bit mask representing the channels + not used by the firmware in ascending order, + i.e. first channel corresponds to LSB. + +Example: + +dma: dma@7e007000 { + compatible = "brcm,bcm2835-dma"; + reg = <0x7e007000 0xf00>; + interrupts = <1 16>, + <1 17>, + <1 18>, + <1 19>, + <1 20>, + <1 21>, + <1 22>, + <1 23>, + <1 24>, + <1 25>, + <1 26>, + <1 27>, + <1 28>; + + #dma-cells = <1>; + brcm,dma-channel-mask = <0x7f35>; +}; + +DMA clients connected to the BCM2835 DMA controller must use the format +described in the dma.txt file, using a two-cell specifier for each channel. + +Example: + +bcm2835_i2s: i2s@7e203000 { + compatible = "brcm,bcm2835-i2s"; + reg = < 0x7e203000 0x20>, + < 0x7e101098 0x02>; + + dmas = <&dma 2>, + <&dma 3>; + dma-names = "tx", "rx"; +}; diff --git a/drivers/dma/Kconfig b/drivers/dma/Kconfig index c823daa..6f74f8c 100644 --- a/drivers/dma/Kconfig +++ b/drivers/dma/Kconfig @@ -304,6 +304,12 @@ config DMA_OMAP select DMA_ENGINE select DMA_VIRTUAL_CHANNELS +config DMA_BCM2835 + tristate "BCM2835 DMA engine support" + depends on (ARCH_BCM2835 || MACH_BCM2708) + select DMA_ENGINE + select DMA_VIRTUAL_CHANNELS + config TI_CPPI41 tristate "AM33xx CPPI41 DMA support" depends on ARCH_OMAP diff --git a/drivers/dma/Makefile b/drivers/dma/Makefile index 0ce2da9..0a6f08e 100644 --- a/drivers/dma/Makefile +++ b/drivers/dma/Makefile @@ -38,6 +38,7 @@ obj-$(CONFIG_EP93XX_DMA) += ep93xx_dma.o obj-$(CONFIG_DMA_SA11X0) += sa11x0-dma.o obj-$(CONFIG_MMP_TDMA) += mmp_tdma.o obj-$(CONFIG_DMA_OMAP) += omap-dma.o +obj-$(CONFIG_DMA_BCM2835) += bcm2835-dma.o obj-$(CONFIG_MMP_PDMA) += mmp_pdma.o obj-$(CONFIG_DMA_JZ4740) += dma-jz4740.o obj-$(CONFIG_TI_CPPI41) += cppi41.o diff --git a/drivers/dma/bcm2835-dma.c b/drivers/dma/bcm2835-dma.c new file mode 100644 index 0000000..6ae0708 --- /dev/null +++ b/drivers/dma/bcm2835-dma.c @@ -0,0 +1,706 @@ +/* + * BCM2835 DMA engine support + * + * This driver only supports cyclic DMA transfers + * as needed for the I2S module. + * + * Author: Florian Meier <florian.meier(a)koalo.de> + * Copyright 2013 + * + * Based on + * OMAP DMAengine support by Russell King + * + * BCM2708 DMA Driver + * Copyright (C) 2010 Broadcom + * + * Raspberry Pi PCM I2S ALSA Driver + * Copyright (c) by Phil Poole 2013 + * + * MARVELL MMP Peripheral DMA Driver + * Copyright 2012 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ +#include <linux/dmaengine.h> +#include <linux/dma-mapping.h> +#include <linux/err.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/list.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <linux/spinlock.h> +#include <linux/of.h> +#include <linux/of_dma.h> + +#include "virt-dma.h" + +struct bcm2835_dmadev { + struct dma_device ddev; + spinlock_t lock; + void __iomem *base; + struct device_dma_parameters dma_parms; +}; + +struct bcm2835_dma_cb { + uint32_t info; + uint32_t src; + uint32_t dst; + uint32_t length; + uint32_t stride; + uint32_t next; + uint32_t pad[2]; +}; + +struct bcm2835_chan { + struct virt_dma_chan vc; + struct list_head node; + + struct dma_slave_config cfg; + bool cyclic; + unsigned int dreq; + + int ch; + struct bcm2835_desc *desc; + + void __iomem *chan_base; + int irq_number; +}; + +struct bcm2835_desc { + struct virt_dma_desc vd; + enum dma_transfer_direction dir; + + unsigned int control_block_size; + struct bcm2835_dma_cb *control_block_base; + dma_addr_t control_block_base_phys; + + unsigned int frames; + size_t size; +}; + +#define BCM2835_DMA_CS 0x00 +#define BCM2835_DMA_ADDR 0x04 +#define BCM2835_DMA_SOURCE_AD 0x0c +#define BCM2835_DMA_DEST_AD 0x10 +#define BCM2835_DMA_NEXTCB 0x1C + +/* DMA CS Control and Status bits */ +#define BCM2835_DMA_ACTIVE BIT(0) +#define BCM2835_DMA_INT BIT(2) +#define BCM2835_DMA_ISPAUSED BIT(4) /* Pause requested or not active */ +#define BCM2835_DMA_ISHELD BIT(5) /* Is held by DREQ flow control */ +#define BCM2835_DMA_ERR BIT(8) +#define BCM2835_DMA_ABORT BIT(30) /* Stop current CB, go to next, WO */ +#define BCM2835_DMA_RESET BIT(31) /* WO, self clearing */ + +#define BCM2835_DMA_INT_EN BIT(0) +#define BCM2835_DMA_D_INC BIT(4) +#define BCM2835_DMA_D_DREQ BIT(6) +#define BCM2835_DMA_S_INC BIT(8) +#define BCM2835_DMA_S_DREQ BIT(10) + +#define BCM2835_DMA_PER_MAP(x) ((x) << 16) + +#define BCM2835_DMA_DATA_TYPE_S8 1 +#define BCM2835_DMA_DATA_TYPE_S16 2 +#define BCM2835_DMA_DATA_TYPE_S32 4 +#define BCM2835_DMA_DATA_TYPE_S128 16 + +#define BCM2835_DMA_BULK_MASK BIT(0) +#define BCM2835_DMA_FIQ_MASK (BIT(2) | BIT(3)) + +/* Valid only for channels 0 - 14, 15 has its own base address */ +#define BCM2835_DMA_CHAN(n) ((n) << 8) /* Base address */ +#define BCM2835_DMA_CHANIO(base, n) ((base) + BCM2835_DMA_CHAN(n)) + +static inline struct bcm2835_dmadev *to_bcm2835_dma_dev(struct dma_device *d) +{ + return container_of(d, struct bcm2835_dmadev, ddev); +} + +static inline struct bcm2835_chan *to_bcm2835_dma_chan(struct dma_chan *c) +{ + return container_of(c, struct bcm2835_chan, vc.chan); +} + +static inline struct bcm2835_desc *to_bcm2835_dma_desc( + struct dma_async_tx_descriptor *t) +{ + return container_of(t, struct bcm2835_desc, vd.tx); +} + +static void bcm2835_dma_desc_free(struct virt_dma_desc *vd) +{ + struct bcm2835_desc *desc = container_of(vd, struct bcm2835_desc, vd); + dma_free_coherent(desc->vd.tx.chan->device->dev, + desc->control_block_size, + desc->control_block_base, + desc->control_block_base_phys); + kfree(desc); +} + +static int bcm2835_dma_abort(void __iomem *chan_base) +{ + unsigned long cs; + long int timeout = 10000; + + cs = readl(chan_base + BCM2835_DMA_CS); + if (!(cs & BCM2835_DMA_ACTIVE)) + return 0; + + /* Write 0 to the active bit - Pause the DMA */ + writel(0, chan_base + BCM2835_DMA_CS); + + /* Wait for any current AXI transfer to complete */ + while ((cs & BCM2835_DMA_ISPAUSED) && --timeout) { + cpu_relax(); + cs = readl(chan_base + BCM2835_DMA_CS); + } + + /* We'll un-pause when we set of our next DMA */ + if (!timeout) + return -ETIMEDOUT; + + if (!(cs & BCM2835_DMA_ACTIVE)) + return 0; + + /* Terminate the control block chain */ + writel(0, chan_base + BCM2835_DMA_NEXTCB); + + /* Abort the whole DMA */ + writel(BCM2835_DMA_ABORT | BCM2835_DMA_ACTIVE, + chan_base + BCM2835_DMA_CS); + + return 0; +} + +static void bcm2835_dma_start_desc(struct bcm2835_chan *c) +{ + struct virt_dma_desc *vd = vchan_next_desc(&c->vc); + struct bcm2835_desc *d; + + if (!vd) { + c->desc = NULL; + return; + } + + list_del(&vd->node); + + c->desc = d = to_bcm2835_dma_desc(&vd->tx); + + writel(d->control_block_base_phys, c->chan_base + BCM2835_DMA_ADDR); + writel(BCM2835_DMA_ACTIVE, c->chan_base + BCM2835_DMA_CS); +} + +static irqreturn_t bcm2835_dma_callback(int irq, void *data) +{ + struct bcm2835_chan *c = data; + struct bcm2835_desc *d; + unsigned long flags; + + spin_lock_irqsave(&c->vc.lock, flags); + + /* Acknowledge interrupt */ + writel(BCM2835_DMA_INT, c->chan_base + BCM2835_DMA_CS); + + d = c->desc; + + if (d) { + /* TODO Only works for cyclic DMA */ + vchan_cyclic_callback(&d->vd); + } + + /* Keep the DMA engine running */ + writel(BCM2835_DMA_ACTIVE, c->chan_base + BCM2835_DMA_CS); + + spin_unlock_irqrestore(&c->vc.lock, flags); + + return IRQ_HANDLED; +} + +static int bcm2835_dma_alloc_chan_resources(struct dma_chan *chan) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + + dev_dbg(c->vc.chan.device->dev, + "Allocating DMA channel %d\n", c->ch); + + return request_irq(c->irq_number, + bcm2835_dma_callback, 0, "DMA IRQ", c); +} + +static void bcm2835_dma_free_chan_resources(struct dma_chan *chan) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + + vchan_free_chan_resources(&c->vc); + free_irq(c->irq_number, c); + + dev_dbg(c->vc.chan.device->dev, "Freeing DMA channel %u\n", c->ch); +} + +static size_t bcm2835_dma_desc_size(struct bcm2835_desc *d) +{ + return d->size; +} + +static size_t bcm2835_dma_desc_size_pos(struct bcm2835_desc *d, dma_addr_t addr) +{ + unsigned int i; + size_t size; + + for (size = i = 0; i < d->frames; i++) { + struct bcm2835_dma_cb *control_block = + &d->control_block_base[i]; + size_t this_size = control_block->length; + dma_addr_t dma; + + if (d->dir == DMA_DEV_TO_MEM) + dma = control_block->dst; + else + dma = control_block->src; + + if (size) + size += this_size; + else if (addr >= dma && addr < dma + this_size) + size += dma + this_size - addr; + } + + return size; +} + +static enum dma_status bcm2835_dma_tx_status(struct dma_chan *chan, + dma_cookie_t cookie, struct dma_tx_state *txstate) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + struct virt_dma_desc *vd; + enum dma_status ret; + unsigned long flags; + + ret = dma_cookie_status(chan, cookie, txstate); + if (ret == DMA_COMPLETE || !txstate) + return ret; + + spin_lock_irqsave(&c->vc.lock, flags); + vd = vchan_find_desc(&c->vc, cookie); + if (vd) { + txstate->residue = + bcm2835_dma_desc_size(to_bcm2835_dma_desc(&vd->tx)); + } else if (c->desc && c->desc->vd.tx.cookie == cookie) { + struct bcm2835_desc *d = c->desc; + dma_addr_t pos; + + if (d->dir == DMA_MEM_TO_DEV) + pos = readl(c->chan_base + BCM2835_DMA_SOURCE_AD); + else if (d->dir == DMA_DEV_TO_MEM) + pos = readl(c->chan_base + BCM2835_DMA_DEST_AD); + else + pos = 0; + + txstate->residue = bcm2835_dma_desc_size_pos(d, pos); + } else { + txstate->residue = 0; + } + + spin_unlock_irqrestore(&c->vc.lock, flags); + + return ret; +} + +static void bcm2835_dma_issue_pending(struct dma_chan *chan) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + unsigned long flags; + + c->cyclic = true; /* Nothing else is implemented */ + + spin_lock_irqsave(&c->vc.lock, flags); + if (vchan_issue_pending(&c->vc) && !c->desc) + bcm2835_dma_start_desc(c); + + spin_unlock_irqrestore(&c->vc.lock, flags); +} + +static struct dma_async_tx_descriptor *bcm2835_dma_prep_dma_cyclic( + struct dma_chan *chan, dma_addr_t buf_addr, size_t buf_len, + size_t period_len, enum dma_transfer_direction direction, + unsigned long flags, void *context) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + enum dma_slave_buswidth dev_width; + struct bcm2835_desc *d; + dma_addr_t dev_addr; + unsigned int es, sync_type; + unsigned int frame; + + /* Grab configuration */ + if (!is_slave_direction(direction)) { + dev_err(chan->device->dev, "%s: bad direction?\n", __func__); + return NULL; + } + + if (direction == DMA_DEV_TO_MEM) { + dev_addr = c->cfg.src_addr; + dev_width = c->cfg.src_addr_width; + sync_type = BCM2835_DMA_S_DREQ; + } else { + dev_addr = c->cfg.dst_addr; + dev_width = c->cfg.dst_addr_width; + sync_type = BCM2835_DMA_D_DREQ; + } + + /* Bus width translates to the element size (ES) */ + switch (dev_width) { + case DMA_SLAVE_BUSWIDTH_4_BYTES: + es = BCM2835_DMA_DATA_TYPE_S32; + break; + default: + return NULL; + } + + /* Now allocate and setup the descriptor. */ + d = kzalloc(sizeof(*d), GFP_NOWAIT); + if (!d) + return NULL; + + d->dir = direction; + d->frames = buf_len / period_len; + + /* Allocate memory for control blocks */ + d->control_block_size = d->frames * sizeof(struct bcm2835_dma_cb); + d->control_block_base = dma_zalloc_coherent(chan->device->dev, + d->control_block_size, &d->control_block_base_phys, + GFP_NOWAIT); + + if (!d->control_block_base) { + kfree(d); + return NULL; + } + + /* + * Iterate over all frames, create a control block + * for each frame and link them together. + */ + for (frame = 0; frame < d->frames; frame++) { + struct bcm2835_dma_cb *control_block = + &d->control_block_base[frame]; + + /* Setup adresses */ + if (d->dir == DMA_DEV_TO_MEM) { + control_block->info = BCM2835_DMA_D_INC; + control_block->src = dev_addr; + control_block->dst = buf_addr + frame * period_len; + } else { + control_block->info = BCM2835_DMA_S_INC; + control_block->src = buf_addr + frame * period_len; + control_block->dst = dev_addr; + } + + /* Enable interrupt */ + control_block->info |= BCM2835_DMA_INT_EN; + + /* Setup synchronization */ + if (sync_type != 0) + control_block->info |= sync_type; + + /* Setup DREQ channel */ + if (c->dreq != 0) + control_block->info |= + BCM2835_DMA_PER_MAP(c->dreq); + + /* Length of a frame */ + control_block->length = period_len; + d->size += control_block->length; + + /* + * Next block is the next frame. + * This DMA engine driver currently only supports cyclic DMA. + * Therefore, wrap around at number of frames. + */ + control_block->next = d->control_block_base_phys + + sizeof(struct bcm2835_dma_cb) + * ((frame + 1) % d->frames); + } + + return vchan_tx_prep(&c->vc, &d->vd, flags); +} + +static int bcm2835_dma_slave_config(struct bcm2835_chan *c, + struct dma_slave_config *cfg) +{ + if ((cfg->direction == DMA_DEV_TO_MEM && + cfg->src_addr_width != DMA_SLAVE_BUSWIDTH_4_BYTES) || + (cfg->direction == DMA_MEM_TO_DEV && + cfg->dst_addr_width != DMA_SLAVE_BUSWIDTH_4_BYTES) || + !is_slave_direction(cfg->direction)) { + return -EINVAL; + } + + c->cfg = *cfg; + + return 0; +} + +static int bcm2835_dma_terminate_all(struct bcm2835_chan *c) +{ + struct bcm2835_dmadev *d = to_bcm2835_dma_dev(c->vc.chan.device); + unsigned long flags; + int timeout = 10000; + LIST_HEAD(head); + + spin_lock_irqsave(&c->vc.lock, flags); + + /* Prevent this channel being scheduled */ + spin_lock(&d->lock); + list_del_init(&c->node); + spin_unlock(&d->lock); + + /* + * Stop DMA activity: we assume the callback will not be called + * after bcm_dma_abort() returns (even if it does, it will see + * c->desc is NULL and exit.) + */ + if (c->desc) { + c->desc = NULL; + bcm2835_dma_abort(c->chan_base); + + /* Wait for stopping */ + while (--timeout) { + if (!(readl(c->chan_base + BCM2835_DMA_CS) & + BCM2835_DMA_ACTIVE)) + break; + + cpu_relax(); + } + + if (!timeout) + dev_err(d->ddev.dev, "DMA transfer could not be terminated\n"); + } + + vchan_get_all_descriptors(&c->vc, &head); + spin_unlock_irqrestore(&c->vc.lock, flags); + vchan_dma_desc_free_list(&c->vc, &head); + + return 0; +} + +static int bcm2835_dma_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd, + unsigned long arg) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + + switch (cmd) { + case DMA_SLAVE_CONFIG: + return bcm2835_dma_slave_config(c, + (struct dma_slave_config *)arg); + + case DMA_TERMINATE_ALL: + return bcm2835_dma_terminate_all(c); + + default: + return -ENXIO; + } +} + +static int bcm2835_dma_chan_init(struct bcm2835_dmadev *d, int chan_id, int irq) +{ + struct bcm2835_chan *c; + + c = devm_kzalloc(d->ddev.dev, sizeof(*c), GFP_KERNEL); + if (!c) + return -ENOMEM; + + c->vc.desc_free = bcm2835_dma_desc_free; + vchan_init(&c->vc, &d->ddev); + INIT_LIST_HEAD(&c->node); + + d->ddev.chancnt++; + + c->chan_base = BCM2835_DMA_CHANIO(d->base, chan_id); + c->ch = chan_id; + c->irq_number = irq; + + return 0; +} + +static void bcm2835_dma_free(struct bcm2835_dmadev *od) +{ + struct bcm2835_chan *c, *next; + + list_for_each_entry_safe(c, next, &od->ddev.channels, + vc.chan.device_node) { + list_del(&c->vc.chan.device_node); + tasklet_kill(&c->vc.task); + } +} + +static const struct of_device_id bcm2835_dma_of_match[] = { + { .compatible = "brcm,bcm2835-dma", }, + {}, +}; +MODULE_DEVICE_TABLE(of, bcm2835_dma_of_match); + +static struct dma_chan *bcm2835_dma_xlate(struct of_phandle_args *spec, + struct of_dma *ofdma) +{ + struct bcm2835_dmadev *d = ofdma->of_dma_data; + struct dma_chan *chan; + + chan = dma_get_any_slave_channel(&d->ddev); + if (!chan) + return NULL; + + /* Set DREQ from param */ + to_bcm2835_dma_chan(chan)->dreq = spec->args[0]; + + return chan; +} + +static int bcm2835_dma_device_slave_caps(struct dma_chan *dchan, + struct dma_slave_caps *caps) +{ + caps->src_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + caps->dstn_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); + caps->cmd_pause = false; + caps->cmd_terminate = true; + + return 0; +} + +static int bcm2835_dma_probe(struct platform_device *pdev) +{ + struct bcm2835_dmadev *od; + struct resource *res; + void __iomem *base; + int rc; + int i; + int irq; + uint32_t chans_available; + + if (!pdev->dev.dma_mask) + pdev->dev.dma_mask = &pdev->dev.coherent_dma_mask; + + rc = dma_set_mask_and_coherent(&pdev->dev, DMA_BIT_MASK(32)); + if (rc) + return rc; + + od = devm_kzalloc(&pdev->dev, sizeof(*od), GFP_KERNEL); + if (!od) + return -ENOMEM; + + pdev->dev.dma_parms = &od->dma_parms; + dma_set_max_seg_size(&pdev->dev, 0x3FFFFFFF); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); + + od->base = base; + + dma_cap_set(DMA_SLAVE, od->ddev.cap_mask); + dma_cap_set(DMA_CYCLIC, od->ddev.cap_mask); + od->ddev.device_alloc_chan_resources = bcm2835_dma_alloc_chan_resources; + od->ddev.device_free_chan_resources = bcm2835_dma_free_chan_resources; + od->ddev.device_tx_status = bcm2835_dma_tx_status; + od->ddev.device_issue_pending = bcm2835_dma_issue_pending; + od->ddev.device_slave_caps = bcm2835_dma_device_slave_caps; + od->ddev.device_prep_dma_cyclic = bcm2835_dma_prep_dma_cyclic; + od->ddev.device_control = bcm2835_dma_control; + od->ddev.dev = &pdev->dev; + INIT_LIST_HEAD(&od->ddev.channels); + spin_lock_init(&od->lock); + + platform_set_drvdata(pdev, od); + + /* Request DMA channel mask from device tree */ + if (of_property_read_u32(pdev->dev.of_node, + "brcm,dma-channel-mask", + &chans_available)) { + dev_err(&pdev->dev, "Failed to get channel mask\n"); + rc = -EINVAL; + goto err_no_dma; + } + + /* + * Do not use the FIQ and BULK channels, + * because they are used by the GPU. + */ + chans_available &= ~(BCM2835_DMA_FIQ_MASK | BCM2835_DMA_BULK_MASK); + + for (i = 0; i < pdev->num_resources; i++) { + irq = platform_get_irq(pdev, i); + if (irq < 0) + break; + + if (chans_available & (1 << i)) { + rc = bcm2835_dma_chan_init(od, i, irq); + if (rc) + goto err_no_dma; + } + } + + dev_dbg(&pdev->dev, "Initialized %i DMA channels\n", i); + + /* Device-tree DMA controller registration */ + rc = of_dma_controller_register(pdev->dev.of_node, + bcm2835_dma_xlate, od); + if (rc) { + dev_err(&pdev->dev, "Failed to register DMA controller\n"); + goto err_no_dma; + } + + rc = dma_async_device_register(&od->ddev); + if (rc) { + dev_err(&pdev->dev, + "Failed to register slave DMA engine device: %d\n", rc); + goto err_no_dma; + } + + dev_dbg(&pdev->dev, "Load BCM2835 DMA engine driver\n"); + + return 0; + +err_no_dma: + bcm2835_dma_free(od); + return rc; +} + +static int bcm2835_dma_remove(struct platform_device *pdev) +{ + struct bcm2835_dmadev *od = platform_get_drvdata(pdev); + + dma_async_device_unregister(&od->ddev); + bcm2835_dma_free(od); + + return 0; +} + +static struct platform_driver bcm2835_dma_driver = { + .probe = bcm2835_dma_probe, + .remove = bcm2835_dma_remove, + .driver = { + .name = "bcm2835-dma", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(bcm2835_dma_of_match), + }, +}; + +module_platform_driver(bcm2835_dma_driver); + +MODULE_ALIAS("platform:bcm2835-dma"); +MODULE_DESCRIPTION("BCM2835 DMA engine driver"); +MODULE_AUTHOR("Florian Meier <florian.meier(a)koalo.de>"); +MODULE_LICENSE("GPL v2"); -- 1.8.3.2

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[alsa-devel] [PATCH] ALSA: hda - Minor code optimization for patch_realtek.c
by Takashi Iwai 07 Jan '14

07 Jan '14

No functional change. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 15 +++------------ 1 file changed, 3 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6a705a81de35..74b496b0d179 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -469,9 +469,7 @@ static bool found_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums) * 7 ~ 0 : Assembly ID * port-A --> pin 39/41, port-E --> pin 14/15, port-D --> pin 35/36 */ -static int alc_subsystem_id(struct hda_codec *codec, - hda_nid_t porta, hda_nid_t porte, - hda_nid_t portd, hda_nid_t porti) +static int alc_subsystem_id(struct hda_codec *codec, const hda_nid_t *ports) { unsigned int ass, tmp, i; unsigned nid; @@ -561,14 +559,7 @@ do_sku: spec->gen.autocfg.line_out_type == AUTO_PIN_HP_OUT)) { hda_nid_t nid; tmp = (ass >> 11) & 0x3; /* HP to chassis */ - if (tmp == 0) - nid = porta; - else if (tmp == 1) - nid = porte; - else if (tmp == 2) - nid = portd; - else if (tmp == 3) - nid = porti; + nid = ports[tmp]; if (found_in_nid_list(nid, spec->gen.autocfg.line_out_pins, spec->gen.autocfg.line_outs)) return 1; @@ -581,7 +572,7 @@ do_sku: * ports contains an array of 4 pin NIDs for port-A, E, D and I */ static void alc_ssid_check(struct hda_codec *codec, const hda_nid_t *ports) { - if (!alc_subsystem_id(codec, ports[0], ports[1], ports[2], ports[3])) { + if (!alc_subsystem_id(codec, ports)) { struct alc_spec *spec = codec->spec; snd_printd("realtek: " "Enable default setup for auto mode as fallback\n"); -- 1.8.5.2

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - firmware patch code cleanup
by Takashi Iwai 07 Jan '14

07 Jan '14

Just a code refactoring: the need_codec flag in hda_patch_item struct can be removed by checking the current mode instead. No functional change. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_hwdep.c | 21 ++++++++++----------- 1 file changed, 10 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 0ec6fb1e9846..79b3dd0d4dd8 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -763,19 +763,18 @@ DEFINE_PARSE_ID_MODE(revision_id); struct hda_patch_item { const char *tag; void (*parser)(char *buf, struct hda_bus *bus, struct hda_codec **retc); - int need_codec; }; static struct hda_patch_item patch_items[NUM_LINE_MODES] = { - [LINE_MODE_CODEC] = { "[codec]", parse_codec_mode, 0 }, - [LINE_MODE_MODEL] = { "[model]", parse_model_mode, 1 }, - [LINE_MODE_VERB] = { "[verb]", parse_verb_mode, 1 }, - [LINE_MODE_PINCFG] = { "[pincfg]", parse_pincfg_mode, 1 }, - [LINE_MODE_HINT] = { "[hint]", parse_hint_mode, 1 }, - [LINE_MODE_VENDOR_ID] = { "[vendor_id]", parse_vendor_id_mode, 1 }, - [LINE_MODE_SUBSYSTEM_ID] = { "[subsystem_id]", parse_subsystem_id_mode, 1 }, - [LINE_MODE_REVISION_ID] = { "[revision_id]", parse_revision_id_mode, 1 }, - [LINE_MODE_CHIP_NAME] = { "[chip_name]", parse_chip_name_mode, 1 }, + [LINE_MODE_CODEC] = { "[codec]", parse_codec_mode }, + [LINE_MODE_MODEL] = { "[model]", parse_model_mode }, + [LINE_MODE_VERB] = { "[verb]", parse_verb_mode }, + [LINE_MODE_PINCFG] = { "[pincfg]", parse_pincfg_mode }, + [LINE_MODE_HINT] = { "[hint]", parse_hint_mode }, + [LINE_MODE_VENDOR_ID] = { "[vendor_id]", parse_vendor_id_mode }, + [LINE_MODE_SUBSYSTEM_ID] = { "[subsystem_id]", parse_subsystem_id_mode }, + [LINE_MODE_REVISION_ID] = { "[revision_id]", parse_revision_id_mode }, + [LINE_MODE_CHIP_NAME] = { "[chip_name]", parse_chip_name_mode }, }; /* check the line starting with '[' -- change the parser mode accodingly */ @@ -846,7 +845,7 @@ int snd_hda_load_patch(struct hda_bus *bus, size_t fw_size, const void *fw_buf) if (*buf == '[') line_mode = parse_line_mode(buf, bus); else if (patch_items[line_mode].parser && - (codec || !patch_items[line_mode].need_codec)) + (codec || line_mode <= LINE_MODE_CODEC)) patch_items[line_mode].parser(buf, bus, &codec); } return 0; -- 1.8.5.2

1 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Correct AD1986A 3stack pin configs
by Takashi Iwai 07 Jan '14

07 Jan '14

The 3stack pin configs for AD1986A codec had incorrect values that resulted in broken mic and line-in. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66621 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index b174eb1567b8..819d4a98abaf 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -281,11 +281,11 @@ static const struct hda_fixup ad1986a_fixups[] = { .v.pins = (const struct hda_pintbl[]) { { 0x1a, 0x02214021 }, /* headphone */ { 0x1b, 0x01014011 }, /* front */ - { 0x1c, 0x01013012 }, /* surround */ - { 0x1d, 0x01019015 }, /* clfe */ + { 0x1c, 0x01813030 }, /* line-in */ + { 0x1d, 0x01a19020 }, /* rear mic */ { 0x1e, 0x411111f0 }, /* N/A */ { 0x1f, 0x02a190f0 }, /* mic */ - { 0x20, 0x018130f0 }, /* line-in */ + { 0x20, 0x411111f0 }, /* N/A */ {} }, }, -- 1.8.5.2

1 1

[alsa-devel] [PATCH] ASoC: pcm: Fix lack of platform bespoke_trigger() call
by Jean-Francois Moine 07 Jan '14

07 Jan '14

When the platform driver has no ops, the platform function bespoke_trigger() is no more called. The problem was introduced by the commit c5914b0aaea6494aaa9e415cbd32f8b7eb604af0 "ASoC: pcm: Check for ops before deferencing them" Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 891b9a9..df79101 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -672,7 +672,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, return ret; } - if (platform->driver->ops && platform->driver->bespoke_trigger) { + if (platform->driver->bespoke_trigger) { ret = platform->driver->bespoke_trigger(substream, cmd); if (ret < 0) return ret; -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH] ASoC: simple-card: Do not set cpu_dai.name for snd_link in case DT
by Alexander Shiyan 07 Jan '14

07 Jan '14

Using cpu_dai.name for DT case will result driver cannot be initialized. For example, device have name "ssi@10010000" in the DT and this value is returning for us by snd_soc_of_get_dai_name() but actual registered name of device is "10010000.ssi". Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/generic/simple-card.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 1d87db8..3a054c4 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -219,7 +219,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) */ cinfo->snd_link.name = cinfo->name; cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; + if (!of_cpu) + cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; cinfo->snd_link.platform_name = cinfo->platform; cinfo->snd_link.codec_name = cinfo->codec; cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: soc-core: add snd_soc_of_parse_sysclk_id() for dt
by Xiubo Li 07 Jan '14

07 Jan '14

This patch adds snd_soc_of_parse_sysclk_id() and supports below style on dt: sysclk-id = <XXX>; It's an optional property, If this property is absent the defualt vaule 0 will be returned. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 16 ++++++++++++++++ 2 files changed, 17 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5a049d9..86ac72f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1170,6 +1170,7 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname); int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); +unsigned int snd_soc_of_parse_sysclk_id(struct device_node *np); unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix); int snd_soc_of_get_dai_name(struct device_node *of_node, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index be88df5..572d95d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4472,6 +4472,22 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_routing); +unsigned int snd_soc_of_parse_sysclk_id(struct device_node *np) +{ + u32 id; + + /* + * check "sysclk-id = <XX>" + * it is an optional property + * 0 will be returned if absent + */ + if (of_property_read_u32(np, "sysclk-id", &id)) + id = 0; + + return id; +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_sysclk_id); + unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix) { -- 1.8.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: simple-card: fix the DAPM routes map parsing
by Xiubo Li 07 Jan '14

07 Jan '14

The simple-card's DAPM route maping is optional. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Resend this patch. sound/soc/generic/simple-card.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index c0d92813..33c816c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -138,10 +138,12 @@ static int asoc_simple_card_parse_of(struct device_node *node, (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); /* DAPM routes */ - ret = snd_soc_of_parse_audio_routing(&info->snd_card, + if (of_property_read_bool(node, "simple-audio-routing")) { + ret = snd_soc_of_parse_audio_routing(&info->snd_card, "simple-audio-routing"); - if (ret) - return ret; + if (ret) + return ret; + } /* CPU sub-node */ ret = -EINVAL; -- 1.8.4

2 1

[alsa-devel] [PATCH 00/10] ASoC: ARM: ux500: Better prepare for full Device Tree
by Lee Jones 07 Jan '14

07 Jan '14

At this moment in time the Ux500 ASoC driver obtains its DMA information via AUXDATA platform data passing. To make the driver more independent we have to start extracting it from the Device Tree. This patch-set goes some way to ensure the ux500 Audio driver is ready for the switch over. arch/arm/mach-ux500/board-mop500-audio.c | 8 +++---- include/linux/platform_data/asoc-ux500-msp.h | 9 +------- sound/soc/codecs/ab8500-codec.c | 4 ++-- sound/soc/ux500/mop500.c | 2 ++ sound/soc/ux500/ux500_msp_dai.c | 146 +++++++++++++++++++++++++++++++++++++++++++++++-------------------------------------------------------------------- sound/soc/ux500/ux500_msp_i2s.c | 56 ++++++++++++++++++++++++++++++++------------ sound/soc/ux500/ux500_msp_i2s.h | 2 +- sound/soc/ux500/ux500_pcm.c | 50 ++++++++++++++++++++++++++-------------- 8 files changed, 144 insertions(+), 133 deletions(-)

4 34

[alsa-devel] ASoC: fix new build warning in fsl_ssi
by Arnd Bergmann 07 Jan '14

07 Jan '14

Commit 6873ee464a9f "ASoC: fsl_ssi: Fix printing return code on clk error" changed a warning message but got the format string wrong, resulting in a warning "sound/soc/fsl/fsl_ssi.c:1196:6: warning: format '%d' expects argument of type 'int', but argument 3 has type 'long int' [-Wformat=]". Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- Please apply or fold into the original patch, whichever works for you. diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 3d74477a..c9d567c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1192,7 +1192,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) */ ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); if (IS_ERR(ssi_private->baudclk)) - dev_warn(&pdev->dev, "could not get baud clock: %d\n", + dev_warn(&pdev->dev, "could not get baud clock: %ld\n", PTR_ERR(ssi_private->baudclk)); else clk_prepare_enable(ssi_private->baudclk);

2 1

[alsa-devel] [PATCH] ASoC: mxs: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag
by Lars-Peter Clausen 07 Jan '14

07 Jan '14

Since commit 7b11304 ("dma: mxs-dma: Report correct residue for cyclic DMA") the mxs dmaengine driver has support for residue reporting. So there is no need to specify the SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag anymore. This allows a finer grained resolution for the PCM pointer as well as avoids the race condition that can occur with the period counting that is used when the dmaengine driver does not support residue reporting. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- I don't have the hardware, so only compiled tested. But I assume that residue reporting was added to the driver for exactly this. --- sound/soc/mxs/mxs-pcm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 04a6b0d..c05c057 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -57,7 +57,6 @@ static const struct snd_dmaengine_pcm_config mxs_dmaengine_pcm_config = { int mxs_pcm_platform_register(struct device *dev) { return devm_snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX); } EXPORT_SYMBOL_GPL(mxs_pcm_platform_register); -- 1.8.0

3 2

[alsa-devel] [PATCH 0/10] ASoC: rsnd: add SRC support
by Kuninori Morimoto 07 Jan '14

07 Jan '14

Hi Mark These are cleanup and SRC support patches for Renesas sound driver #1 - #4 are cleanup #5 - #9 are prepare for SRC support #a is SRC support Kuninori Morimoto (10): 1) ASoC: rsnd: add rsnd_adg_set_ssi_clk() and cleanup adg 2) ASoC: rsnd: tidyup ssi comment 3) ASoC: rsnd: tidyup register naming 4) ASoC: rsnd: make sure variable name for 44.1kHz/48kHz 5) ASoC: rsnd: route setting is needed only Gen1 6) ASoC: rsnd: make sure Gen2 only method on ssi 7) ASoC: rsnd: add rsnd_scu_transfer_start() 8) ASoC: rsnd: scu cleanup: add rsnd_scu_rate_ctrl() 9) ASoC: rsnd: tidyup rsnd_ssi_master_clk_start() parameter a) ASoC: rsnd: add SRC (Sampling Rate Converter) support include/sound/rcar_snd.h | 1 + sound/soc/sh/rcar/adg.c | 151 +++++++++++++++++++++++++++----------- sound/soc/sh/rcar/gen.c | 14 +++- sound/soc/sh/rcar/rsnd.h | 32 ++++++-- sound/soc/sh/rcar/scu.c | 181 +++++++++++++++++++++++++++++++++++++++++----- sound/soc/sh/rcar/ssi.c | 17 ++--- 6 files changed, 314 insertions(+), 82 deletions(-) Best regards --- Kuninori Morimoto

3 16

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fixed wrong printf format identifier
by Alexander Shiyan 07 Jan '14

07 Jan '14

sound/soc/fsl/fsl_ssi.c: In function 'fsl_ssi_probe': sound/soc/fsl/fsl_ssi.c:1180:6: warning: format '%d' expects argument of type 'int', but argument 3 has type 'long int' [-Wformat=] Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 3d74477a..c9d567c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1192,7 +1192,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) */ ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); if (IS_ERR(ssi_private->baudclk)) - dev_warn(&pdev->dev, "could not get baud clock: %d\n", + dev_warn(&pdev->dev, "could not get baud clock: %ld\n", PTR_ERR(ssi_private->baudclk)); else clk_prepare_enable(ssi_private->baudclk); -- 1.8.3.2

3 2

[alsa-devel] [PATCH 1/5] ASoC: mc13783: Use module_platform_driver_probe()
by Alexander Shiyan 07 Jan '14

07 Jan '14

mc13783-codec is probed only by MC13XXX MFD core driver so use module_platform_driver_probe(). Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/codecs/mc13783.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index bae6016..8ab9668 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -750,7 +750,7 @@ static struct snd_soc_codec_driver soc_codec_dev_mc13783 = { .num_dapm_routes = ARRAY_SIZE(mc13783_routes), }; -static int mc13783_codec_probe(struct platform_device *pdev) +static int __init mc13783_codec_probe(struct platform_device *pdev) { struct mc13xxx *mc13xxx; struct mc13783_priv *priv; @@ -804,11 +804,9 @@ static struct platform_driver mc13783_codec_driver = { .name = "mc13783-codec", .owner = THIS_MODULE, }, - .probe = mc13783_codec_probe, .remove = mc13783_codec_remove, }; - -module_platform_driver(mc13783_codec_driver); +module_platform_driver_probe(mc13783_codec_driver, mc13783_codec_probe); MODULE_DESCRIPTION("ASoC MC13783 driver"); MODULE_AUTHOR("Sascha Hauer, Pengutronix <s.hauer(a)pengutronix.de>"); -- 1.8.3.2

2 13

[alsa-devel] [PATCH] ASoC: fsl: fsl_ssi: Use '%ld' to print 'long int'
by Fabio Estevam 07 Jan '14

07 Jan '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Commit 6873ee464a (ASoC: fsl_ssi: Fix printing return code on clk error) caused the following build warning: sound/soc/fsl/fsl_ssi.c: In function 'fsl_ssi_probe': sound/soc/fsl/fsl_ssi.c:1196:6: warning: format '%d' expects argument of type 'int', but argument 3 has type 'long int' [-Wformat] Fix it by using '%ld' to print the 'long int' format. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 3d74477a..c9d567c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1192,7 +1192,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) */ ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); if (IS_ERR(ssi_private->baudclk)) - dev_warn(&pdev->dev, "could not get baud clock: %d\n", + dev_warn(&pdev->dev, "could not get baud clock: %ld\n", PTR_ERR(ssi_private->baudclk)); else clk_prepare_enable(ssi_private->baudclk); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: atmel-pcm-dma: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag
by Lars-Peter Clausen 07 Jan '14

07 Jan '14

Since commit d48de6f1a ("DMA: AT91: Get residual bytes in dma buffer") the at91-dma driver has support for residue reporting. So there is no need to specify the SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag. This allows for a finer grained resolution of the PCM pointer as well as avoids the race condition that can occur with the period counting that is used when the dmaengine driver does not support residue reporting. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- I don't have the hardware, so only compile tested. It should work though if the residue calculation in the dma driver is implemented correctly. --- sound/soc/atmel/atmel-pcm-dma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index 06082e5..ba1366a 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -129,8 +129,7 @@ static const struct snd_dmaengine_pcm_config atmel_dmaengine_pcm_config = { int atmel_pcm_dma_platform_register(struct device *dev) { - return snd_dmaengine_pcm_register(dev, &atmel_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); + return snd_dmaengine_pcm_register(dev, &atmel_dmaengine_pcm_config, 0); } EXPORT_SYMBOL(atmel_pcm_dma_platform_register); -- 1.8.0

2 2

[alsa-devel] [PATCH 2/2] ASoC: simple-card: keep the property's name the same pattern
by Xiubo Li 07 Jan '14

07 Jan '14

Even though we might not have rigor rule for the simple card property names, according to the existing ones, they are all in a same pattern: [simple-audio-card,]XXX; Rename simple-audio-routing to simple-audio-card,routing, and make the simple card's properties has one unified name. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Documentation/devicetree/bindings/sound/simple-card.txt | 2 +- sound/soc/generic/simple-card.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index 2ee80c76..e9e20ec 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -11,7 +11,7 @@ Optional properties: - simple-audio-card,format : CPU/CODEC common audio format. "i2s", "right_j", "left_j" , "dsp_a" "dsp_b", "ac97", "pdm", "msb", "lsb" -- simple-audio-routing : A list of the connections between audio components. +- simple-audio-card,routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the connection's sink, the second being the connection's source. diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 33c816c..d007760 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -138,9 +138,9 @@ static int asoc_simple_card_parse_of(struct device_node *node, (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); /* DAPM routes */ - if (of_property_read_bool(node, "simple-audio-routing")) { + if (of_property_read_bool(node, "simple-audio-card,routing")) { ret = snd_soc_of_parse_audio_routing(&info->snd_card, - "simple-audio-routing"); + "simple-audio-card,routing"); if (ret) return ret; } -- 1.8.4

1 0

[alsa-devel] [PATCH] ASoC: simple-card: fix the DAPM routes map parsing
by Xiubo Li 07 Jan '14

07 Jan '14

The simple-card's DAPM route maping is optional. For example, the spdif has no need of this. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index e86ddf5..889908a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -145,10 +145,12 @@ static int asoc_simple_card_parse_of(struct device_node *node, (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); /* DAPM routes */ - ret = snd_soc_of_parse_audio_routing(&info->snd_card, + if (of_property_read_bool(node, "simple-audio-card,routing")) { + ret = snd_soc_of_parse_audio_routing(&info->snd_card, "simple-audio-card,routing"); - if (ret) - return ret; + if (ret) + return ret; + } /* CPU sub-node */ ret = -EINVAL; -- 1.8.4

3 2

[alsa-devel] [PATCH 01/19] ASoC: atmel: Don't set unused struct snd_pcm_hardware fields
by Lars-Peter Clausen 06 Jan '14

06 Jan '14

The ASoC core assumes that the PCM component of the ASoC card transparently moves data around and does not impose any restrictions on the memory layout or the transfer speed. It ignores all fields from the snd_pcm_hardware struct for the PCM driver that are related to this. Setting these fields in the PCM driver might suggest otherwise though, so rather not set them. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Bo Shen <voice.shen(a)atmel.com> Cc: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/atmel/atmel-pcm-dma.c | 1 - sound/soc/atmel/atmel-pcm-pdc.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index 06082e5..b79a2a8 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -50,7 +50,6 @@ static const struct snd_pcm_hardware atmel_pcm_dma_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_PAUSE, - .formats = SNDRV_PCM_FMTBIT_S16_LE, .period_bytes_min = 256, /* lighting DMA overhead */ .period_bytes_max = 2 * 0xffff, /* if 2 bytes format */ .periods_min = 8, diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c index 054ea4d..33ec592 100644 --- a/sound/soc/atmel/atmel-pcm-pdc.c +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -58,7 +58,6 @@ static const struct snd_pcm_hardware atmel_pcm_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE, - .formats = SNDRV_PCM_FMTBIT_S16_LE, .period_bytes_min = 32, .period_bytes_max = 8192, .periods_min = 2, -- 1.8.0

10 46

[alsa-devel] [PATCH] ASoC: simple-card: Fix the sysclk selection.
by Xiubo Li 06 Jan '14

06 Jan '14

For spdif there is no need to do the sysclk setting. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2eb8005..747334c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -113,12 +113,8 @@ asoc_simple_card_sub_parse_of(struct device_node *np, &dai->sysclk); } else { clk = of_clk_get(*node, 0); - if (IS_ERR(clk)) { - ret = PTR_ERR(clk); - goto parse_error; - } - - dai->sysclk = clk_get_rate(clk); + if (!IS_ERR(clk)) + dai->sysclk = clk_get_rate(clk); } ret = 0; -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: imx: Allow using sound with DT descriptions only
by Alexander Shiyan 06 Jan '14

06 Jan '14

This patch adds new Kconfig symbol to allow using generic sound cards/codecs based on DT description. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/Kconfig | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index bb4ab96..51fabc6 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -126,6 +126,16 @@ config SND_SOC_IMX_PCM_DMA config SND_SOC_IMX_AUDMUX tristate +config SND_SOC_IMX_DT + tristate "SoC Audio support for i.MX DT boards" + depends on OF + select SND_SOC_FSL_SSI + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_PCM_DMA + help + Say Y if you want to add support for SoC audio on an i.MX boards + described in devicetree. + config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the i.MX31ADS with WM1133-EV1 fitted" depends on MACH_MX31ADS_WM1133_EV1 -- 1.8.3.2

3 4

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix printing return code on clk error
by Alexander Shiyan 06 Jan '14

06 Jan '14

Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/fsl_ssi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2101fc5..3d74477a 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1192,7 +1192,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) */ ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); if (IS_ERR(ssi_private->baudclk)) - dev_warn(&pdev->dev, "could not get baud clock: %d\n", ret); + dev_warn(&pdev->dev, "could not get baud clock: %d\n", + PTR_ERR(ssi_private->baudclk)); else clk_prepare_enable(ssi_private->baudclk); -- 1.8.3.2

2 1

[alsa-devel] [PATCH] alsa-lib: minor typo in pcm doc
by Olivier Langlois 06 Jan '14

06 Jan '14

Added a missing 'f' in the word 'diferent' Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> --- src/pcm/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index c54fa06..7e46014 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -105,7 +105,7 @@ The poll or select functions (see 'man 2 poll' or 'man 2 select' for further details) allows to receive requests/events from the device while an application is waiting on events from other sources (like keyboard, screen, network etc.), too. \ref snd_pcm_poll_descriptors can be used to get file -descriptors to poll or select on (note that wait direction might be diferent +descriptors to poll or select on (note that wait direction might be different than expected - do not use only returned file descriptors, but handle events member as well - see \ref snd_pcm_poll_descriptors function description for more details and \ref snd_pcm_poll_descriptors_revents for -- 1.8.5.2

2 1

Re: [alsa-devel] [PATCH] ASoC: simple-card: Add device's module clock selection.
by Jean-Francois Moine 06 Jan '14

06 Jan '14

Li, I am trying to have many DAI links in the simple-card, but I get an error on getting the device clock: only the dma device needs a clock and the spdif codec has none. What was the purpose of this default device clock? -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCHv9] dmaengine: Add support for BCM2835
by Florian Meier 05 Jan '14

05 Jan '14

Add support for DMA controller of BCM2835 as used in the Raspberry Pi. Currently it only supports cyclic DMA. Signed-off-by: Florian Meier <florian.meier(a)koalo.de> Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- Andy, thank you for the review! The changes are just for the minor issues you raised. .../devicetree/bindings/dma/bcm2835-dma.txt | 57 ++ drivers/dma/Kconfig | 6 + drivers/dma/Makefile | 1 + drivers/dma/bcm2835-dma.c | 732 +++++++++++++++++++++ 4 files changed, 796 insertions(+) create mode 100644 Documentation/devicetree/bindings/dma/bcm2835-dma.txt create mode 100644 drivers/dma/bcm2835-dma.c diff --git a/Documentation/devicetree/bindings/dma/bcm2835-dma.txt b/Documentation/devicetree/bindings/dma/bcm2835-dma.txt new file mode 100644 index 0000000..e7c31c1 --- /dev/null +++ b/Documentation/devicetree/bindings/dma/bcm2835-dma.txt @@ -0,0 +1,57 @@ +* BCM2835 DMA controller + +The BCM2835 DMA controller has 16 channels in total. +Only the lower 13 channels have an associated IRQ. +Some arbitrary channels are used by the firmware +(1,3,6,7 in the current firmware version). +The channels 0,2 and 3 have special functionality +and should not be used by the driver. + +Required properties: +- compatible: Should be "brcm,bcm2835-dma". +- reg: Should contain DMA registers location and length. +- interrupts: Should contain the DMA interrupts associated + to the DMA channels in ascending order. +- #dma-cells: Must be <1>, the cell in the dmas property of the + client device represents the DREQ number. +- brcm,dma-channel-mask: Bit mask representing the channels + not used by the firmware in ascending order, + i.e. first channel corresponds to LSB. + +Example: + +dma: dma@7e007000 { + compatible = "brcm,bcm2835-dma"; + reg = <0x7e007000 0xf00>; + interrupts = <1 16> + <1 17> + <1 18> + <1 19> + <1 20> + <1 21> + <1 22> + <1 23> + <1 24> + <1 25> + <1 26> + <1 27> + <1 28>; + + #dma-cells = <1>; + brcm,dma-channel-mask = <0x7f35>; +}; + +DMA clients connected to the BCM2835 DMA controller must use the format +described in the dma.txt file, using a two-cell specifier for each channel. + +Example: + +bcm2835_i2s: i2s@7e203000 { + compatible = "brcm,bcm2835-i2s"; + reg = < 0x7e203000 0x20> + < 0x7e101098 0x02>; + + dmas = <&dma 2> + <&dma 3>; + dma-names = "tx", "rx"; +}; diff --git a/drivers/dma/Kconfig b/drivers/dma/Kconfig index c61a6ec..880e723 100644 --- a/drivers/dma/Kconfig +++ b/drivers/dma/Kconfig @@ -300,6 +300,12 @@ config DMA_OMAP select DMA_ENGINE select DMA_VIRTUAL_CHANNELS +config DMA_BCM2835 + tristate "BCM2835 DMA engine support" + depends on (ARCH_BCM2835 || MACH_BCM2708) + select DMA_ENGINE + select DMA_VIRTUAL_CHANNELS + config TI_CPPI41 tristate "AM33xx CPPI41 DMA support" depends on ARCH_OMAP diff --git a/drivers/dma/Makefile b/drivers/dma/Makefile index 0ce2da9..0a6f08e 100644 --- a/drivers/dma/Makefile +++ b/drivers/dma/Makefile @@ -38,6 +38,7 @@ obj-$(CONFIG_EP93XX_DMA) += ep93xx_dma.o obj-$(CONFIG_DMA_SA11X0) += sa11x0-dma.o obj-$(CONFIG_MMP_TDMA) += mmp_tdma.o obj-$(CONFIG_DMA_OMAP) += omap-dma.o +obj-$(CONFIG_DMA_BCM2835) += bcm2835-dma.o obj-$(CONFIG_MMP_PDMA) += mmp_pdma.o obj-$(CONFIG_DMA_JZ4740) += dma-jz4740.o obj-$(CONFIG_TI_CPPI41) += cppi41.o diff --git a/drivers/dma/bcm2835-dma.c b/drivers/dma/bcm2835-dma.c new file mode 100644 index 0000000..02a329f --- /dev/null +++ b/drivers/dma/bcm2835-dma.c @@ -0,0 +1,732 @@ +/* + * BCM2835 DMA engine support + * + * This driver only supports cyclic DMA transfers + * as needed for the I2S module. + * + * Author: Florian Meier <florian.meier(a)koalo.de> + * Copyright 2013 + * + * Based on + * OMAP DMAengine support by Russell King + * + * BCM2708 DMA Driver + * Copyright (C) 2010 Broadcom + * + * Raspberry Pi PCM I2S ALSA Driver + * Copyright (c) by Phil Poole 2013 + * + * MARVELL MMP Peripheral DMA Driver + * Copyright 2012 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ +#include <linux/dmaengine.h> +#include <linux/dma-mapping.h> +#include <linux/err.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/list.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <linux/spinlock.h> +#include <linux/of.h> +#include <linux/of_dma.h> + +#include "virt-dma.h" + +struct bcm2835_dmadev { + struct dma_device ddev; + spinlock_t lock; + void __iomem *base; + struct device_dma_parameters dma_parms; +}; + +struct bcm2835_dma_cb { + uint32_t info; + uint32_t src; + uint32_t dst; + uint32_t length; + uint32_t stride; + uint32_t next; + uint32_t pad[2]; +}; + +struct bcm2835_chan { + struct virt_dma_chan vc; + struct list_head node; + + struct dma_slave_config cfg; + bool cyclic; + unsigned int dreq; + + int ch; + struct bcm2835_desc *desc; + + void __iomem *chan_base; + int irq_number; +}; + +struct bcm2835_desc { + struct virt_dma_desc vd; + enum dma_transfer_direction dir; + + unsigned int control_block_size; + struct bcm2835_dma_cb *control_block_base; + dma_addr_t control_block_base_phys; + + unsigned int frames; + size_t size; +}; + +#define BCM2835_DMA_CS 0x00 +#define BCM2835_DMA_ADDR 0x04 +#define BCM2835_DMA_SOURCE_AD 0x0c +#define BCM2835_DMA_DEST_AD 0x10 +#define BCM2835_DMA_NEXTCB 0x1C + +/* DMA CS Control and Status bits */ +#define BCM2835_DMA_ACTIVE BIT(0) +#define BCM2835_DMA_INT BIT(2) +#define BCM2835_DMA_ISPAUSED BIT(4) /* Pause requested or not active */ +#define BCM2835_DMA_ISHELD BIT(5) /* Is held by DREQ flow control */ +#define BCM2835_DMA_ERR BIT(8) +#define BCM2835_DMA_ABORT BIT(30) /* Stop current CB, go to next, WO */ +#define BCM2835_DMA_RESET BIT(31) /* WO, self clearing */ + +#define BCM2835_DMA_INT_EN BIT(0) +#define BCM2835_DMA_D_INC BIT(4) +#define BCM2835_DMA_D_DREQ BIT(6) +#define BCM2835_DMA_S_INC BIT(8) +#define BCM2835_DMA_S_DREQ BIT(10) + +#define BCM2835_DMA_PER_MAP(x) ((x) << 16) + +#define BCM2835_DMA_DATA_TYPE_S8 1 +#define BCM2835_DMA_DATA_TYPE_S16 2 +#define BCM2835_DMA_DATA_TYPE_S32 4 +#define BCM2835_DMA_DATA_TYPE_S128 16 + +#define BCM2835_DMA_BULK_MASK BIT(0) +#define BCM2835_DMA_FIQ_MASK (BIT(2) | BIT(3)) + +/* Valid only for channels 0 - 14, 15 has its own base address */ +#define BCM2835_DMA_CHAN(n) ((n) << 8) /* Base address */ +#define BCM2835_DMA_CHANIO(base, n) ((base) + BCM2835_DMA_CHAN(n)) + +static inline struct bcm2835_dmadev *to_bcm2835_dma_dev(struct dma_device *d) +{ + return container_of(d, struct bcm2835_dmadev, ddev); +} + +static inline struct bcm2835_chan *to_bcm2835_dma_chan(struct dma_chan *c) +{ + return container_of(c, struct bcm2835_chan, vc.chan); +} + +static inline struct bcm2835_desc *to_bcm2835_dma_desc( + struct dma_async_tx_descriptor *t) +{ + return container_of(t, struct bcm2835_desc, vd.tx); +} + +static void bcm2835_dma_desc_free(struct virt_dma_desc *vd) +{ + struct bcm2835_desc *desc = container_of(vd, struct bcm2835_desc, vd); + dma_free_coherent(desc->vd.tx.chan->device->dev, + desc->control_block_size, + desc->control_block_base, + desc->control_block_base_phys); + kfree(desc); +} + +static int bcm2835_dma_abort(void __iomem *chan_base) +{ + unsigned long cs; + long int timeout = 10000; + + cs = readl(chan_base + BCM2835_DMA_CS); + if (!(cs & BCM2835_DMA_ACTIVE)) + return 0; + + /* Write 0 to the active bit - Pause the DMA */ + writel(0, chan_base + BCM2835_DMA_CS); + + /* Wait for any current AXI transfer to complete */ + while ((cs & BCM2835_DMA_ISPAUSED) && --timeout) { + cpu_relax(); + cs = readl(chan_base + BCM2835_DMA_CS); + } + + /* We'll un-pause when we set of our next DMA */ + if (!timeout) + return -ETIMEDOUT; + + if (!(cs & BCM2835_DMA_ACTIVE)) + return 0; + + /* Terminate the control block chain */ + writel(0, chan_base + BCM2835_DMA_NEXTCB); + + /* Abort the whole DMA */ + writel(BCM2835_DMA_ABORT | BCM2835_DMA_ACTIVE, + chan_base + BCM2835_DMA_CS); + + return 0; +} + +static void bcm2835_dma_start_desc(struct bcm2835_chan *c) +{ + struct virt_dma_desc *vd = vchan_next_desc(&c->vc); + struct bcm2835_desc *d; + + if (!vd) { + c->desc = NULL; + return; + } + + list_del(&vd->node); + + c->desc = d = to_bcm2835_dma_desc(&vd->tx); + + writel(d->control_block_base_phys, c->chan_base + BCM2835_DMA_ADDR); + writel(BCM2835_DMA_ACTIVE, c->chan_base + BCM2835_DMA_CS); +} + +static irqreturn_t bcm2835_dma_callback(int irq, void *data) +{ + struct bcm2835_chan *c = data; + struct bcm2835_desc *d; + unsigned long flags; + + spin_lock_irqsave(&c->vc.lock, flags); + + /* Acknowledge interrupt */ + writel(BCM2835_DMA_INT, c->chan_base + BCM2835_DMA_CS); + + d = c->desc; + + if (d) { + /* TODO Only works for cyclic DMA */ + vchan_cyclic_callback(&d->vd); + } + + /* Keep the DMA engine running */ + writel(BCM2835_DMA_ACTIVE, c->chan_base + BCM2835_DMA_CS); + + spin_unlock_irqrestore(&c->vc.lock, flags); + + return IRQ_HANDLED; +} + +static int bcm2835_dma_alloc_chan_resources(struct dma_chan *chan) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + + dev_dbg(c->vc.chan.device->dev, + "Allocating DMA channel %d\n", c->ch); + + return request_irq(c->irq_number, + bcm2835_dma_callback, 0, "DMA IRQ", c); +} + +static void bcm2835_dma_free_chan_resources(struct dma_chan *chan) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + + vchan_free_chan_resources(&c->vc); + free_irq(c->irq_number, c); + + dev_dbg(c->vc.chan.device->dev, "Freeing DMA channel %u\n", c->ch); +} + +static size_t bcm2835_dma_desc_size(struct bcm2835_desc *d) +{ + return d->size; +} + +static size_t bcm2835_dma_desc_size_pos(struct bcm2835_desc *d, dma_addr_t addr) +{ + unsigned int i; + size_t size; + + for (size = i = 0; i < d->frames; i++) { + struct bcm2835_dma_cb *control_block = + &d->control_block_base[i]; + size_t this_size = control_block->length; + dma_addr_t dma; + + if (d->dir == DMA_DEV_TO_MEM) + dma = control_block->dst; + else + dma = control_block->src; + + if (size) + size += this_size; + else if (addr >= dma && addr < dma + this_size) + size += dma + this_size - addr; + } + + return size; +} + +static enum dma_status bcm2835_dma_tx_status(struct dma_chan *chan, + dma_cookie_t cookie, struct dma_tx_state *txstate) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + struct virt_dma_desc *vd; + enum dma_status ret; + unsigned long flags; + + ret = dma_cookie_status(chan, cookie, txstate); + if (ret == DMA_SUCCESS || !txstate) + return ret; + + spin_lock_irqsave(&c->vc.lock, flags); + vd = vchan_find_desc(&c->vc, cookie); + if (vd) { + txstate->residue = + bcm2835_dma_desc_size(to_bcm2835_dma_desc(&vd->tx)); + } else if (c->desc && c->desc->vd.tx.cookie == cookie) { + struct bcm2835_desc *d = c->desc; + dma_addr_t pos; + + if (d->dir == DMA_MEM_TO_DEV) + pos = readl(c->chan_base + BCM2835_DMA_SOURCE_AD); + else if (d->dir == DMA_DEV_TO_MEM) + pos = readl(c->chan_base + BCM2835_DMA_DEST_AD); + else + pos = 0; + + txstate->residue = bcm2835_dma_desc_size_pos(d, pos); + } else { + txstate->residue = 0; + } + + spin_unlock_irqrestore(&c->vc.lock, flags); + + return ret; +} + +static void bcm2835_dma_issue_pending(struct dma_chan *chan) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + unsigned long flags; + + c->cyclic = true; /* Nothing else is implemented */ + + spin_lock_irqsave(&c->vc.lock, flags); + if (vchan_issue_pending(&c->vc) && !c->desc) + bcm2835_dma_start_desc(c); + + spin_unlock_irqrestore(&c->vc.lock, flags); +} + +static struct dma_async_tx_descriptor *bcm2835_dma_prep_dma_cyclic( + struct dma_chan *chan, dma_addr_t buf_addr, size_t buf_len, + size_t period_len, enum dma_transfer_direction direction, + unsigned long flags, void *context) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + enum dma_slave_buswidth dev_width; + struct bcm2835_desc *d; + dma_addr_t dev_addr; + unsigned int es, sync_type; + unsigned int frame; + + /* Grab configuration */ + if (!is_slave_direction(direction)) { + dev_err(chan->device->dev, "%s: bad direction?\n", __func__); + return NULL; + } + + if (direction == DMA_DEV_TO_MEM) { + dev_addr = c->cfg.src_addr; + dev_width = c->cfg.src_addr_width; + sync_type = BCM2835_DMA_S_DREQ; + } else { + dev_addr = c->cfg.dst_addr; + dev_width = c->cfg.dst_addr_width; + sync_type = BCM2835_DMA_D_DREQ; + } + + /* Bus width translates to the element size (ES) */ + switch (dev_width) { + case DMA_SLAVE_BUSWIDTH_4_BYTES: + es = BCM2835_DMA_DATA_TYPE_S32; + break; + default: + return NULL; + } + + /* Now allocate and setup the descriptor. */ + d = kzalloc(sizeof(*d), GFP_NOWAIT); + if (!d) + return NULL; + + d->dir = direction; + d->frames = buf_len / period_len; + + /* Allocate memory for control blocks */ + d->control_block_size = d->frames * sizeof(struct bcm2835_dma_cb); + d->control_block_base = dma_zalloc_coherent(chan->device->dev, + d->control_block_size, &d->control_block_base_phys, + GFP_NOWAIT); + + if (!d->control_block_base) { + kfree(d); + return NULL; + } + + /* + * Iterate over all frames, create a control block + * for each frame and link them together. + */ + for (frame = 0; frame < d->frames; frame++) { + struct bcm2835_dma_cb *control_block = + &d->control_block_base[frame]; + + /* Setup adresses */ + if (d->dir == DMA_DEV_TO_MEM) { + control_block->info = BCM2835_DMA_D_INC; + control_block->src = dev_addr; + control_block->dst = buf_addr + frame * period_len; + } else { + control_block->info = BCM2835_DMA_S_INC; + control_block->src = buf_addr + frame * period_len; + control_block->dst = dev_addr; + } + + /* Enable interrupt */ + control_block->info |= BCM2835_DMA_INT_EN; + + /* Setup synchronization */ + if (sync_type != 0) + control_block->info |= sync_type; + + /* Setup DREQ channel */ + if (c->dreq != 0) + control_block->info |= + BCM2835_DMA_PER_MAP(c->dreq); + + /* Length of a frame */ + control_block->length = period_len; + d->size += control_block->length; + + /* + * Next block is the next frame. + * This DMA engine driver currently only supports cyclic DMA. + * Therefore, wrap around at number of frames. + */ + control_block->next = d->control_block_base_phys + + sizeof(struct bcm2835_dma_cb) + * ((frame + 1) % d->frames); + } + + return vchan_tx_prep(&c->vc, &d->vd, flags); +} + +static int bcm2835_dma_slave_config(struct bcm2835_chan *c, + struct dma_slave_config *cfg) +{ + if ((cfg->direction == DMA_DEV_TO_MEM && + cfg->src_addr_width != DMA_SLAVE_BUSWIDTH_4_BYTES) || + (cfg->direction == DMA_MEM_TO_DEV && + cfg->dst_addr_width != DMA_SLAVE_BUSWIDTH_4_BYTES) || + !is_slave_direction(cfg->direction)) { + return -EINVAL; + } + + c->cfg = *cfg; + + return 0; +} + +static int bcm2835_dma_terminate_all(struct bcm2835_chan *c) +{ + struct bcm2835_dmadev *d = to_bcm2835_dma_dev(c->vc.chan.device); + unsigned long flags; + int timeout = 10000; + LIST_HEAD(head); + + spin_lock_irqsave(&c->vc.lock, flags); + + /* Prevent this channel being scheduled */ + spin_lock(&d->lock); + list_del_init(&c->node); + spin_unlock(&d->lock); + + /* + * Stop DMA activity: we assume the callback will not be called + * after bcm_dma_abort() returns (even if it does, it will see + * c->desc is NULL and exit.) + */ + if (c->desc) { + c->desc = NULL; + bcm2835_dma_abort(c->chan_base); + + /* Wait for stopping */ + while (--timeout) { + if (!(readl(c->chan_base + BCM2835_DMA_CS) & + BCM2835_DMA_ACTIVE)) + break; + + cpu_relax(); + } + + if (!timeout) + dev_err(d->ddev.dev, "DMA transfer could not be terminated\n"); + } + + vchan_get_all_descriptors(&c->vc, &head); + spin_unlock_irqrestore(&c->vc.lock, flags); + vchan_dma_desc_free_list(&c->vc, &head); + + return 0; +} + +static int bcm2835_dma_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd, + unsigned long arg) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + + switch (cmd) { + case DMA_SLAVE_CONFIG: + return bcm2835_dma_slave_config(c, + (struct dma_slave_config *)arg); + + case DMA_TERMINATE_ALL: + return bcm2835_dma_terminate_all(c); + + default: + return -ENXIO; + } +} + +static int bcm2835_dma_chan_init(struct bcm2835_dmadev *d, int chan_id, int irq) +{ + struct bcm2835_chan *c; + + c = devm_kzalloc(d->ddev.dev, sizeof(*c), GFP_KERNEL); + if (!c) + return -ENOMEM; + + c->vc.desc_free = bcm2835_dma_desc_free; + vchan_init(&c->vc, &d->ddev); + INIT_LIST_HEAD(&c->node); + + d->ddev.chancnt++; + + c->chan_base = BCM2835_DMA_CHANIO(d->base, chan_id); + c->ch = chan_id; + c->irq_number = irq; + + return 0; +} + +static void bcm2835_dma_free(struct bcm2835_dmadev *od) +{ + struct bcm2835_chan *c, *next; + + list_for_each_entry_safe(c, next, &od->ddev.channels, + vc.chan.device_node) { + list_del(&c->vc.chan.device_node); + tasklet_kill(&c->vc.task); + } +} + +#if defined(CONFIG_OF) +static const struct of_device_id bcm2835_dma_of_match[] = { + { .compatible = "brcm,bcm2835-dma", }, + {}, +}; +MODULE_DEVICE_TABLE(of, bcm2835_dma_of_match); +#endif + +static struct dma_chan *bcm2835_dma_xlate(struct of_phandle_args *spec, + struct of_dma *ofdma) +{ + struct bcm2835_dmadev *d = ofdma->of_dma_data; + struct dma_chan *chan, *candidate; + +retry: + candidate = NULL; + + /* + * Walk the list of channels registered with the current instance and + * find one that is currently unused. + */ + list_for_each_entry(chan, &d->ddev.channels, device_node) + if (chan->client_count == 0) { + candidate = chan; + break; + } + + if (!candidate) + return NULL; + + /* + * dma_get_slave_channel will return NULL if we lost a race between + * the lookup and the reservation. + */ + chan = dma_get_slave_channel(candidate); + + if (!chan) + goto retry; + + /* Set DREQ from param */ + to_bcm2835_dma_chan(chan)->dreq = spec->args[0]; + + return chan; +} + +static int bcm2835_dma_device_slave_caps(struct dma_chan *dchan, + struct dma_slave_caps *caps) +{ + caps->src_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + caps->dstn_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); + caps->cmd_pause = false; + caps->cmd_terminate = true; + + return 0; +} + +static int bcm2835_dma_probe(struct platform_device *pdev) +{ + struct bcm2835_dmadev *od; + struct resource *res; + void __iomem *base; + int rc; + int i; + int irq; + uint32_t chans_available; + + if (!pdev->dev.dma_mask) + pdev->dev.dma_mask = &pdev->dev.coherent_dma_mask; + + rc = dma_set_mask(&pdev->dev, DMA_BIT_MASK(32)); + if (rc) + return rc; + dma_set_coherent_mask(&pdev->dev, DMA_BIT_MASK(32)); + + od = devm_kzalloc(&pdev->dev, sizeof(*od), GFP_KERNEL); + if (!od) + return -ENOMEM; + + pdev->dev.dma_parms = &od->dma_parms; + dma_set_max_seg_size(&pdev->dev, 0x3FFFFFFF); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); + + od->base = base; + + dma_cap_set(DMA_SLAVE, od->ddev.cap_mask); + dma_cap_set(DMA_CYCLIC, od->ddev.cap_mask); + od->ddev.device_alloc_chan_resources = bcm2835_dma_alloc_chan_resources; + od->ddev.device_free_chan_resources = bcm2835_dma_free_chan_resources; + od->ddev.device_tx_status = bcm2835_dma_tx_status; + od->ddev.device_issue_pending = bcm2835_dma_issue_pending; + od->ddev.device_slave_caps = bcm2835_dma_device_slave_caps; + od->ddev.device_prep_dma_cyclic = bcm2835_dma_prep_dma_cyclic; + od->ddev.device_control = bcm2835_dma_control; + od->ddev.dev = &pdev->dev; + INIT_LIST_HEAD(&od->ddev.channels); + spin_lock_init(&od->lock); + + platform_set_drvdata(pdev, od); + + /* Request DMA channel mask from device tree */ + if (of_property_read_u32(pdev->dev.of_node, + "brcm,dma-channel-mask", + &chans_available)) { + dev_err(&pdev->dev, "Failed to get channel mask\n"); + rc = -EINVAL; + goto err_no_dma; + } + + /* + * Do not use the FIQ and BULK channels, + * because they are used by the GPU. + */ + chans_available &= ~(BCM2835_DMA_FIQ_MASK | BCM2835_DMA_BULK_MASK); + + for (i = 0; i < pdev->num_resources; i++) { + irq = platform_get_irq(pdev, i); + if (irq < 0) + break; + + if (chans_available & (1 << i)) { + rc = bcm2835_dma_chan_init(od, i, irq); + if (rc) + goto err_no_dma; + } + } + + dev_dbg(&pdev->dev, "Initialized %i DMA channels\n", i); + + if (pdev->dev.of_node) { + /* Device-tree DMA controller registration */ + rc = of_dma_controller_register(pdev->dev.of_node, + bcm2835_dma_xlate, od); + if (rc) { + dev_err(&pdev->dev, "Failed to register DMA controller\n"); + goto err_no_dma; + } + } + + rc = dma_async_device_register(&od->ddev); + if (rc) { + dev_err(&pdev->dev, + "Failed to register slave DMA engine device: %d\n", rc); + goto err_no_dma; + } + + dev_dbg(&pdev->dev, "Load BCM2835 DMA engine driver\n"); + + return 0; + +err_no_dma: + bcm2835_dma_free(od); + return rc; +} + +static int bcm2835_dma_remove(struct platform_device *pdev) +{ + struct bcm2835_dmadev *od = platform_get_drvdata(pdev); + + dma_async_device_unregister(&od->ddev); + bcm2835_dma_free(od); + + return 0; +} + +static struct platform_driver bcm2835_dma_driver = { + .probe = bcm2835_dma_probe, + .remove = bcm2835_dma_remove, + .driver = { + .name = "bcm2835-dma", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(bcm2835_dma_of_match), + }, +}; + +module_platform_driver(bcm2835_dma_driver); + +MODULE_ALIAS("platform:bcm2835-dma"); +MODULE_DESCRIPTION("BCM2835 DMA engine driver"); +MODULE_AUTHOR("Florian Meier <florian.meier(a)koalo.de>"); +MODULE_LICENSE("GPL v2"); -- 1.8.3.2

2 6

[alsa-devel] [PATCH 1/2] usbtv: split core and video implementation
by Federico Simoncelli 05 Jan '14

05 Jan '14

Signed-off-by: Federico Simoncelli <fsimonce(a)redhat.com> Reviewed-by: Lubomir Rintel <lkundrak(a)v3.sk> --- Makefile | 3 + usbtv-core.c | 136 +++++++++++++++++++++++++++++++++++++++ usbtv.c => usbtv-video.c | 163 +++-------------------------------------------- usbtv.h | 98 ++++++++++++++++++++++++++++ 4 files changed, 246 insertions(+), 154 deletions(-) create mode 100644 usbtv-core.c rename usbtv.c => usbtv-video.c (82%) create mode 100644 usbtv.h diff --git a/Makefile b/Makefile index 28b872f..775316a 100644 --- a/Makefile +++ b/Makefile @@ -1 +1,4 @@ +usbtv-y := usbtv-core.o \ + usbtv-video.o + obj-$(CONFIG_VIDEO_USBTV) += usbtv.o diff --git a/usbtv-core.c b/usbtv-core.c new file mode 100644 index 0000000..e89e48b --- /dev/null +++ b/usbtv-core.c @@ -0,0 +1,136 @@ +/* + * Fushicai USBTV007 Video Grabber Driver + * + * Product web site: + * http://www.fushicai.com/products_detail/&productId=d05449ee-b690-42f9-a661-… + * + * Following LWN articles were very useful in construction of this driver: + * Video4Linux2 API series: http://lwn.net/Articles/203924/ + * videobuf2 API explanation: http://lwn.net/Articles/447435/ + * Thanks go to Jonathan Corbet for providing this quality documentation. + * He is awesome. + * + * Copyright (c) 2013 Lubomir Rintel + * All rights reserved. + * No physical hardware was harmed running Windows during the + * reverse-engineering activity + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + * 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions, and the following disclaimer, + * without modification. + * 2. The name of the author may not be used to endorse or promote products + * derived from this software without specific prior written permission. + * + * Alternatively, this software may be distributed under the terms of the + * GNU General Public License ("GPL"). + */ + +#include <linux/module.h> + +#include "usbtv.h" + +int usbtv_set_regs(struct usbtv *usbtv, const u16 regs[][2], int size) +{ + int ret; + int pipe = usb_rcvctrlpipe(usbtv->udev, 0); + int i; + + for (i = 0; i < size; i++) { + u16 index = regs[i][0]; + u16 value = regs[i][1]; + + ret = usb_control_msg(usbtv->udev, pipe, USBTV_REQUEST_REG, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + value, index, NULL, 0, 0); + if (ret < 0) + return ret; + } + + return 0; +} + +static int usbtv_probe(struct usb_interface *intf, + const struct usb_device_id *id) +{ + int ret; + int size; + struct device *dev = &intf->dev; + struct usbtv *usbtv; + + /* Checks that the device is what we think it is. */ + if (intf->num_altsetting != 2) + return -ENODEV; + if (intf->altsetting[1].desc.bNumEndpoints != 4) + return -ENODEV; + + /* Packet size is split into 11 bits of base size and count of + * extra multiplies of it.*/ + size = usb_endpoint_maxp(&intf->altsetting[1].endpoint[0].desc); + size = (size & 0x07ff) * (((size & 0x1800) >> 11) + 1); + + /* Device structure */ + usbtv = kzalloc(sizeof(struct usbtv), GFP_KERNEL); + if (usbtv == NULL) + return -ENOMEM; + usbtv->dev = dev; + usbtv->udev = usb_get_dev(interface_to_usbdev(intf)); + + usbtv->iso_size = size; + + usb_set_intfdata(intf, usbtv); + + ret = usbtv_video_init(usbtv); + if (ret < 0) + goto usbtv_video_fail; + + /* for simplicity we exploit the v4l2_device reference counting */ + v4l2_device_get(&usbtv->v4l2_dev); + + dev_info(dev, "Fushicai USBTV007 Video Grabber\n"); + return 0; + +usbtv_video_fail: + kfree(usbtv); + + return ret; +} + +static void usbtv_disconnect(struct usb_interface *intf) +{ + struct usbtv *usbtv = usb_get_intfdata(intf); + usb_set_intfdata(intf, NULL); + + if (!usbtv) + return; + + usbtv_video_free(usbtv); + + usb_put_dev(usbtv->udev); + usbtv->udev = NULL; + + /* the usbtv structure will be deallocated when v4l2 will be + done using it */ + v4l2_device_put(&usbtv->v4l2_dev); +} + +struct usb_device_id usbtv_id_table[] = { + { USB_DEVICE(0x1b71, 0x3002) }, + {} +}; +MODULE_DEVICE_TABLE(usb, usbtv_id_table); + +MODULE_AUTHOR("Lubomir Rintel"); +MODULE_DESCRIPTION("Fushicai USBTV007 Video Grabber Driver"); +MODULE_LICENSE("Dual BSD/GPL"); + +struct usb_driver usbtv_usb_driver = { + .name = "usbtv", + .id_table = usbtv_id_table, + .probe = usbtv_probe, + .disconnect = usbtv_disconnect, +}; + +module_usb_driver(usbtv_usb_driver); diff --git a/usbtv.c b/usbtv-video.c similarity index 82% rename from usbtv.c rename to usbtv-video.c index 6222a4a..496bc2e 100644 --- a/usbtv.c +++ b/usbtv-video.c @@ -28,45 +28,10 @@ * GNU General Public License ("GPL"). */ -#include <linux/init.h> -#include <linux/list.h> -#include <linux/module.h> -#include <linux/slab.h> -#include <linux/usb.h> -#include <linux/videodev2.h> - -#include <media/v4l2-device.h> #include <media/v4l2-ioctl.h> #include <media/videobuf2-core.h> -#include <media/videobuf2-vmalloc.h> - -/* Hardware. */ -#define USBTV_VIDEO_ENDP 0x81 -#define USBTV_BASE 0xc000 -#define USBTV_REQUEST_REG 12 - -/* Number of concurrent isochronous urbs submitted. - * Higher numbers was seen to overly saturate the USB bus. */ -#define USBTV_ISOC_TRANSFERS 16 -#define USBTV_ISOC_PACKETS 8 - -#define USBTV_CHUNK_SIZE 256 -#define USBTV_CHUNK 240 - -/* Chunk header. */ -#define USBTV_MAGIC_OK(chunk) ((be32_to_cpu(chunk[0]) & 0xff000000) \ - == 0x88000000) -#define USBTV_FRAME_ID(chunk) ((be32_to_cpu(chunk[0]) & 0x00ff0000) >> 16) -#define USBTV_ODD(chunk) ((be32_to_cpu(chunk[0]) & 0x0000f000) >> 15) -#define USBTV_CHUNK_NO(chunk) (be32_to_cpu(chunk[0]) & 0x00000fff) - -#define USBTV_TV_STD (V4L2_STD_525_60 | V4L2_STD_PAL) - -/* parameters for supported TV norms */ -struct usbtv_norm_params { - v4l2_std_id norm; - int cap_width, cap_height; -}; + +#include "usbtv.h" static struct usbtv_norm_params norm_params[] = { { @@ -81,43 +46,6 @@ static struct usbtv_norm_params norm_params[] = { } }; -/* A single videobuf2 frame buffer. */ -struct usbtv_buf { - struct vb2_buffer vb; - struct list_head list; -}; - -/* Per-device structure. */ -struct usbtv { - struct device *dev; - struct usb_device *udev; - struct v4l2_device v4l2_dev; - struct video_device vdev; - struct vb2_queue vb2q; - struct mutex v4l2_lock; - struct mutex vb2q_lock; - - /* List of videobuf2 buffers protected by a lock. */ - spinlock_t buflock; - struct list_head bufs; - - /* Number of currently processed frame, useful find - * out when a new one begins. */ - u32 frame_id; - int chunks_done; - - enum { - USBTV_COMPOSITE_INPUT, - USBTV_SVIDEO_INPUT, - } input; - v4l2_std_id norm; - int width, height; - int n_chunks; - int iso_size; - unsigned int sequence; - struct urb *isoc_urbs[USBTV_ISOC_TRANSFERS]; -}; - static int usbtv_configure_for_norm(struct usbtv *usbtv, v4l2_std_id norm) { int i, ret = 0; @@ -142,26 +70,6 @@ static int usbtv_configure_for_norm(struct usbtv *usbtv, v4l2_std_id norm) return ret; } -static int usbtv_set_regs(struct usbtv *usbtv, const u16 regs[][2], int size) -{ - int ret; - int pipe = usb_rcvctrlpipe(usbtv->udev, 0); - int i; - - for (i = 0; i < size; i++) { - u16 index = regs[i][0]; - u16 value = regs[i][1]; - - ret = usb_control_msg(usbtv->udev, pipe, USBTV_REQUEST_REG, - USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, - value, index, NULL, 0, 0); - if (ret < 0) - return ret; - } - - return 0; -} - static int usbtv_select_input(struct usbtv *usbtv, int input) { int ret; @@ -560,12 +468,6 @@ start_fail: return ret; } -struct usb_device_id usbtv_id_table[] = { - { USB_DEVICE(0x1b71, 0x3002) }, - {} -}; -MODULE_DEVICE_TABLE(usb, usbtv_id_table); - static int usbtv_querycap(struct file *file, void *priv, struct v4l2_capability *cap) { @@ -759,33 +661,9 @@ static void usbtv_release(struct v4l2_device *v4l2_dev) kfree(usbtv); } -static int usbtv_probe(struct usb_interface *intf, - const struct usb_device_id *id) +int usbtv_video_init(struct usbtv *usbtv) { int ret; - int size; - struct device *dev = &intf->dev; - struct usbtv *usbtv; - - /* Checks that the device is what we think it is. */ - if (intf->num_altsetting != 2) - return -ENODEV; - if (intf->altsetting[1].desc.bNumEndpoints != 4) - return -ENODEV; - - /* Packet size is split into 11 bits of base size and count of - * extra multiplies of it.*/ - size = usb_endpoint_maxp(&intf->altsetting[1].endpoint[0].desc); - size = (size & 0x07ff) * (((size & 0x1800) >> 11) + 1); - - /* Device structure */ - usbtv = kzalloc(sizeof(struct usbtv), GFP_KERNEL); - if (usbtv == NULL) - return -ENOMEM; - usbtv->dev = dev; - usbtv->udev = usb_get_dev(interface_to_usbdev(intf)); - - usbtv->iso_size = size; (void)usbtv_configure_for_norm(usbtv, V4L2_STD_525_60); @@ -805,20 +683,18 @@ static int usbtv_probe(struct usb_interface *intf, usbtv->vb2q.lock = &usbtv->vb2q_lock; ret = vb2_queue_init(&usbtv->vb2q); if (ret < 0) { - dev_warn(dev, "Could not initialize videobuf2 queue\n"); - goto usbtv_fail; + dev_warn(usbtv->dev, "Could not initialize videobuf2 queue\n"); + return ret; } /* v4l2 structure */ usbtv->v4l2_dev.release = usbtv_release; - ret = v4l2_device_register(dev, &usbtv->v4l2_dev); + ret = v4l2_device_register(usbtv->dev, &usbtv->v4l2_dev); if (ret < 0) { - dev_warn(dev, "Could not register v4l2 device\n"); + dev_warn(usbtv->dev, "Could not register v4l2 device\n"); goto v4l2_fail; } - usb_set_intfdata(intf, usbtv); - /* Video structure */ strlcpy(usbtv->vdev.name, "usbtv", sizeof(usbtv->vdev.name)); usbtv->vdev.v4l2_dev = &usbtv->v4l2_dev; @@ -832,52 +708,31 @@ static int usbtv_probe(struct usb_interface *intf, video_set_drvdata(&usbtv->vdev, usbtv); ret = video_register_device(&usbtv->vdev, VFL_TYPE_GRABBER, -1); if (ret < 0) { - dev_warn(dev, "Could not register video device\n"); + dev_warn(usbtv->dev, "Could not register video device\n"); goto vdev_fail; } - dev_info(dev, "Fushicai USBTV007 Video Grabber\n"); return 0; vdev_fail: v4l2_device_unregister(&usbtv->v4l2_dev); v4l2_fail: vb2_queue_release(&usbtv->vb2q); -usbtv_fail: - kfree(usbtv); return ret; } -static void usbtv_disconnect(struct usb_interface *intf) +void usbtv_video_free(struct usbtv *usbtv) { - struct usbtv *usbtv = usb_get_intfdata(intf); - mutex_lock(&usbtv->vb2q_lock); mutex_lock(&usbtv->v4l2_lock); usbtv_stop(usbtv); - usb_set_intfdata(intf, NULL); video_unregister_device(&usbtv->vdev); v4l2_device_disconnect(&usbtv->v4l2_dev); - usb_put_dev(usbtv->udev); - usbtv->udev = NULL; mutex_unlock(&usbtv->v4l2_lock); mutex_unlock(&usbtv->vb2q_lock); v4l2_device_put(&usbtv->v4l2_dev); } - -MODULE_AUTHOR("Lubomir Rintel"); -MODULE_DESCRIPTION("Fushicai USBTV007 Video Grabber Driver"); -MODULE_LICENSE("Dual BSD/GPL"); - -struct usb_driver usbtv_usb_driver = { - .name = "usbtv", - .id_table = usbtv_id_table, - .probe = usbtv_probe, - .disconnect = usbtv_disconnect, -}; - -module_usb_driver(usbtv_usb_driver); diff --git a/usbtv.h b/usbtv.h new file mode 100644 index 0000000..536343d --- /dev/null +++ b/usbtv.h @@ -0,0 +1,98 @@ +/* + * Fushicai USBTV007 Video Grabber Driver + * + * Copyright (c) 2013 Lubomir Rintel + * All rights reserved. + * No physical hardware was harmed running Windows during the + * reverse-engineering activity + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + * 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions, and the following disclaimer, + * without modification. + * 2. The name of the author may not be used to endorse or promote products + * derived from this software without specific prior written permission. + * + * Alternatively, this software may be distributed under the terms of the + * GNU General Public License ("GPL"). + */ + +#include <linux/slab.h> +#include <linux/usb.h> + +#include <media/v4l2-device.h> +#include <media/videobuf2-vmalloc.h> + +/* Hardware. */ +#define USBTV_VIDEO_ENDP 0x81 +#define USBTV_BASE 0xc000 +#define USBTV_REQUEST_REG 12 + +/* Number of concurrent isochronous urbs submitted. + * Higher numbers was seen to overly saturate the USB bus. */ +#define USBTV_ISOC_TRANSFERS 16 +#define USBTV_ISOC_PACKETS 8 + +#define USBTV_CHUNK_SIZE 256 +#define USBTV_CHUNK 240 + +/* Chunk header. */ +#define USBTV_MAGIC_OK(chunk) ((be32_to_cpu(chunk[0]) & 0xff000000) \ + == 0x88000000) +#define USBTV_FRAME_ID(chunk) ((be32_to_cpu(chunk[0]) & 0x00ff0000) >> 16) +#define USBTV_ODD(chunk) ((be32_to_cpu(chunk[0]) & 0x0000f000) >> 15) +#define USBTV_CHUNK_NO(chunk) (be32_to_cpu(chunk[0]) & 0x00000fff) + +#define USBTV_TV_STD (V4L2_STD_525_60 | V4L2_STD_PAL) + +/* parameters for supported TV norms */ +struct usbtv_norm_params { + v4l2_std_id norm; + int cap_width, cap_height; +}; + +/* A single videobuf2 frame buffer. */ +struct usbtv_buf { + struct vb2_buffer vb; + struct list_head list; +}; + +/* Per-device structure. */ +struct usbtv { + struct device *dev; + struct usb_device *udev; + + /* video */ + struct v4l2_device v4l2_dev; + struct video_device vdev; + struct vb2_queue vb2q; + struct mutex v4l2_lock; + struct mutex vb2q_lock; + + /* List of videobuf2 buffers protected by a lock. */ + spinlock_t buflock; + struct list_head bufs; + + /* Number of currently processed frame, useful find + * out when a new one begins. */ + u32 frame_id; + int chunks_done; + + enum { + USBTV_COMPOSITE_INPUT, + USBTV_SVIDEO_INPUT, + } input; + v4l2_std_id norm; + int width, height; + int n_chunks; + int iso_size; + unsigned int sequence; + struct urb *isoc_urbs[USBTV_ISOC_TRANSFERS]; +}; + +int usbtv_set_regs(struct usbtv *usbtv, const u16 regs[][2], int size); + +int usbtv_video_init(struct usbtv *usbtv); +void usbtv_video_free(struct usbtv *usbtv); -- 1.8.4.2

2 3

[alsa-devel] [RFC] ASoC: Adding trivial codec drivers
by Florian Meier 05 Jan '14

05 Jan '14

This is not intended to be a real patch. I would just like to ask if this is the preferred way to add codec and machine driver for codecs without I2C or SPI configuration possibilities or should I rather use a more generic approach? Grettings, Florian sound/soc/bcm/Kconfig | 7 +++ sound/soc/bcm/Makefile | 2 + sound/soc/bcm/rpi-dac.c | 97 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm1794a.c | 62 +++++++++++++++++++++++++ 6 files changed, 174 insertions(+) create mode 100644 sound/soc/bcm/rpi-dac.c create mode 100644 sound/soc/codecs/pcm1794a.c diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig index 8b14e37..1c1f5cf 100644 --- a/sound/soc/bcm/Kconfig +++ b/sound/soc/bcm/Kconfig @@ -15,3 +15,10 @@ +config SND_BCM2708_SOC_RPI_DAC + tristate "Support for RPi-DAC" + depends on SND_BCM2708_SOC_I2S + select SND_SOC_PCM1794A + help + Say Y or M if you want to add support for RPi-DAC. \ No newline at end of file diff --git a/sound/soc/bcm/Makefile b/sound/soc/bcm/Makefile index 6834b4d..95a9d44 100644 --- a/sound/soc/bcm/Makefile +++ b/sound/soc/bcm/Makefile @@ -5,5 +5,7 @@ obj-$(CONFIG_SND_BCM2708_SOC_I2S) += snd-soc-bcm2708-i2s.o +snd-soc-rpi-dac-objs := rpi-dac.o +obj-$(CONFIG_SND_BCM2708_SOC_RPI_DAC) += snd-soc-rpi-dac.o diff --git a/sound/soc/bcm/rpi-dac.c b/sound/soc/bcm/rpi-dac.c new file mode 100644 index 0000000..ef3cd93 --- /dev/null +++ b/sound/soc/bcm/rpi-dac.c @@ -0,0 +1,97 @@ +/* + * ASoC Driver for RPi-DAC. + * + * Author: Florian Meier <florian.meier(a)koalo.de> + * Copyright 2013 + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> + +static int snd_rpi_rpi_dac_init(struct snd_soc_pcm_runtime *rtd) +{ + return 0; +} + +static int snd_rpi_rpi_dac_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + + return snd_soc_dai_set_bclk_ratio(cpu_dai, 32*2); +} + +/* machine stream operations */ +static struct snd_soc_ops snd_rpi_rpi_dac_ops = { + .hw_params = snd_rpi_rpi_dac_hw_params, +}; + +static struct snd_soc_dai_link snd_rpi_rpi_dac_dai[] = { +{ + .name = "RPi-DAC", + .stream_name = "RPI-DAC HiFi", + .cpu_dai_name = "bcm2708-i2s.0", + .codec_dai_name = "pcm1794a-hifi", + .platform_name = "bcm2708-i2s.0", + .codec_name = "pcm1794a-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ops = &snd_rpi_rpi_dac_ops, + .init = snd_rpi_rpi_dac_init, +}, +}; + +/* audio machine driver */ +static struct snd_soc_card snd_rpi_rpi_dac = { + .name = "snd_rpi_rpi_dac", + .dai_link = snd_rpi_rpi_dac_dai, + .num_links = ARRAY_SIZE(snd_rpi_rpi_dac_dai), +}; + +static int snd_rpi_rpi_dac_probe(struct platform_device *pdev) +{ + int ret = 0; + + snd_rpi_rpi_dac.dev = &pdev->dev; + ret = snd_soc_register_card(&snd_rpi_rpi_dac); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); + + return ret; +} + +static int snd_rpi_rpi_dac_remove(struct platform_device *pdev) +{ + return snd_soc_unregister_card(&snd_rpi_rpi_dac); +} + +static struct platform_driver snd_rpi_rpi_dac_driver = { + .driver = { + .name = "snd-rpi-dac", + .owner = THIS_MODULE, + }, + .probe = snd_rpi_rpi_dac_probe, + .remove = snd_rpi_rpi_dac_remove, +}; + +module_platform_driver(snd_rpi_rpi_dac_driver); + +MODULE_AUTHOR("Florian Meier <florian.meier(a)koalo.de>"); +MODULE_DESCRIPTION("ASoC Driver for RPi-DAC"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7c43413..ef9ac1e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -55,6 +55,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_OMAP_HDMI_CODEC if OMAP4_DSS_HDMI select SND_SOC_PCM3008 + select SND_SOC_PCM1794A select SND_SOC_PCM5102A select SND_SOC_RT5631 if I2C select SND_SOC_SGTL5000 if I2C @@ -294,6 +295,9 @@ config SND_SOC_OMAP_HDMI_CODEC config SND_SOC_PCM3008 tristate +config SND_SOC_PCM1794A + tristate + config SND_SOC_PCM5102A tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 0677eb5..e98b865 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -43,6 +43,7 @@ snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-omap-hdmi-codec-objs := omap-hdmi.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-pcm1794a-objs := pcm1794a.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -171,6 +172,7 @@ obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_OMAP_HDMI_CODEC) += snd-soc-omap-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_PCM1794A) += snd-soc-pcm1794a.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/pcm1794a.c b/sound/soc/codecs/pcm1794a.c new file mode 100644 index 0000000..b4eaa44 --- /dev/null +++ b/sound/soc/codecs/pcm1794a.c @@ -0,0 +1,62 @@ +/* + * Driver for the PCM1794A codec + * + * Author: Florian Meier <florian.meier(a)koalo.de> + * Copyright 2013 + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> + +#include <sound/soc.h> + +static struct snd_soc_dai_driver pcm1794a_dai = { + .name = "pcm1794a-hifi", + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm1794a; + +static int pcm1794a_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pcm1794a, + &pcm1794a_dai, 1); +} + +static int pcm1794a_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver pcm1794a_codec_driver = { + .probe = pcm1794a_probe, + .remove = pcm1794a_remove, + .driver = { + .name = "pcm1794a-codec", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(pcm1794a_codec_driver); + +MODULE_DESCRIPTION("ASoC PCM1794A codec driver"); +MODULE_AUTHOR("Florian Meier <florian.meier(a)koalo.de>"); +MODULE_LICENSE("GPL v2"); -- 1.8.3.2

2 1

[alsa-devel] Question on compress offload framework memory corruption
by gsantosh＠codeaurora.org 05 Jan '14

05 Jan '14

Hi, I have following questions in the compressed offload framework API. static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, struct snd_compr_params *params) { ... hw_params = kzalloc(sizeof(*hw_params), GFP_KERNEL); if (hw_params == NULL) return -ENOMEM; /*1st question is, what is the use of above allocated memory I do not see this being used in this function*/ ... memcpy(&fe->dpcm[fe_substream->stream].hw_params, params, sizeof(struct snd_pcm_hw_params)); /* 2nd question is in above memcpy there is parameter mismatch &fe->dpcm[fe_substream->stream].hw_params is of structure type struct snd_pcm_hw_params params argument is of the structure type struct snd_compr_params the definition of the two structures are different, how this is working? this will over right the destination with junk value which will return in error when this memory is accessed, after memcpy this cpu_dai hw_params returing with EINVAL error, please explain why this is done this way. */ ... } Regards, Santosh M G.

4 4

[alsa-devel] [PATCH 1/1] hdspmixer: Add support for RME AIO AEB boards
by Adrian Knoth 05 Jan '14

05 Jan '14

AIO allows to connect Analogue Extension Boards (4in/4out). Provide those channels in hdspmixer, too. The channel maps were copied from the driver, hence the changed casing in the comments. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/hdspmixer/src/HDSPMixerCard.cxx b/hdspmixer/src/HDSPMixerCard.cxx index 78b88b6..ce40ba7 100644 --- a/hdspmixer/src/HDSPMixerCard.cxx +++ b/hdspmixer/src/HDSPMixerCard.cxx @@ -348,9 +348,9 @@ void HDSPMixerCard::adjustSettings() { switch (speed_mode) { case 0: // SS - channels_input = 14; - channels_playback = 16; - channels_output = 16; /* SS 2xAnalog+2xAES+2xSPDIF+4xADAT+2xHeadphones */ + channels_input = 18; + channels_playback = 20; + channels_output = 20; /* SS 2xAnalog+2xAES+2xSPDIF+8xADAT+2xHeadphones+4xAEB */ channel_map_input = channel_map_aio_in_ss; channel_map_playback = channel_map_aio_out_ss; dest_map = dest_map_aio_ss; @@ -358,9 +358,9 @@ void HDSPMixerCard::adjustSettings() { meter_map_playback = channel_map_aio_out_ss; break; case 1: // DS - channels_input = 10; - channels_playback = 12; - channels_output = 12; /* DS 2xAnalog+2xAES+2xSPDIF+4xADAT(SMUX)+2xHeadphones */ + channels_input = 14; + channels_playback = 16; + channels_output = 16; /* DS 2xAnalog+2xAES+2xSPDIF+4xADAT(SMUX)+2xHeadphones+4xAEB */ channel_map_input = channel_map_aio_in_ds; channel_map_playback = channel_map_aio_out_ds; dest_map = dest_map_aio_ds; @@ -368,9 +368,9 @@ void HDSPMixerCard::adjustSettings() { meter_map_playback = channel_map_aio_out_ds; break; case 2: // QS - channels_input = 8; - channels_playback = 10; - channels_output = 10; /* QS 2xAnalog+2xAES+2xSPDIF+2xADAT(SMUX)+2xHeadphones */ + channels_input = 12; + channels_playback = 14; + channels_output = 14; /* QS 2xAnalog+2xAES+2xSPDIF+2xADAT(SMUX)+2xHeadphones+4xAEB */ channel_map_input = channel_map_aio_in_qs; channel_map_playback = channel_map_aio_out_qs; dest_map = dest_map_aio_qs; diff --git a/hdspmixer/src/HDSPMixerOutput.cxx b/hdspmixer/src/HDSPMixerOutput.cxx index 5a88f41..0053fe0 100644 --- a/hdspmixer/src/HDSPMixerOutput.cxx +++ b/hdspmixer/src/HDSPMixerOutput.cxx @@ -87,51 +87,57 @@ static char const *labels_rpm_input[5] = { "Mic" }; -static char const *labels_aio_ss_input[14] = { +static char const *labels_aio_ss_input[18] = { "AN 1", "AN 2", "SP.L", "SP.R", "AES.L", "AES.R", "A 1", "A 2", "A 3", "A 4", - "A 5", "A 6", "A 7", "A 8" + "A 5", "A 6", "A 7", "A 8", + "AEB 1", "AEB 2", "AEB 3," "AEB 4" }; -static char const *labels_aio_ss_playback[16] = { +static char const *labels_aio_ss_playback[20] = { "AN 1", "AN 2", "SP.L", "SP.R", "AES.L", "AES.R", "A 1", "A 2", "A 3", "A 4", "A 5", "A 6", "A 7", "A 8", - "PH.L", "PH.R" + "PH.L", "PH.R", + "AEB 1", "AEB 2", "AEB 3", "AEB 4" }; -static char const *labels_aio_ds_input[10] = { +static char const *labels_aio_ds_input[14] = { "AN 1", "AN 2", "SP.L", "SP.R", "AES.L", "AES.R", - "A 1", "A 2", "A 3", "A 4" + "A 1", "A 2", "A 3", "A 4", + "AEB 1", "AEB 2", "AEB 3", "AEB 4" }; -static char const *labels_aio_ds_playback[12] = { +static char const *labels_aio_ds_playback[16] = { "AN 1", "AN 2", "SP.L", "SP.R", "AES.L", "AES.R", "A 1", "A 2", "A 3", "A 4", - "PH.L", "PH.R" + "PH.L", "PH.R", + "AEB 1", "AEB 2", "AEB 3", "AEB 4" }; -static char const *labels_aio_qs_input[8] = { +static char const *labels_aio_qs_input[12] = { "AN 1", "AN 2", "SP.L", "SP.R", "AES.L", "AES.R", - "A 1", "A 2" + "A 1", "A 2", + "AEB 1", "AEB 2", "AEB 3", "AEB 4" }; -static char const *labels_aio_qs_playback[10] = { +static char const *labels_aio_qs_playback[14] = { "AN 1", "AN 2", "SP.L", "SP.R", "AES.L", "AES.R", "A 1", "A 2", "PH.L", "PH.R", + "AEB 1", "AEB 2", "AEB 3", "AEB 4" }; diff --git a/hdspmixer/src/HDSPMixerSelector.cxx b/hdspmixer/src/HDSPMixerSelector.cxx index 084c80d..3dcbad3 100644 --- a/hdspmixer/src/HDSPMixerSelector.cxx +++ b/hdspmixer/src/HDSPMixerSelector.cxx @@ -81,28 +81,34 @@ static char const *destinations_raydat_qs[6] = { }; -static char const *destinations_aio_ss[8] = { +static char const *destinations_aio_ss[10] = { "AN 1+2", "AES", "SPDIF", "A 1+2", "A 3+4", "A 5+6", "A 7+8", - "Phones" + "Phones", + "AEB 1+2", + "AEB 3+4" }; -static char const *destinations_aio_ds[6] = { +static char const *destinations_aio_ds[8] = { "AN 1+2", "AES", "SPDIF", "A 1+2", "A 3+4", - "Phones" + "Phones", + "AEB 1+2", + "AEB 3+4" }; -static char const *destinations_aio_qs[5] = { +static char const *destinations_aio_qs[7] = { "AN 1+2", "AES", "SPDIF", "A 1+2", - "Phones" + "Phones", + "AEB 1+2", + "AEB 3+4" }; static char const *destinations_mf_ss[10] = { @@ -318,15 +324,15 @@ void HDSPMixerSelector::setLabels() } else if (HDSPeAIO == type) { switch (sm) { case 0: - max_dest = 8; + max_dest = 10; destinations = destinations_aio_ss; break; case 1: - max_dest = 6; + max_dest = 8; destinations = destinations_aio_ds; break; case 2: - max_dest = 5; + max_dest = 7; destinations = destinations_aio_qs; break; } diff --git a/hdspmixer/src/HDSPMixerWindow.cxx b/hdspmixer/src/HDSPMixerWindow.cxx index e1fe18c..342efb2 100644 --- a/hdspmixer/src/HDSPMixerWindow.cxx +++ b/hdspmixer/src/HDSPMixerWindow.cxx @@ -753,12 +753,12 @@ void HDSPMixerWindow::restoreDefaults(int card) num_modes = 3; break; case HDSPeAIO: - chnls[0] = 14; - chnls[1] = 10; - chnls[2] = 8; - maxdest[0] = 8; - maxdest[1] = 6; - maxdest[2] = 5; + chnls[0] = 18; + chnls[1] = 14; + chnls[2] = 12; + maxdest[0] = 10; + maxdest[1] = 8; + maxdest[2] = 7; num_modes = 3; break; case HDSPeRayDAT: diff --git a/hdspmixer/src/channelmap.cxx b/hdspmixer/src/channelmap.cxx index e3ca15c..b7bdded 100644 --- a/hdspmixer/src/channelmap.cxx +++ b/hdspmixer/src/channelmap.cxx @@ -247,38 +247,44 @@ char channel_map_raydat_qs[HDSPM_MAX_CHANNELS] = { // HDSPe AIO -char dest_map_aio_ss[8] = { +char dest_map_aio_ss[10] = { 0, // Analogue 8, // AES 10, // SPDIF 12, 14, 16, 18, // ADAT - 6 // Phones + 6, // Phones + 2, // AEB 1+2 + 4 // AEB 3+4 }; -char dest_map_aio_ds[6] = { +char dest_map_aio_ds[8] = { 0, // Analogue 8, // AES 10, // SPDIF 12, 16, // ADAT - 6 // Phones + 6, // Phones + 2, // AEB 1+2 + 4 // AEB 3+4 }; -char dest_map_aio_qs[5] = { +char dest_map_aio_qs[7] = { 0, // Analogue 8, // AES 10, // SPDIF 12, // ADAT - 6 // Phone + 6, // Phone + 2, // AEB 1+2 + 4 // AEB 3+4 }; char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS] = { 0, 1, /* line in */ - 8, 9, /* AES in, */ - 10, 11, /* SPDIF in */ + 8, 9, /* aes in, */ + 10, 11, /* spdif in */ 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT in */ - -1, -1, - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -288,11 +294,12 @@ char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS] = { char channel_map_aio_out_ss[HDSPM_MAX_CHANNELS] = { 0, 1, /* line out */ - 8, 9, /* AES out */ - 10, 11, /* SPDIF out */ + 8, 9, /* aes out */ + 10, 11, /* spdif out */ 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT out */ 6, 7, /* phone out */ - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -302,10 +309,11 @@ char channel_map_aio_out_ss[HDSPM_MAX_CHANNELS] = { char channel_map_aio_in_ds[HDSPM_MAX_CHANNELS] = { 0, 1, /* line in */ - 8, 9, /* AES in */ - 10, 11, /* SPDIF in */ - 12, 14, 16, 18, /* ADAT in */ - -1, -1, -1, -1, -1, -1, + 8, 9, /* aes in */ + 10, 11, /* spdif in */ + 12, 14, 16, 18, /* adat in */ + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -316,11 +324,11 @@ char channel_map_aio_in_ds[HDSPM_MAX_CHANNELS] = { char channel_map_aio_out_ds[HDSPM_MAX_CHANNELS] = { 0, 1, /* line out */ - 8, 9, /* AES out */ - 10, 11, /* SPDIF out */ - 12, 14, 16, 18, /* ADAT out */ + 8, 9, /* aes out */ + 10, 11, /* spdif out */ + 12, 14, 16, 18, /* adat out */ 6, 7, /* phone out */ - -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -331,10 +339,11 @@ char channel_map_aio_out_ds[HDSPM_MAX_CHANNELS] = { char channel_map_aio_in_qs[HDSPM_MAX_CHANNELS] = { 0, 1, /* line in */ - 8, 9, /* AES in */ - 10, 11, /* SPDIF in */ - 12, 16, /* ADAT in */ - -1, -1, -1, -1, -1, -1, -1, -1, + 8, 9, /* aes in */ + 10, 11, /* spdif in */ + 12, 16, /* adat in */ + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -345,11 +354,12 @@ char channel_map_aio_in_qs[HDSPM_MAX_CHANNELS] = { char channel_map_aio_out_qs[HDSPM_MAX_CHANNELS] = { 0, 1, /* line out */ - 8, 9, /* AES out */ - 10, 11, /* SPDIF out */ - 12, 16, /* ADAT out */ + 8, 9, /* aes out */ + 10, 11, /* spdif out */ + 12, 16, /* adat out */ 6, 7, /* phone out */ - -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, diff --git a/hdspmixer/src/channelmap.h b/hdspmixer/src/channelmap.h index 51b69ce..854e5a1 100644 --- a/hdspmixer/src/channelmap.h +++ b/hdspmixer/src/channelmap.h @@ -112,12 +112,12 @@ extern char channel_map_raydat_qs[HDSPM_MAX_CHANNELS]; // HDSPe AIO -extern char dest_map_aio_ss[8]; +extern char dest_map_aio_ss[10]; -extern char dest_map_aio_ds[6]; +extern char dest_map_aio_ds[8]; -extern char dest_map_aio_qs[5]; +extern char dest_map_aio_qs[7]; extern char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS]; -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH 0/3]: ALSA: compress: sampling rate updates
by Vinod Koul 05 Jan '14

05 Jan '14

Here are three patches which add more comments for sampling rate usage, fix the descriptor usage and lastly remove a check for SNDRV_PCM_RATE_xxx check Thanks Vinod Koul (3): ALSA: compress: remove the sample rate check ALSA: compress: update comment for sample rate in snd_codec ALSA: compress: update struct snd_codec_desc for sample rate include/uapi/sound/compress_params.h | 6 ++++-- sound/core/compress_offload.c | 3 --- 2 files changed, 4 insertions(+), 5 deletions(-

2 4

[alsa-devel] fix typo in rme9652.c
by Alexander Aristov 05 Jan '14

05 Jan '14

--- a/sound/pci/rme9652/rme9652.c 2013-12-20 15:51:33.000000000 +0000 +++ b/sound/pci/rme9652/rme9652.c 2013-12-24 23:17:20.740138731 +0000 @@ -285,7 +285,7 @@ /* ADAT channels are remapped */ 1, 3, 5, 7, 9, 11, 13, 15, /* channels 8 and 9 are S/PDIF */ - 24, 25 + 24, 25, /* others don't exist */ -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1 }; Regards, Alexander

2 1

[alsa-devel] [PATCH 0/3] use named constants for pci_power_t values
by Julia Lawall 05 Jan '14

05 Jan '14

2 2

[alsa-devel] [PATCH] ALSA: hda - Disable Front HP jack detection on Gigabyte Z87X-UD3H
by David Henningsson 05 Jan '14

05 Jan '14

This motherboard seems to have a flaky jack detection - when the front HP is not present, the jack state quickly switches on and off. This has been reported by three people in the bug, so I doubt it's a user error this time. BugLink: https://bugs.launchpad.net/bugs/1248116 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 16c0caa..6a705a8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1810,6 +1810,7 @@ enum { ALC882_FIXUP_ACER_ASPIRE_7736, ALC882_FIXUP_ASUS_W90V, ALC889_FIXUP_CD, + ALC889_FIXUP_FRONT_HP_NO_PRESENCE, ALC889_FIXUP_VAIO_TT, ALC888_FIXUP_EEE1601, ALC882_FIXUP_EAPD, @@ -2027,6 +2028,15 @@ static const struct hda_fixup alc882_fixups[] = { { } } }, + [ALC889_FIXUP_FRONT_HP_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x02214120 }, /* Front HP jack is flaky, disable jack detect */ + { } + }, + .chained = true, + .chain_id = ALC889_FIXUP_CD, + }, [ALC889_FIXUP_VAIO_TT] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -2258,7 +2268,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), - SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), -- 1.7.9.5

2 1

[alsa-devel] [PATCH (alsa-plugins)] a52: Propagate errors from hw
by David Henningsson 05 Jan '14

05 Jan '14

Several callbacks returned 0 even though the hw was failing, leading applications to believe everything was fine when it wasn't. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- a52/pcm_a52.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/a52/pcm_a52.c b/a52/pcm_a52.c index 7188809..59a49ed 100644 --- a/a52/pcm_a52.c +++ b/a52/pcm_a52.c @@ -195,8 +195,8 @@ static int a52_drain(snd_pcm_ioplug_t *io) err = write_out_pending(io, rec); if (err < 0) return err; - snd_pcm_drain(rec->slave); - return 0; + + return snd_pcm_drain(rec->slave); } /* check whether the areas consist of a continuous interleaved stream */ @@ -479,16 +479,14 @@ static int a52_start(snd_pcm_ioplug_t *io) { struct a52_ctx *rec = io->private_data; - snd_pcm_start(rec->slave); - return 0; + return snd_pcm_start(rec->slave); } static int a52_stop(snd_pcm_ioplug_t *io) { struct a52_ctx *rec = io->private_data; - snd_pcm_drop(rec->slave); - return 0; + return snd_pcm_drop(rec->slave); } /* release resources */ @@ -645,7 +643,7 @@ static int a52_close(snd_pcm_ioplug_t *io) a52_free(rec); if (rec->slave) - snd_pcm_close(rec->slave); + return snd_pcm_close(rec->slave); return 0; } -- 1.8.4.2

2 1

Re: [alsa-devel] ALSA lots of namespace issues
by Takashi Iwai 05 Jan '14

05 Jan '14

At Sun, 29 Dec 2013 12:09:36 -0800, Stephen Hemminger wrote: > > When doing my namespace checks, I find lots of local functions declared > global (and dead code) in ALSA. Which tree is used for submitting patches > for -next? For sound/* in general, use my sound.git tree for-next branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git For sound/soc/*, maybe better to use Mark's sound git tree. git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git Mark is managing many small topic branches, but for-next branch should be available to make patches, at least. > Some examples from 'make namespacecheck' There are quite lots of hits indeed. But I see a few of them are false-positive (depending on kconfig) and a few would be still better to keep for API consistency. If you can give a complete list beforehand, we can sort out such. Ideally, it'd be best if we have a on-demand loading of library code in the kernel, instead of loading a whole "library module"... thanks, Takashi > > sound/core/compress_offload.o > snd_compress_deregister > snd_compress_register > sound/core/control.o > snd_ctl_activate_id > snd_ctl_rename_id > snd_ctl_replace > sound/core/device.o > snd_device_disconnect > sound/core/isadma.o > snd_dma_disable > sound/core/jack.o > snd_jack_set_key > snd_jack_set_parent > sound/core/oss/mixer_oss.o > snd_mixer_oss_ioctl_card > sound/core/pcm_dmaengine.o > snd_dmaengine_pcm_close_release_chan > snd_dmaengine_pcm_open_request_chan > sound/core/pcm_lib.o > snd_interval_ratnum > sound/core/pcm_memory.o > snd_pcm_lib_preallocate_free > sound/core/pcm_misc.o > snd_pcm_format_size > snd_pcm_rate_bit_to_rate > sound/core/pcm_native.o > snd_pcm_hw_constraints_complete > snd_pcm_hw_constraints_init > sound/core/rawmidi.o > snd_rawmidi_drain_input > snd_rawmidi_drop_output > sound/core/timer.o > snd_timer_continue > sound/drivers/dummy.o > dummy_models > model_ac97 > model_ca0106 > model_emu10k1 > model_ice1712 > model_rme9652 > model_uda1341 > sound/drivers/mpu401/mpu401_uart.o > snd_mpu401_uart_interrupt_tx > sound/drivers/opl3/opl3_lib.o > snd_opl3_interrupt > sound/firewire/amdtp.o > amdtp_syt_intervals > sound/i2c/i2c.o > snd_i2c_probeaddr > sound/i2c/other/ak4113.o > snd_ak4113_check_rate_and_errors > snd_ak4113_reg_write > snd_ak4113_reinit > sound/i2c/other/ak4114.o > snd_ak4114_check_rate_and_errors > snd_ak4114_reg_write > sound/isa/sb/sb16_main.o > snd_sb16dsp_configure > snd_sb16dsp_get_pcm_ops > sound/isa/sb/sb_common.o > snd_sbdsp_get_byte > sound/isa/sb/sb_mixer.o > snd_sbmixer_add_ctl (export only) > sound/oss/dmabuf.o > DMAbuf_start_dma > sound/oss/sb_ess.o > ess_midi_init > sound/oss/soundcard.o > dev_list > sound/pci/asihpi/hpidebug.o > hpi_debug_data > hpi_debug_init > hpi_debug_level_get > hpi_debug_level_set > sound/pci/asihpi/hpifunc.o > hpi_adapter_close > hpi_adapter_enumerate_property > hpi_adapter_get_mode > hpi_adapter_get_module_by_index > hpi_adapter_open > hpi_adapter_set_mode > hpi_adapter_set_mode_ex > hpi_adapter_set_property > hpi_aesebu_receiver_get_channel_status > hpi_aesebu_receiver_get_sample_rate > hpi_aesebu_receiver_get_user_data > hpi_aesebu_receiver_query_format > hpi_aesebu_transmitter_get_channel_status > hpi_aesebu_transmitter_query_format > hpi_aesebu_transmitter_set_channel_status > hpi_aesebu_transmitter_set_sample_rate > hpi_aesebu_transmitter_set_user_data > hpi_bitstream_get_activity > hpi_bitstream_set_clock_edge > hpi_bitstream_set_data_polarity > hpi_cobranet_get_ip_address > hpi_cobranet_get_macaddress > hpi_cobranet_get_static_ip_address > hpi_cobranet_hmi_get_status > hpi_cobranet_hmi_read > hpi_cobranet_hmi_write > hpi_cobranet_set_ip_address > hpi_cobranet_set_static_ip_address > hpi_compander_get_attack_time_constant > hpi_compander_get_decay_time_constant > hpi_compander_get_enable > hpi_compander_get_makeup_gain > hpi_compander_get_ratio > hpi_compander_get_threshold > hpi_compander_set_attack_time_constant > hpi_compander_set_decay_time_constant > hpi_compander_set_enable > hpi_compander_set_makeup_gain > hpi_compander_set_ratio > hpi_compander_set_threshold > hpi_format_to_msg > hpi_indexes_to_handle > hpi_instream_ancillary_get_info > hpi_instream_ancillary_reset > hpi_instream_ancillary_write > hpi_instream_group_get_map > hpi_instream_host_buffer_allocate > hpi_instream_host_buffer_get_info > hpi_instream_wait_start > hpi_meter_get_peak_ballistics > hpi_meter_get_rms > hpi_meter_get_rms_ballistics > hpi_meter_set_peak_ballistics > hpi_meter_set_rms_ballistics > hpi_microphone_get_phantom_power > hpi_microphone_set_phantom_power > hpi_mixer_close > hpi_mixer_store > hpi_outstream_ancillary_get_info > hpi_outstream_ancillary_read > hpi_outstream_ancillary_reset > hpi_outstream_group_get_map > hpi_outstream_host_buffer_allocate > hpi_outstream_host_buffer_free > hpi_outstream_host_buffer_get_info > hpi_outstream_set_format > hpi_outstream_set_punch_in_out > hpi_outstream_set_time_scale > hpi_outstream_set_velocity > hpi_outstream_sinegen > hpi_outstream_wait_start > hpi_pad_get_artist > hpi_pad_get_channel_name > hpi_pad_get_comment > hpi_pad_get_program_type > hpi_pad_get_rdsPI > hpi_pad_get_title > hpi_parametric_eq_get_band > hpi_parametric_eq_get_coeffs > hpi_parametric_eq_get_info > hpi_parametric_eq_set_band > hpi_parametric_eq_set_state > hpi_sample_clock_get_auto > hpi_sample_clock_get_local_rate_lock > hpi_sample_clock_set_auto > hpi_sample_clock_set_local_rate_lock > hpi_silence_detector_get_delay > hpi_silence_detector_get_enable > hpi_silence_detector_get_event_enable > hpi_silence_detector_get_state > hpi_silence_detector_get_threshold > hpi_silence_detector_set_delay > hpi_silence_detector_set_enable > hpi_silence_detector_set_event_enable > hpi_silence_detector_set_threshold > hpi_stream_estimate_buffer_size > hpi_stream_response_to_legacy > hpi_subsys_get_adapter > hpi_subsys_get_num_adapters > hpi_subsys_get_version_ex > hpi_tone_detector_get_enable > hpi_tone_detector_get_event_enable > hpi_tone_detector_get_frequency > hpi_tone_detector_get_state > hpi_tone_detector_get_threshold > hpi_tone_detector_set_enable > hpi_tone_detector_set_event_enable > hpi_tone_detector_set_threshold > hpi_tuner_get_deemphasis > hpi_tuner_get_hd_radio_dsp_version > hpi_tuner_get_hd_radio_sdk_version > hpi_tuner_get_hd_radio_signal_blend > hpi_tuner_get_hd_radio_signal_quality > hpi_tuner_get_mode > hpi_tuner_get_program > hpi_tuner_get_raw_rf_level > hpi_tuner_get_rds > hpi_tuner_get_rf_level > hpi_tuner_get_status > hpi_tuner_query_deemphasis > hpi_tuner_query_program > hpi_tuner_set_deemphasis > hpi_tuner_set_hd_radio_signal_blend > hpi_tuner_set_mode > hpi_tuner_set_program > hpi_volume_auto_fade > hpi_volume_auto_fade_profile > hpi_volume_query_auto_fade_profile > hpi_volume_query_channels > hpi_vox_get_threshold > hpi_vox_set_threshold > sound/pci/asihpi/hpimsginit.o > hpi_init_responseV1 > sound/pci/cs46xx/dsp_spos.o > cs46xx_dsp_set_iec958_volume > sound/pci/emu10k1/io.o > snd_emu10k1_voice_half_loop_intr_enable > snd_emu10k1_voice_set_loop_stop > sound/pci/hda/hda_codec.o > snd_hda_check_board_codec_sid_config > snd_hda_codec_amp_init > snd_hda_ctls_clear > snd_hda_is_supported_format > snd_hda_lock_devices > snd_hda_unlock_devices > sound/pci/hda/hda_eld.o > snd_hdmi_get_eld_size > sound/pci/hda/hda_generic.o > hda_extra_out_badness > hda_main_out_badness > snd_hda_add_new_path > snd_hda_parse_nid_path > sound/pci/hda/hda_hwdep.o > snd_hda_get_hint > sound/pci/hda/hda_jack.o > snd_hda_jack_tbl_new > sound/pci/ice1712/wm8766.o > snd_wm8766_set_master_mode > snd_wm8766_set_power > sound/pci/ice1712/wm8776.o > snd_wm8776_set_adc_if > snd_wm8776_set_dac_if > snd_wm8776_set_master_mode > sound/pci/lola/lola_clock.o > lola_set_clock > sound/pci/lola/lola_mixer.o > lola_restore_mixer > lola_save_mixer > sound/pci/lx6464es/lx_core.o > lx_buffer_cancel > lx_buffer_free > lx_dsp_es_check_pipeline > lx_dsp_read_async_events > lx_pipe_sample_count > lx_pipe_state > lx_plx_mbox_read > lx_plx_mbox_write > lx_stream_sample_position > lx_stream_state > sound/pci/trident/trident_main.o > snd_trident_alloc_voice > snd_trident_free_voice > snd_trident_start_voice > snd_trident_stop_voice > snd_trident_write_voice_regs > sound/soc/atmel/atmel-pcm.o > atmel_pcm_free > atmel_pcm_mmap > atmel_pcm_new > sound/soc/codecs/arizona.o > arizona_set_output_mode > sound/soc/codecs/cx20442.o > v253_ops > sound/soc/codecs/max98090.o > max98090_mic_detect > sound/soc/codecs/max98095.o > max98095_jack_detect > sound/soc/codecs/sigmadsp.o > process_sigma_firmware_regmap > sound/soc/codecs/tpa6130a2.o > tpa6130a2_add_controls > tpa6130a2_stereo_enable > sound/soc/codecs/wl1273.o > wl1273_get_format > sound/soc/codecs/wm5100.o > wm5100_detect > sound/soc/codecs/wm8962.o > wm8962_mic_detect > sound/soc/codecs/wm8996.o > wm8996_detect > sound/soc/codecs/wm_hubs.o > wm_hubs_dcs_done > wm_hubs_update_class_w > sound/soc/mxs/mxs-pcm.o > mxs_pcm_platform_register > sound/soc/mxs/mxs-saif.o > mxs_saif_get_mclk > sound/soc/soc-core.o > __tracepoint_snd_soc_preg_read > __tracepoint_snd_soc_preg_write > __tracepoint_snd_soc_reg_read > __tracepoint_snd_soc_reg_write > snd_soc_add_platform_controls > snd_soc_codec_readable_register > snd_soc_codec_set_pll > snd_soc_codec_set_sysclk > snd_soc_dai_set_bclk_ratio > snd_soc_dai_set_channel_map > snd_soc_dai_set_clkdiv > snd_soc_dai_set_pll > snd_soc_dai_set_tdm_slot > snd_soc_dai_set_tristate > snd_soc_debugfs_root > snd_soc_free_ac97_codec > snd_soc_get_dai_substream > snd_soc_get_pcm_runtime > snd_soc_limit_volume > snd_soc_new_ac97_codec > snd_soc_of_get_dai_name > snd_soc_of_parse_daifmt > snd_soc_poweroff > snd_soc_resume > snd_soc_set_ac97_ops > snd_soc_set_ac97_ops_of_reset > snd_soc_suspend > snd_soc_update_bits_locked > soc_ac97_ops > sound/soc/soc-dapm.o > dapm_mark_dirty > snd_soc_dapm_get_pin_status > snd_soc_dapm_ignore_suspend > snd_soc_dapm_mux_update_power > snd_soc_dapm_weak_routes > sound/soc/soc-devres.o > devm_snd_soc_register_card > sound/soc/soc-jack.o > snd_soc_jack_add_zones > snd_soc_jack_get_type > snd_soc_jack_notifier_register > snd_soc_jack_notifier_unregister > sound/soc/soc-pcm.o > snd_soc_dpcm_be_can_update > snd_soc_dpcm_be_get_state > snd_soc_dpcm_be_set_state > snd_soc_dpcm_can_be_free_stop > snd_soc_dpcm_can_be_params > snd_soc_dpcm_fe_can_update > snd_soc_dpcm_get_substream > snd_soc_platform_trigger > soc_dpcm_be_digital_mute > sound/soc/soc-utils.o > snd_soc_calc_frame_size > sound/soc/tegra/tegra_asoc_utils.o > tegra_asoc_utils_init > tegra_asoc_utils_set_ac97_rate > sound/soc/tegra/tegra_pcm.o > tegra_pcm_platform_register > tegra_pcm_platform_unregister > sound/synth/emux/emux_synth.o > snd_emux_lock_voice > snd_emux_unlock_voice > sound/synth/emux/soundfont.o > snd_sf_linear_to_log > sound/synth/util_mem.o > snd_util_mem_alloc > snd_util_mem_free > sound/usb/endpoint.o > snd_usb_handle_sync_urb > sound/usb/pcm.o > snd_usb_pcm_delay > sound/usb/usx2y/us122l.o > usb_stream_free > usb_stream_new > usb_stream_start > usb_stream_stop >

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[alsa-devel] Conexant CX20588 in Acer 3830TG fix for internal mic amps
by Christopher K. 05 Jan '14

05 Jan '14

Hi, the internal mic in my notebook does not work (has never worked with pulse) with the default driver settings. Setting Node 0x23 (internal mic boost volume) from Amp-In vals: [0x03 0x03] to Amp-In vals: [0x02 0x00] makes it work in Skype and shows peaks in pulseaudio. I have no idea what these numbers represent, but apparently it doesn't work if they're the same. Is there any way to figure out what this does and which settings are correct? Info: http://www.alsa-project.org/db/?f=be2d83a353d47586ac2197fdb56d2085eec61b3f Any help would be greatly appreciated. Regards, Chris

2 1

[alsa-devel] ASoC updates for v3.14
by Mark Brown 05 Jan '14

05 Jan '14

The following changes since commit 802eee95bde72fd0cd0f3a5b2098375a487d1eda: Linux 3.13-rc6 (2013-12-29 16:01:33 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14 for you to fetch changes up to 2cde51fbd0f310c8a2c5f977e665c0ac3945b46d: Merge remote-tracking branches 'asoc/topic/ad1836', 'asoc/topic/ad193x', 'asoc/topic/adav80x', 'asoc/topic/adsp', 'asoc/topic/ak4641', 'asoc/topic/ak4642', 'asoc/topic/arizona', 'asoc/topic/atmel', 'asoc/topic/au1x', 'asoc/topic/axi', 'asoc/topic/bcm2835', 'asoc/topic/blackfin', 'asoc/topic/cs4271', 'asoc/topic/cs42l52', 'asoc/topic/da7210', 'asoc/topic/davinci', 'asoc/topic/ep93xx', 'asoc/topic/fsl', 'asoc/topic/fsl-mxs', 'asoc/topic/generic', 'asoc/topic/hdmi', 'asoc/topic/jack', 'asoc/topic/jz4740', 'asoc/topic/max98090', 'asoc/topic/mxs', 'asoc/topic/omap', 'asoc/topic/pxa', 'asoc/topic/rcar', 'asoc/topic/s6000', 'asoc/topic/sai', 'asoc/topic/samsung', 'asoc/topic/sgtl5000', 'asoc/topic/spear', 'asoc/topic/ssm2518', 'asoc/topic/ssm2602', 'asoc/topic/tegra', 'asoc/topic/tlv320aic3x', 'asoc/topic/twl6040', 'asoc/topic/txx9', 'asoc/topic/uda1380', 'asoc/topic/width', 'asoc/topic/wm8510', 'asoc/topic/wm8523', 'asoc/topic/wm8580', 'asoc/topic/wm8711', 'asoc/topic/wm8728', 'asoc/topic/wm8731', 'asoc/topic/wm8741', 'asoc/topic/wm8750', 'asoc/topic/wm8753', 'asoc/topic/wm8776', 'asoc/topic/wm8804', 'asoc/topic/wm8900', 'asoc/topic/wm8901', 'asoc/topic/wm8940', 'asoc/topic/wm8962', 'asoc/topic/wm8974', 'asoc/topic/wm8985', 'asoc/topic/wm8988', 'asoc/topic/wm8990', 'asoc/topic/wm8991', 'asoc/topic/wm8994', 'asoc/topic/wm8995', 'asoc/topic/wm9081' and 'asoc/topic/x86' into asoc-next (2014-01-02 13:01:55 +0000) I see I missed from the tag that we also got a driver for the Freescale SAI controller. ---------------------------------------------------------------- ASoC: Updates for v3.14 Not a lot going on framework wise, partly due to Christmas at least in the case of the work I've been doing, but there's been quite a lot of cleanup activity going on and the usual trickle of new drivers: - Update to the generic DMA code to support deferred probe and managed resources. - New drivers for BCM2835 (used in Raspberry Pi), Tegra with MAX98090 and Analog Devices AXI I2S and S/PDIF controller IPs. - Device tree support for the simple card, max98090 and cs42l52. - Conversion of the Samsung drivers to native dmaengine, making them multiplatform compatible and hopefully helping keep them more modern and up to date. - More regmap conversions, including a very welcome one for twl6040 from Peter Ujfalusi. - A big overhaul of the DaVinci drivers also from Peter Ujfalusi. ---------------------------------------------------------------- Bo Shen (1): ASoC: atmel: sam9x5_wm8731: remove platform_set_drvdata Brian Austin (4): ASoC: cs42l52: Reorganize MICA/B Config and Select ASoC: cs42l52: Make MICA/B mixer dependent on mic config ASoC: cs42l52: Add devicetree support for CS42L52 ASoC: dt: binding: sound cs42l52 driver Charles Keepax (10): mfd: wm5110: Make DSP memories readable ASoC: wm5110: Expose input high pass filter controls ASoC: wm5110: Add basic support for ISRCs mfd: wm5110: Give new AIF2 registers defaults and mark as readable ASoC: wm_adsp: Stop region iteration when the desired region is found ASoC: wm_adsp: Remove duplicate info message for DSP RAM ready mfd: wm5110: Expose DRE control registers ASoC: wm5110: Expose switch controls for DRE ASoC: wm5110: Add FSH for ISRCs ASoC: wm5110: Add support for ASRC RATE 1 Dan Carpenter (1): ASoC: rcar: some dubious one-bit signed bitfields Fabio Estevam (29): ASoC: ad193x: Use IS_ENABLED() macro ASoC: adav80x: Use IS_ENABLED() macro ASoC: ak4642: Use IS_ENABLED() macro ASoC: uda1380: Use IS_ENABLED() macro ASoC: wm8510: Use IS_ENABLED() macro ASoC: wm8523: Use IS_ENABLED() macro ASoC: wm8580: Use IS_ENABLED() macro ASoC: wm8711: Use IS_ENABLED() macro ASoC: wm8728: Use IS_ENABLED() macro ASoC: wm8962: Use IS_ENABLED() macro ASoC: wm8995: Use IS_ENABLED() macro ASoC: soc-io: Use IS_ENABLED() macro ASoC: cs4271: Use IS_ENABLED() macro ASoC: da7210: Use IS_ENABLED() macro ASoC: ssm2602: Use IS_ENABLED() macro ASoC: wm8731: Use IS_ENABLED() macro ASoC: wm8741: Use IS_ENABLED() macro ASoC: wm8750: Use IS_ENABLED() macro ASoC: wm8753: Use IS_ENABLED() macro ASoC: wm8776: Use IS_ENABLED() macro ASoC: wm8804: Use IS_ENABLED() macro ASoC: wm8900: Use IS_ENABLED() macro ASoC: wm8985: Use IS_ENABLED() macro ASoC: wm8988: Use IS_ENABLED() macro AsoC: wm9081: Use IS_ENABLED() macro ASoC: wm9081: Use IS_ENABLED() macro ASoC: cs42l52: Use IS_ENABLED() macro ASoC: imx-spdif: Remove error message upon devm_kzalloc() failure ASoC: imx-ssi: Check the return value from clk_prepare_enable() Fengguang Wu (2): ASoC: axi-spdif: Use devm_ioremap_resource() instead of devm_request_and_ioremap() ASoC: axi-spdif: Use devm_ioremap_resource() instead of devm_request_and_ioremap() Florian Meier (1): ASoC: Add support for BCM2835 Jan Weitzel (1): ASoC: tlv320aic3x: no mono controls 3007 model Jarkko Nikula (1): ASoC: Rename mid-x86 directory to intel Jyri Sarha (2): ASoC: hdmi-codec: Add SNDRV_PCM_FMTBIT_32_LE playback format ASoC: hdmi-codec: Add devicetree binding with documentation Kuninori Morimoto (21): ASoC: rcar: rename GEN2_SRU to GEN2_SCU ASoC: rcar: remove unused register settings ASoC: rcar: separate regmap init common field ASoC: rcar: add rsnd_is_accessible_reg() ASoC: rcar: remove .path_init/exit from rsnd_gen_ops ASoC: rcar: remove rcar_gen_ops ASoC: rcar: add Gen2 sound support ASoC: simple-card: add Device Tree support ASoC: rcar: use devm_clk_get() instead of clk_get() ASoC: rsnd: gen: fixup Gen2 channel size ASoC: fsi: remove original filter from fsi_dma_probe() ASoC: rsnd: add rsnd_adg_set_ssi_clk() and cleanup adg ASoC: rsnd: tidyup ssi comment ASoC: rsnd: tidyup register naming ASoC: rsnd: make sure variable name for 44.1kHz/48kHz ASoC: rsnd: route setting is needed only Gen1 ASoC: rsnd: add rsnd_scu_transfer_start() ASoC: rsnd: INT_ENABLE is needed only Gen2 ASoC: rsnd: scu cleanup: add rsnd_scu_rate_ctrl() ASoC: rsnd: tidyup rsnd_ssi_master_clk_start() parameter ASoC: rsnd: add SRC (Sampling Rate Converter) support Lars-Peter Clausen (26): ASoC: Add resource managed snd_dmaengine_pcm_register() ASoC: spear: Use devm_snd_dmaengine_pcm_register ASoC: generic-dmaengine-pcm: Set BATCH flag when residue reporting is not supported ASoC: Set SNDRV_PCM_INFO_JOINT_DUPLEX for PCMs with symmetry constraints ASoC: bcm2835-i2s: Use devm_snd_dmaengine_pcm_register() ASoC: ep93xx: Use devm_snd_dmaengine_pcm_register() ASoC: fsl: Use devm_snd_dmaengine_pcm_register() ASoC: mxs: Use devm_snd_dmaengine_pcm_register() ASoC: ssm2602: Use core for applying symmetry constraints ASoC: jz4740-i2s: Use managed resources ASoC: jz4740: Use the generic dmaengine PCM driver dt: Add bindings documentation for the ADI AXI-I2S controller ASoC: Add support for the Analog Devices AXI-I2S core dt: Add bindings documentation for the ADI AXI-SPDIF audio controller ASoC: Add support for the Analog Devices AXI-SPDIF driver ASoC: au1x: Don't set unused struct snd_pcm_hardware fields ASoC: tegra: Don't set unused struct snd_pcm_hardware fields ASoC: blackfin: Don't set unused struct snd_pcm_hardware fields ASoC: davinci: Don't set unused struct snd_pcm_hardware fields ASoC: s6000: Don't set unused struct snd_pcm_hardware fields ASoC: omap: Don't set unused struct snd_pcm_hardware fields ASoC: soc-utils: Don't set unused struct snd_pcm_hardware fields ASoC: samsung: Don't set unused struct snd_pcm_hardware fields ASoC: txx9: Don't set unused struct snd_pcm_hardware fields ASoC: mmp: Don't set unused struct snd_pcm_hardware fields ASoC: ep93xx: Don't set unused struct snd_pcm_hardware fields Lucas Stach (1): ASoC: tegra: Tweak matching of AC97 components Mark Brown (45): Merge tag 'ib-asoc-1' of git://git.linaro.org/people/ljones/mfd into asoc-arizona ASoC: wm8940: Convert to table based control and DAPM init ASoC: wm8940: Convert to direct regmap API usage ASoC: wm8974: Convert to direct regmap API usage ASoC: wm8991: Use a supply to manage input power ASoC: wm8991: Convert to table based control and widget init ASoC: wm8991: Convert to direct regmap API usage ASoC: wm8991: Move basic initialisation to I2C level probe ASoC: wm8991: Verify device ID during probe() ASoC: ak4641: Convert to direct regmap API usage Merge tag 'v3.13-rc1' into asoc-arizona ASoC: wm5110: Hook up ADSP2 cores ASoC: wm8990: Convet to module_i2c_driver() ASoC: wm8990: Convert to table based control and DAPM init ASoC: wm8990: Use supplies to manage input power ASoC: wm8990: Convert to direct regmap API usage ASoC: cs42l52: Fix build Merge remote-tracking branch 'asoc/topic/dma' into asoc-bcm2835 Merge remote-tracking branch 'asoc/topic/dma' into asoc-fsl ASoC: ak4642: Convert to table based control init ASoC: ak4642: Convert to direct regmap API usage ASoC: ak4642: Convert to module_i2c_driver() Merge remote-tracking branch 'asoc/topic/symmetry' into asoc-fsl Merge remote-tracking branch 'asoc/topic/symmetry' into asoc-ssm2602 Merge remote-tracking branch 'asoc/topic/symmetry' into asoc-core ASoC: adsp: Use async writes where possible Merge tag 's3c64xx-dmaengine' of git://git.kernel.org/.../broonie/misc into asoc-samsung Merge remote-tracking branch 'asoc/fix/dma' into asoc-dma Merge branch 'topic/defer_probe' of git://git.infradead.org/users/vkoul/slave-dma into asoc-dma ASoC: samsung: Provide helper for DMA init ASoC: samsung: Use ASoC dmaengine code where possible ASoC: arizona: Use async writes ASoC: wm5102: Use async writes ASoC: wm5110: Use async writes ASoC: wm8997: Use async writes ASoC: fsl/mxs: Remove unnecessarily gendered language Merge tag 'ib-asoc-3.14-1' of git://git.linaro.org/people/ljones/mfd into asoc-twl6040 ALSA: Add params_width() helpers ASoC: ad1836: Reject unsupported bit sizes Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/dt' into asoc-next Merge remote-tracking branches 'asoc/topic/ad1836', 'asoc/topic/ad193x', 'asoc/topic/adav80x', 'asoc/topic/adsp', 'asoc/topic/ak4641', 'asoc/topic/ak4642', 'asoc/topic/arizona', 'asoc/topic/atmel', 'asoc/topic/au1x', 'asoc/topic/axi', 'asoc/topic/bcm2835', 'asoc/topic/blackfin', 'asoc/topic/cs4271', 'asoc/topic/cs42l52', 'asoc/topic/da7210', 'asoc/topic/davinci', 'asoc/topic/ep93xx', 'asoc/topic/fsl', 'asoc/topic/fsl-mxs', 'asoc/topic/generic', 'asoc/topic/hdmi', 'asoc/topic/jack', 'asoc/topic/jz4740', 'asoc/topic/max98090', 'asoc/topic/mxs', 'asoc/topic/omap', 'asoc/topic/pxa', 'asoc/topic/rcar', 'asoc/topic/s6000', 'asoc/topic/sai', 'asoc/topic/samsung', 'asoc/topic/sgtl5000', 'asoc/topic/spear', 'asoc/topic/ssm2518', 'asoc/topic/ssm2602', 'asoc/topic/tegra', 'asoc/topic/tlv320aic3x', 'asoc/topic/twl6040', 'asoc/topic/txx9', 'asoc/topic/uda1380', 'asoc/topic/width', 'asoc/topic/wm8510', 'asoc/topic/wm8523', 'asoc/topic/wm8580', 'asoc/topic/wm8711', 'asoc/topic/wm8728', 'asoc/topic/wm8731', 'asoc/topic/wm8741', 'asoc/topic/wm8750', 'asoc/topic/wm8753', 'asoc/topic/wm8776', 'asoc/topic/wm8804', 'asoc/topic/wm8900', 'asoc/topic/wm8901', 'asoc/topic/wm8940', 'asoc/topic/wm8962', 'asoc/topic/wm8974', 'asoc/topic/wm8985', 'asoc/topic/wm8988', 'asoc/topic/wm8990', 'asoc/topic/wm8991', 'asoc/topic/wm8994', 'asoc/topic/wm8995', 'asoc/topic/wm9081' and 'asoc/topic/x86' into asoc-next Markus Pargmann (1): ASoC: fsl-ssi: Move ac97 specific setup to seperate function Matthew Garrett (1): ASoC: fsl/mxs: Remove unnecessarily gendered language Nenghua Cao (1): ASoC: dapm: update DPCM runtime when mixer/mux changes Nicolin Chen (13): ASoC: soc-pcm: add symmetry for channels and sample bits ASoC: soc-pcm: move DAIs parameters cleaning into hw_free() ASoC: fsl_ssi: Add monaural audio support for non-ac97 interface ASoC: fsl_ssi: Implement symmetric_channels and symmetric_samplebits ASoC: soc-pcm: Drop the redundant snd_soc_dai_digital_mute() in soc_pcm_close() ASoC: fsl_ssi: Add DAI master mode support for SSI on i.MX series ASoC: fsl_sai: Keep symmetry for clk_enable() and clk_disable() ASoC: fsl_sai: Use snd_pcm_format_width() ASoC: fsl_sai: Drop useless channels check in hw_params() ASoC: fsl_sai: Drop useless ret in startup() ASoC: fsl_sai: Make dev_err information neater ASoC: fsl_sai: Sort local variable in general way ASoC: imx-spdif: Use snd-soc-dummy CODEC driver to link card Oskar Schirmer (1): ASoC: fsl: imx-ssi: omit ssi counter to avoid harm in unbalanced situation Peter Ujfalusi (26): ASoC: davinci: Kconfig: Remove help section for SND_DAVINCI_SOC ASoC: davinci: Kconfig/Makefile: Generic EVM machine driver related cleanup ASoC: davinci: kconfig: Prepare for AM43xx support ASoC: davinci-evm: Do not include davinci-mcasp.h ASoC: davinci-evm: Switch to use .dai_fmt of snd_soc_dai_link(s) ASoC: davinci-mcasp: Move DAVINCI_MCASP_RATE from header to source file ASoC: davinci-mcasp: Remove unused DAVINCI_MCASP_I2S/DIT_DAI defines ASoC: davinci-mcasp: Correct dai driver struct initialization for 2nd dai ASoC: davinci-mcasp: Move register definitions to header file ASoC: davinci-mcasp: Move private struct definition to source file ASoC: davinci-mcasp: Remove unused DAVINCI_MCASP_NUM_SERIALIZER define ASoC: davinci-mcasp: Do not inline the mcasp_set_ctl_reg function ASoC: davinci-mcasp: Rename private struct and it's users (dev -> mcasp) ASoC: davinci-mcasp: Be consistent with the use of base in davinci_mcasp_set_dai_fmt ASoC: davinci-mcasp: Simplify FIFO configuration code ASoC: davinci-mcasp: Data source (bus) selection support ASoC: davinci-mcasp: Fix synchronous master receive mode ASoC: davinci-mcasp: Support for McASP version found in DRA7xx ASoC: davinci-mcasp: Change IO functions parameter list ASoC: davinci-mcasp: Correct am33xx snd_platform_data name ASoC: davinci-mcasp: Support for fck reparenting mfd: twl6040: reg_defaults support for regmap ASoC: twl6040: Rename twl6040_is_path_unmuted -> twl6040_can_write_to_chip ASoC: twl6040: Custom caching for sensitive DL1/2 path registers ASoC: twl6040: Remove register restore functionality ASoC: twl6040: Remove self managed local reg_cache support Qiao Zhou (1): ASoC: mmp-pcm: config pcm slave via generic dmaengine Richard Fitzgerald (1): ASoC: wm5110: Add extra AIF2 channels Sachin Kamat (2): ASoC: mcbsp: Trivial cleanup in asoc-ti-mcbsp.h ASoC: samsung: Trivial cleanups in header files Shawn Guo (2): ASoC: sgtl5000: read chip revision for once ASoC: sgtl5000: clean up sgtl5000_enable_regulators() Stephen Warren (11): ASoC: tegra: add tegra+MAX98090 machine driver ASoC: max98090: add DT binding document for MAX98090 CODEC ASoC: restructure dmaengine_pcm_request_chan_of() ASoC: dmaengine: add custom DMA config to snd_dmaengine_pcm_config dma: add channel request API that supports deferred probe ASoC: dmaengine: support deferred probe for DMA channels ASoC: dmaengine: fix deferred probe detection ASoC: ep93xx: get rid of ep93xx-pcm-audio struct device ASoC: ep93xx: remove custom DMA alloc compat function ASoC: SPEAr: get rid of spear-pcm-audio struct device ASoC: SPEAr: remove custom DMA alloc compat function Takashi Iwai (2): ASoC: Fix build without CONFIG_GPIOLIB ASoC: ssm2518: Fix off-by-one error by ffs() Tomasz Figa (6): spi: s3c64xx: Do not require legacy DMA API in case of S3C64XX ARM: s3c64xx: Add support for DMA using generic amba-pl08x driver clk: samsung: s3c64xx: Add aliases for DMA clocks ARM: s3c64xx: Remove legacy DMA driver clk: samsung: s3c64xx: Remove clock aliases of old DMA driver spi: s3c64xx: Always select S3C64XX_PL080 when ARCH_S3C64XX is enabled Victor Kamensky (1): ASoC: omap: mcbsp, mcpdm, dmic: raw read and write endian fix Wei Yongjun (1): ASoC: tegra20-ac97: add missing clk_disable_unprepare() on error path Xiubo Li (16): ASoC: dmaengine: Add config parameter NULL check. ASoC: Add SAI SoC Digital Audio Interface driver. ASoC: fsl-sai: Add device tree bindings for Freescale SAI. ASoC: simple-card: Use devm_snd_soc_register_card() ASoC: fsl-sai: Remove fsl_sai_remove() ASoC: fsl-sai: Use devm_snd_dmaengine_pcm_register() ASoC: fsl-sai: Use snd_soc_dai_init_dma_data() ASoC: simple-card: Add cpu_dai and codec_dai names NULL check ASoC: fsl_sai: Move the global registers setting to _dai_probe() ASoC: fsl_sai: Add disable operation for the corresponding data channel. ASoC: fsl_sai: Fix one bug for hardware limitation. ASoC: fsl_sai: fix the endianess for SAI fifo data. ASoC: simple-card: Add DAPM routes parse from device tree ASoC: simple-card: Cleanup __asoc_simple_card_dai_init() ret check ASoC: core: Fix the DAI name getting. ASoC: simple-card: Add device's module clock selection. wangbiao (1): ASoC: wm8994: Move DCS done IRQ request later .../devicetree/bindings/sound/adi,axi-i2s.txt | 31 + .../devicetree/bindings/sound/adi,axi-spdif-tx.txt | 30 + .../devicetree/bindings/sound/bcm2835-i2s.txt | 25 + .../devicetree/bindings/sound/cs42l52.txt | 46 + .../bindings/sound/davinci-mcasp-audio.txt | 6 +- .../devicetree/bindings/sound/fsl-sai.txt | 40 + Documentation/devicetree/bindings/sound/hdmi.txt | 17 + .../devicetree/bindings/sound/max98090.txt | 43 + .../bindings/sound/nvidia,tegra-audio-max98090.txt | 51 + .../devicetree/bindings/sound/simple-card.txt | 77 ++ arch/arm/Kconfig | 1 + arch/arm/mach-s3c64xx/Kconfig | 7 +- arch/arm/mach-s3c64xx/Makefile | 2 +- arch/arm/mach-s3c64xx/common.h | 5 + arch/arm/mach-s3c64xx/dma.c | 762 --------------- arch/arm/mach-s3c64xx/include/mach/dma.h | 144 +-- arch/arm/mach-s3c64xx/pl080.c | 244 +++++ arch/arm/plat-samsung/devs.c | 10 +- arch/arm/plat-samsung/dma-ops.c | 8 +- drivers/clk/samsung/clk-s3c64xx.c | 4 +- drivers/dma/dmaengine.c | 35 +- drivers/dma/of-dma.c | 15 +- drivers/mfd/twl6040.c | 92 +- drivers/mfd/wm5110-tables.c | 139 ++- drivers/spi/Kconfig | 2 +- include/linux/dmaengine.h | 8 + include/linux/mfd/arizona/registers.h | 162 ++++ include/linux/platform_data/asoc-ti-mcbsp.h | 6 +- include/linux/platform_data/davinci_asp.h | 1 + include/sound/cs42l52.h | 14 +- include/sound/dmaengine_pcm.h | 10 + include/sound/pcm_params.h | 12 + include/sound/rcar_snd.h | 3 +- include/sound/soc-dai.h | 6 + include/sound/soc.h | 17 +- include/sound/spear_dma.h | 1 - sound/soc/Kconfig | 4 +- sound/soc/Makefile | 4 +- sound/soc/adi/Kconfig | 21 + sound/soc/adi/Makefile | 5 + sound/soc/adi/axi-i2s.c | 277 ++++++ sound/soc/adi/axi-spdif.c | 272 ++++++ sound/soc/atmel/sam9x5_wm8731.c | 2 - sound/soc/au1x/dbdma2.c | 9 - sound/soc/au1x/dma.c | 14 - sound/soc/bcm/Kconfig | 10 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/bcm2835-i2s.c | 879 +++++++++++++++++ sound/soc/blackfin/bf5xx-ac97-pcm.c | 1 - sound/soc/blackfin/bf5xx-i2s-pcm.c | 3 - sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 20 +- sound/soc/cirrus/ep93xx-i2s.c | 22 +- sound/soc/cirrus/ep93xx-pcm.c | 48 +- sound/soc/cirrus/ep93xx-pcm.h | 22 + sound/soc/cirrus/simone.c | 2 +- sound/soc/cirrus/snappercl15.c | 2 +- sound/soc/codecs/Kconfig | 2 + sound/soc/codecs/ad1836.c | 2 + sound/soc/codecs/ad193x.c | 6 +- sound/soc/codecs/adav80x.c | 6 +- sound/soc/codecs/ak4641.c | 38 +- sound/soc/codecs/ak4642.c | 138 ++- sound/soc/codecs/arizona.c | 204 ++-- sound/soc/codecs/arizona.h | 5 +- sound/soc/codecs/cs4271.c | 8 +- sound/soc/codecs/cs42l52.c | 105 +- sound/soc/codecs/da7210.c | 6 +- sound/soc/codecs/hdmi.c | 12 +- sound/soc/codecs/sgtl5000.c | 76 +- sound/soc/codecs/ssm2518.c | 8 +- sound/soc/codecs/ssm2602.c | 56 +- sound/soc/codecs/tlv320aic3x.c | 126 ++- sound/soc/codecs/twl6040.c | 227 ++--- sound/soc/codecs/uda1380.c | 6 +- sound/soc/codecs/wm5102.c | 4 +- sound/soc/codecs/wm5110.c | 379 +++++++- sound/soc/codecs/wm8510.c | 6 +- sound/soc/codecs/wm8523.c | 6 +- sound/soc/codecs/wm8580.c | 6 +- sound/soc/codecs/wm8711.c | 6 +- sound/soc/codecs/wm8728.c | 6 +- sound/soc/codecs/wm8731.c | 6 +- sound/soc/codecs/wm8741.c | 6 +- sound/soc/codecs/wm8750.c | 6 +- sound/soc/codecs/wm8753.c | 6 +- sound/soc/codecs/wm8776.c | 6 +- sound/soc/codecs/wm8804.c | 6 +- sound/soc/codecs/wm8900.c | 6 +- sound/soc/codecs/wm8940.c | 221 +++-- sound/soc/codecs/wm8962.c | 4 +- sound/soc/codecs/wm8974.c | 54 +- sound/soc/codecs/wm8985.c | 6 +- sound/soc/codecs/wm8988.c | 6 +- sound/soc/codecs/wm8990.c | 258 ++--- sound/soc/codecs/wm8990.h | 9 - sound/soc/codecs/wm8991.c | 293 +++--- sound/soc/codecs/wm8991.h | 9 - sound/soc/codecs/wm8994.c | 11 +- sound/soc/codecs/wm8995.c | 6 +- sound/soc/codecs/wm8997.c | 4 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm_adsp.c | 20 +- sound/soc/davinci/Kconfig | 29 +- sound/soc/davinci/Makefile | 8 +- sound/soc/davinci/davinci-evm.c | 48 +- sound/soc/davinci/davinci-mcasp.c | 1016 +++++++++----------- sound/soc/davinci/davinci-mcasp.h | 304 +++++- sound/soc/davinci/davinci-pcm.c | 28 - sound/soc/fsl/Kconfig | 5 +- sound/soc/fsl/Makefile | 4 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_sai.c | 460 +++++++++ sound/soc/fsl/fsl_sai.h | 114 +++ sound/soc/fsl/fsl_spdif.c | 8 - sound/soc/fsl/fsl_ssi.c | 435 ++++++--- sound/soc/fsl/fsl_ssi.h | 2 + sound/soc/fsl/imx-pcm-dma.c | 9 +- sound/soc/fsl/imx-pcm.h | 5 - sound/soc/fsl/imx-spdif.c | 85 +- sound/soc/fsl/imx-ssi.c | 12 +- sound/soc/fsl/imx-ssi.h | 1 - sound/soc/generic/simple-card.c | 193 +++- sound/soc/{mid-x86 => intel}/Kconfig | 0 sound/soc/{mid-x86 => intel}/Makefile | 0 sound/soc/{mid-x86 => intel}/mfld_machine.c | 0 sound/soc/{mid-x86 => intel}/sst_dsp.h | 0 sound/soc/{mid-x86 => intel}/sst_platform.c | 0 sound/soc/{mid-x86 => intel}/sst_platform.h | 0 sound/soc/jz4740/Kconfig | 1 + sound/soc/jz4740/jz4740-i2s.c | 135 +-- sound/soc/jz4740/jz4740-pcm.c | 358 ------- sound/soc/jz4740/jz4740-pcm.h | 20 - sound/soc/jz4740/qi_lb60.c | 2 +- sound/soc/mxs/mxs-pcm.c | 8 +- sound/soc/mxs/mxs-pcm.h | 1 - sound/soc/mxs/mxs-saif.c | 18 +- sound/soc/omap/mcbsp.c | 12 +- sound/soc/omap/omap-dmic.c | 4 +- sound/soc/omap/omap-mcpdm.c | 4 +- sound/soc/omap/omap-pcm.c | 2 - sound/soc/pxa/Kconfig | 2 +- sound/soc/pxa/mmp-pcm.c | 24 +- sound/soc/s6000/s6000-pcm.c | 7 - sound/soc/samsung/Kconfig | 13 +- sound/soc/samsung/Makefile | 6 +- sound/soc/samsung/ac97.c | 51 +- sound/soc/samsung/dma.c | 14 +- sound/soc/samsung/dma.h | 6 + sound/soc/samsung/dmaengine.c | 84 ++ sound/soc/samsung/i2s.c | 9 +- sound/soc/samsung/idma.c | 8 - sound/soc/samsung/pcm.c | 18 +- sound/soc/samsung/regs-ac97.h | 9 +- sound/soc/samsung/regs-iis.h | 9 +- sound/soc/sh/fsi.c | 36 +- sound/soc/sh/rcar/adg.c | 151 ++- sound/soc/sh/rcar/gen.c | 256 +++-- sound/soc/sh/rcar/rsnd.h | 46 +- sound/soc/sh/rcar/scu.c | 181 +++- sound/soc/sh/rcar/ssi.c | 19 +- sound/soc/soc-core.c | 10 +- sound/soc/soc-dapm.c | 28 +- sound/soc/soc-devres.c | 41 + sound/soc/soc-generic-dmaengine-pcm.c | 56 +- sound/soc/soc-io.c | 4 +- sound/soc/soc-pcm.c | 156 ++- sound/soc/soc-utils.c | 4 - sound/soc/spear/spdif_in.c | 17 +- sound/soc/spear/spdif_out.c | 18 +- sound/soc/spear/spear_pcm.c | 42 +- sound/soc/spear/spear_pcm.h | 24 + sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra20_ac97.c | 6 +- sound/soc/tegra/tegra_max98090.c | 275 ++++++ sound/soc/tegra/tegra_pcm.c | 3 - sound/soc/tegra/tegra_wm9712.c | 1 - sound/soc/txx9/txx9aclc.c | 5 - 179 files changed, 7232 insertions(+), 3841 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,axi-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt create mode 100644 Documentation/devicetree/bindings/sound/bcm2835-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/cs42l52.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl-sai.txt create mode 100644 Documentation/devicetree/bindings/sound/hdmi.txt create mode 100644 Documentation/devicetree/bindings/sound/max98090.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-card.txt delete mode 100644 arch/arm/mach-s3c64xx/dma.c create mode 100644 arch/arm/mach-s3c64xx/pl080.c create mode 100644 sound/soc/adi/Kconfig create mode 100644 sound/soc/adi/Makefile create mode 100644 sound/soc/adi/axi-i2s.c create mode 100644 sound/soc/adi/axi-spdif.c create mode 100644 sound/soc/bcm/Kconfig create mode 100644 sound/soc/bcm/Makefile create mode 100644 sound/soc/bcm/bcm2835-i2s.c create mode 100644 sound/soc/cirrus/ep93xx-pcm.h create mode 100644 sound/soc/fsl/fsl_sai.c create mode 100644 sound/soc/fsl/fsl_sai.h rename sound/soc/{mid-x86 => intel}/Kconfig (100%) rename sound/soc/{mid-x86 => intel}/Makefile (100%) rename sound/soc/{mid-x86 => intel}/mfld_machine.c (100%) rename sound/soc/{mid-x86 => intel}/sst_dsp.h (100%) rename sound/soc/{mid-x86 => intel}/sst_platform.c (100%) rename sound/soc/{mid-x86 => intel}/sst_platform.h (100%) delete mode 100644 sound/soc/jz4740/jz4740-pcm.c delete mode 100644 sound/soc/jz4740/jz4740-pcm.h create mode 100644 sound/soc/samsung/dmaengine.c create mode 100644 sound/soc/spear/spear_pcm.h create mode 100644 sound/soc/tegra/tegra_max98090.c

2 1

[alsa-devel] MOTU Micro Lite (5x5 MIDI USB Interface)
by Studio65 05 Jan '14

05 Jan '14

Hello Alsa Team, may I try to explain my/our problem as short as possible. We have been truing to hack a kernel patch to get a MOTU Micro Lite (5x5 MIDI USB Interface) to run within a Debian "wheezy" with RT Kernel. We did get the five channels appear in #qjackctl or #patchage. we tried to receive input from #vkeybd and get output to #gmidimonitor while on the Hardware we connected: MOTU Midi In 1 <--> Motu Midi Out 1 MOTU Midi In 2 <--> Motu Midi Out 2 MOTU Midi In 3 <--> Motu Midi Out 3 MOTU Midi In 4 <--> Motu Midi Out 4 MOTU Midi In 5 <--> Motu Midi Out 5 Here is some Logs: Output of #tail -f /var/log/messages http://pastebin.com/R6xppq7k Output of #lsusb -vv http://pastebin.com/Du0N6hCa Cheers, Nathanael Studio65

1 0

[alsa-devel] disconnect between ported driver (kernel) and alplay (userspace)
by Anthony Tzouris 03 Jan '14

03 Jan '14

_Hardware/Software Platform_ * Custom Hardware based on Freescale i.MX28EVK platform * Original Codec Freescale SGTL5000 on Freescale i.MX28EVK platform * Ported a Codec Texas Instruments PCM1681 * Embedded Linux using Freescale LTIB 2013 (linux kernel 2.6.35.3) distribution _Good News_ I have the I2S clocks verified with digital scope on the hardware for the desired 256 * fs, 64 * fs, and fs. _Bad News_ Audio when played out with aplay and a 44.1 kHz S16_LE Stereo file plays out too fast (guesstimating about 3 times too fast). _Due Diligence_ I've been digging through the linux kernel code to figure out why my driver is passing the incorrect fs (sample rate) up to the linux kernel alsa subsystem between aplay and my pcm1681 device driver. I've forced the driver to only run 44.1kHz (which shows up everwhere from my kernel debug printk) and that is what is showing up with a scope on the hardware fs clock with a digital scope but ... between aplay and the driver, there is a perception of a lower sample rate needed, therefore there is a rate reduction of the sample sequence which makes the audio clip play faster out on the audio port. I've been digging through the code and trying different things for a few days now and I have not made any progress. _Help/Advice/Understanding Needed_ What I need to understand is how from aplay down through the alsa layer in the kernel the sample rate is set down to the driver layer ... and in the reverse direction. Of course the kernel is patched by Freescale so I'm not exactly sure how to ask for help on this but perhaps there is a way to get some ideas from the aplay (userspace layer) down through the alsa kernel layer that I can look at that might downsample the audio file before it gets to the driver. Using alsa-lib-1.0.24.1 alsa-utils-1.0.24.2 My idea is to approach debug from the userspace down versus the driver porting which was from the lowest level (driver layer) and up. That is why I'm posting on alsa-devel as I think that this is where I need to focus where the sample rate misconception is and thereby where the downsampling is occurring which is either in userspace (alsa-lib) code or possibly alsa layer kernel code. Looking to leverage some alsa knowledge and expertise. -- Anthony

1 0

[alsa-devel] CA0132 Issues with Alienware
by Antonio Vázquez Blanco 03 Jan '14

03 Jan '14

Dear Alsa developing members, I'm sorry to contact you via this mailing list even if I don't know if it used for this, but I felt I had no other option when after various attempts I saw the bug tracker was down and various members of the project didn't answer my emails. My computer has three jack ports on one side (Microphone, headphone and earphones). The problem is that Alsa currently doesn't have jack detection on earphones jack plug, so I'm stuck with using only left audio on the headphone jack. I would really like to be able to use earphones plug which in my hardware is node 0x0f. Although I know C programming I don't have the knowledge to do this myself. Thanks in advance. Your sincerely, Antonio Vázquez Blanco. -- If comunications require exchange of sensible data, please consider using my PGP public key<https://drive.google.com/file/d/0Bw0Vql7Olv4LMVlrVlJNMkVjUUU/edit?usp=shari…>for data encryption. Si la comunicación requiere el intercambio de información sensible, por favor, considere usar mi clave PGP pública<https://drive.google.com/file/d/0Bw0Vql7Olv4LMVlrVlJNMkVjUUU/edit?usp=shari…>para encriptar los datos.

1 0

[alsa-devel] /dev/snd/timer failing on Raspberry Pi
by David Preece 02 Jan '14

02 Jan '14

Hi, As per subject, the dmix plugin is failing on Arch and a Raspberry Pi with a poll blocking around the file descriptor for /dev/snd/timer. Could it be related to the RPi not having a real time clock? There are copious details on the RPi forums... http://www.raspberrypi.org/forum/viewtopic.php?f=53&t=64936 Thanks, Dave

1 0

[alsa-devel] [PATCH] ASoC: generic: add generic compound card with DT support
by Jean-Francois Moine 02 Jan '14

02 Jan '14

Some audio cards are built from different hardware components. When such compound cards don't need specific code, this driver creates them with the required DAI links and routes from a DT. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- This code was first developped on the generic simple card, but its recent DT extension cannot be easily extended again to support compound cards as the one in the Cubox. Note also that the example relies on a proposed patch of mine aiming to render the codec name / OF node optional in DAI links (http://mailman.alsa-project.org/pipermail/alsa-devel/2013-December/070082.h…). --- .../devicetree/bindings/sound/compound-card.txt | 95 ++++++++++++ sound/soc/generic/Kconfig | 6 + sound/soc/generic/Makefile | 2 + sound/soc/generic/compound-card.c | 247 +++++++++++++++++++++++++ 4 file changed, 350 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/compound-card.txt b/Documentation/devicetree/bindings/sound/compound-card.txt new file mode 100644 index 0000000..554a796 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/compound-card.txt @@ -0,0 +1,95 @@ +Device-Tree bindings for compound audio card + +Compound audio card describes the links between the different parts +of an audio card built from different hardware components. + +Required properties: + - compatible: should be "compound-audio-card" + - audio-controller: phandle of the audio controller + +Optional properties: + - routes: list of couple of strings (sink, source) + +Required subnodes: + - link: DAI link subnode + At least one link must be specified. + +Required link subnode properties: + - link-name: names of the DAI link and of the stream + - cpu-dai-name: name of the CPU or CODEC DAI + An empty string indicates that the CPU DAI is + the same as the audio controller. + - codec-dai-name: name of the CODEC DAI + +Optional link subnode properties: + - audio-codec or codec-name: phandle or name of the CODEC + in case the codec-dai-name is not unique + - format: DAI format. One of: + "i2s", "right_j", "left_j" , "dsp_a" + "dsp_b", "ac97", "pdm", "msb", "lsb" + - front-end or back-end: present if the DAI link describes resp. + a front-end CPU DAI or a back-end CODEC DAI + - playback or capture: present if the DAI link is used for + playback or capture only + +Example node: + +The audio subsystem found in the SolidRun Cubox is composed of: + +- an audio controller which outputs playback streams on either one or both + of I2S and S/PDIF, + +- a HDMI transmitter which can get its input from either I2S or S/PDIF, + +- an optical S/PDIF connector. + + sound { + compatible = "compound-audio-card"; + audio-controller = <&audio1>; + routes = + "I2S Playback", "System Playback", + "SPDIF Playback", "System Playback", + "hdmi-out", "I2S Playback", + "hdmi-out", "SPDIF Playback", + "spdif-out", "SPDIF Playback"; + link@0 { + link-name = "System audio"; + front-end; + cpu-dai-name = "mvebu-audio"; + codec-dai-name = "snd-soc-dummy-dai"; + format = "i2s"; + playback; + }; + link@1 { + link-name = "hdmi-i2s"; + back-end; + cpu-dai-name = "i2s"; + codec-dai-name = "i2s-hifi"; + playback; + }; + link@2 { + link-name = "hdmi-spdif"; + back-end; + cpu-dai-name = "spdif"; + codec-dai-name = "spdif-hifi"; + playback; + }; + link@3 { + link-name = "spdif"; + back-end; + cpu-dai-name = "spdif"; + codec-dai-name = "dit-hifi"; + playback; + }; + }; + + hdmi_codec: hdmi-codec { + compatible = "nxp,tda998x-codec"; + audio-ports = <0x03>, <0x04>; + }; + + spdif_codec: spdif-codec { + compatible = "linux,spdif-dit"; + }; + + diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index 610f612..f2678ae 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -2,3 +2,9 @@ config SND_SIMPLE_CARD tristate "ASoC Simple sound card support" help This option enables generic simple sound card support + +config SND_COMPOUND_CARD + tristate "ASoC Compound sound card support" + depends on OF + help + This option enables the generic compound sound card support. diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 9c3b246..434cb79 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -1,3 +1,5 @@ snd-soc-simple-card-objs := simple-card.o +snd-soc-compound-card-objs := compound-card.o obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o +obj-$(CONFIG_SND_COMPOUND_CARD) += snd-soc-compound-card.o diff --git a/sound/soc/generic/compound-card.c b/sound/soc/generic/compound-card.c new file mode 100644 index 0000000..f57e159 --- /dev/null +++ b/sound/soc/generic/compound-card.c @@ -0,0 +1,247 @@ +/* + * ASoC compound sound card support + * + * Copyright (C) 2013 Jean-Francois Moine <moinejf(a)free.fr> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/platform_device.h> +#include <linux/module.h> +#include <linux/of.h> +#include <sound/soc.h> + +// link types +#define LINK_FE 0 /* front-end */ +#define LINK_BE 1 /* back-end */ +#define LINK_CC 2 /* (controller or codec) to codec */ + +struct compound_card_data { + struct snd_soc_card snd_card; + struct snd_soc_dai_link *snd_links; +}; + +/* get a DAI link */ +static int get_link(struct compound_card_data *priv, + struct snd_soc_dai_link *link, + struct device_node *ctrl_np, + struct device_node *np) +{ + struct device_node *codec_np; + const char *cpu_dai_name; + int type, ret; + + /* optional link-name */ + of_property_read_string(np, "link-name", &link->name); + link->stream_name = link->name; + + /* link type */ + if (of_property_read_bool(np, "front-end")) + type = LINK_FE; + else if (of_property_read_bool(np, "back-end")) + type = LINK_BE; + else + type = LINK_CC; + + /* cpu-dai-name */ + ret = of_property_read_string(np, "cpu-dai-name", + &cpu_dai_name); + if (ret) { + dev_err(priv->snd_card.dev, + "No valid cpu-dai-name found\n"); + return ret; + } + + /* an empty cpu-dai-name forces the audio-controller */ + if (cpu_dai_name[0] == '\0') + link->cpu_of_node = ctrl_np; + else + link->cpu_dai_name = cpu_dai_name; + + /* the codec may be omitted or specified by name or phandle */ + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (codec_np) + link->codec_of_node = codec_np; + else + of_property_read_string(np, "codec-name", + &link->codec_name); + + /* codec-dai-name */ + ret = of_property_read_string(np, "codec-dai-name", + &link->codec_dai_name); + if (ret) { + dev_err(priv->snd_card.dev, + "Failed to parse codec-dai-name\n"); + return ret; + } + + if (type == LINK_BE) { + link->no_pcm = 1; + /* don't set the platform, otherwise, the platform functions + * are called many times */ + } else { + link->platform_of_node = ctrl_np; + if (type == LINK_FE) + link->dynamic = 1; + } + + /* DAI format */ + link->dai_fmt = snd_soc_of_parse_daifmt(np, NULL) & + SND_SOC_DAIFMT_FORMAT_MASK; + + if (of_property_read_bool(np, "playback")) + link->playback_only = 1; + else if (of_property_read_bool(np, "capture")) + link->capture_only = 1; + + return 0; +} + +static int compound_card_dt_probe(struct platform_device *pdev, + struct compound_card_data *priv) +{ + struct device_node *np, *ctrl_np, *link_np; + struct snd_soc_dai_link *link; + int ret, num_links; + + np = pdev->dev.of_node; + + ctrl_np = of_parse_phandle(np, "audio-controller", 0); + if (!ctrl_np) { + dev_err(&pdev->dev, "No valid audio-controller phandle found\n"); + return -EINVAL; + } + + /* get the optional routes */ + if (of_property_count_strings(np, "routes")) + snd_soc_of_parse_audio_routing(&priv->snd_card, "routes"); + + priv->snd_card.name = "ASoC compound sound card"; + + /* check the number of links */ + num_links = 0; + for_each_child_of_node(np, link_np) + num_links++; + + if (num_links == 0) { + dev_err(&pdev->dev, "No DAI link\n"); + return -EINVAL; + } + + link = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_dai_link) * num_links, + GFP_KERNEL); + if (!link) { + ret = -ENOMEM; + goto err; + } + priv->snd_links = link; + priv->snd_card.num_links = num_links; + for_each_child_of_node(np, link_np) { + ret = get_link(priv, link, ctrl_np, link_np); + if (ret) + goto err; + link++; + } + + return 0; + +err: + if (link_np) + of_node_put(link_np); + if (ctrl_np) + of_node_put(ctrl_np); + for (num_links = 0, link = priv->snd_links; + num_links < priv->snd_card.num_links; + num_links++, link++) { + struct device_node *codec_np; + + codec_np = (struct device_node *) link->codec_of_node; + if (!codec_np) + break; + of_node_put(codec_np); + } + return ret; +} + +static int compound_card_dt_remove(struct platform_device *pdev, + struct compound_card_data *priv) +{ + struct device_node *codec_np; + struct snd_soc_dai_link *link; + int num_links; + + if (!priv->snd_links) + return 0; + if (priv->snd_links->platform_of_node) + of_node_put((struct device_node *) + priv->snd_links->platform_of_node); + for (num_links = 0, link = priv->snd_links; + num_links < priv->snd_card.num_links; + num_links++, link++) { + codec_np = (struct device_node *) link->codec_of_node; + if (!codec_np) + break; + of_node_put(codec_np); + } + return 0; +} + +static int compound_card_probe(struct platform_device *pdev) +{ + struct compound_card_data *priv; + int ret; + + priv = devm_kzalloc(&pdev->dev, sizeof *priv, GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->snd_card.owner = THIS_MODULE; + priv->snd_card.dev = &pdev->dev; + + ret = compound_card_dt_probe(pdev, priv); + + if (ret) + return ret; + + priv->snd_card.dai_link = priv->snd_links; + + dev_set_drvdata(&pdev->dev, priv); + + return snd_soc_register_card(&priv->snd_card); +} + +static int compound_card_remove(struct platform_device *pdev) +{ + struct compound_card_data *priv = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_card(&priv->snd_card); + + compound_card_dt_remove(pdev, priv); + + return 0; +} + +static const struct of_device_id compound_card_of_dev_ids[] = { + { .compatible = "compound-audio-card", }, + {} +}; +MODULE_DEVICE_TABLE(of, compound_card_of_dev_ids); + +static struct platform_driver compound_card = { + .driver = { + .name = "asoc-compound-card", + .owner = THIS_MODULE, + .of_match_table = compound_card_of_dev_ids, + }, + .probe = compound_card_probe, + .remove = compound_card_remove, +}; + +module_platform_driver(compound_card); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("ASoC Compound Sound Card"); +MODULE_AUTHOR("Jean-Francois Moine <moinejf(a)free.fr>"); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

3 14

[alsa-devel] [PATCH] ASoC: ssm2518: Fix off-by-one error by ffs()
by Takashi Iwai 02 Jan '14

02 Jan '14

ffs() returns the bit position from 1, while the ssm2158 driver code assumes it being 0-based. Also, the bit mask computation of the two channel slots are incorrect; it must have worked just casually. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- I found it casully while checking ab8500 bug. The patch is untested, so please review/test first, and give Cc to stable if needed. sound/soc/codecs/ssm2518.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 95aed552139a..cc8debce752f 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -549,13 +549,13 @@ static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, right_slot = 0; } else { /* We assume the left channel < right channel */ - left_slot = ffs(tx_mask); - tx_mask &= ~(1 << tx_mask); + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); if (tx_mask == 0) { right_slot = left_slot; } else { - right_slot = ffs(tx_mask); - tx_mask &= ~(1 << tx_mask); + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); } } -- 1.8.5.2

3 2

Re: [alsa-devel] [PATCH 08/19] ASoC: kirkwood: Don't set unused struct snd_pcm_hardware fields
by Jean-Francois Moine 02 Jan '14

02 Jan '14

> On 12/20/2013 08:13 PM, Jean-Francois Moine wrote: > > On Fri, 20 Dec 2013 18:18:49 +0100 > > Lars-Peter Clausen <lars(a)metafoo.de> wrote: > > > >> On 12/20/2013 07:05 PM, Jean-Francois Moine wrote: > >>>> diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c > >>>> index 4af1936..aac22fc 100644 > >>>> --- a/sound/soc/kirkwood/kirkwood-dma.c > >>>> +++ b/sound/soc/kirkwood/kirkwood-dma.c > >>> [snip] > >>>> @@ -43,12 +33,6 @@ static struct snd_pcm_hardware kirkwood_dma_snd_hw = { > >>>> SNDRV_PCM_INFO_MMAP_VALID | > >>>> SNDRV_PCM_INFO_BLOCK_TRANSFER | > >>>> SNDRV_PCM_INFO_PAUSE), > >>>> - .formats = KIRKWOOD_FORMATS, > >>>> - .rates = KIRKWOOD_RATES, > >>>> - .rate_min = 8000, > >>>> - .rate_max = 384000, > >>>> - .channels_min = 1, > >>>> - .channels_max = 8, > >>>> .buffer_bytes_max = KIRKWOOD_SND_MAX_BUFFER_BYTES, > >>>> .period_bytes_min = KIRKWOOD_SND_MIN_PERIOD_BYTES, > >>>> .period_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES, > >>> > >>> Lars, > >>> > >>> You removed too many things. The 'formats' field is mandatory. > >> > >> No it is not. While snd_soc_set_runtime_hwparams() uses it it is later > >> overwritten again in soc_pcm_init_runtime_hw(). > > > > I have a DPCM system and soc_pcm_init_runtime_hw() is not called > > (either 'dynamic' or 'no_pcm' is set in the DAI links). > > Ok, I see dpcm_set_fe_runtime() does things slightly different. Well, I advanced a bit with DPCM, and I have a problem with this field. In the driver, the front-end DAI is the audio controller and the back-ends DAIs are the HDMI and SPDIF outputs. These back-end DAIs have different rates and formats, as have the audio controller outputs (I2S and SPDIF). So, I used intermediate DAIs which represent the audio controller outputs, and they are described in the DAI links: link 0 (FE): audio controller <-> dummy DAI link 1 (BE): i2s audio controller output <-> HDMI output link 2 (BE): spdif audio controller output <-> HDMI output link 3 (BE): spdif audio controller output <-> SPDIF output Without any patch in the core, the rates and formats are always the rates and formats of the audio controller (FE). This is due to the 'goto dynamic' in soc_pcm_open(): as the back-ends are linked to real DAIs, the rate and format constraints must be checked. So, as a temporary patch, I replaced: if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) goto dynamic; by: if (rtd->dai_link->dynamic) goto dynamic; (indeed, this will not work if the back-end is linked to the dummy DAI) and, I get the correct rates and formats in the runtime hardware parameters. But these values are lost: - on DMA open (FE pcm open), when the driver calls snd_soc_set_runtime_hwparams() (loss of the formats), and - on dpcm_set_fe_runtime() call (loss of the rates). The first problem can be fixed in the audio controller by a hack, saving /restoring the formats on calling snd_soc_set_runtime_hwparams(), and the second problem is easily fixed moving dpcm_set_fe_runtime() at the beginning of dpcm_fe_dai_startup(). Are these good solutions? -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

3 2

[alsa-devel] hda problem on Gigabyte GA-Z87X-UD3H motherboard
by Guillaume Desclaux 02 Jan '14

02 Jan '14

Hi, I've reported an issue with snd-hda-intel with my Gigabyte GA-Z87X-UD3H motherboard on Launchpad but it seems not the best place to that See :https://bugs.launchpad.net/bugs/1248116 In "Volume control PulseAudio", I can see pulseaudio changing the audio output "Internal Audio Analogic Stereo". The port switch rapidly from "Analogic output" to "Analogic headphones" making a jerky sound. You can find the trace.log file here : https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1248116/+attachme… It seems that there are many hda_power_count 1/ This problem appears with the front connector on the motherboard connected to a HDA front panel on the case (I've double-checked the connection) 2/ The problem still appears when the front connector on the motherboard is not connected at all to the case ! 3/ This bug seems quite common with Gigabyte motherboard : If you search with Google "Gigabyte Z87X hda" you will find other descriptions of this same strange behavior : https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1242302 https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1067434 If I disable jack detection by specifying a hint for early patching (jack_detect = no), the sound is cristal clear ! If there is anything I can do to help you understand what happens, don't hesitate to ask. Best regards Guillaume

2 2

[alsa-devel] [PATCH] ASoC: simple-card: cleanup asoc_simple_card_parse_of() code
by Xiubo Li 02 Jan '14

02 Jan '14

Since there is only one common format via "simple-audio-card,format" for simple card driver(CPU & CODEC DAI), there is no need to do the INV mask. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d340320..8189e68 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -135,7 +135,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, /* get CPU/CODEC common format via simple-audio-card,format */ info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & - (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); + SND_SOC_DAIFMT_FORMAT_MASK; ret = asoc_simple_card_get_component(node, info); if (ret) -- 1.8.4

3 6

[alsa-devel] [PATCH] ASoC: simple-card: Add DAPM routes parse from device tree
by Xiubo Li 02 Jan '14

02 Jan '14

Parses a simple DAPM route table from device tree. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Documentation/devicetree/bindings/sound/simple-card.txt | 13 +++++++++++-- sound/soc/generic/simple-card.c | 11 +++++++++-- 2 files changed, 20 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index 769a346..2ee80c76 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -9,8 +9,13 @@ Required properties: Optional properties: - simple-audio-card,format : CPU/CODEC common audio format. - "i2s", "right_j", "left_j" , "dsp_a" - "dsp_b", "ac97", "pdm", "msb", "lsb" + "i2s", "right_j", "left_j" , "dsp_a" + "dsp_b", "ac97", "pdm", "msb", "lsb" +- simple-audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. + Required subnodes: - simple-audio-card,cpu : CPU sub-node @@ -38,6 +43,10 @@ Example: sound { compatible = "simple-audio-card"; simple-audio-card,format = "left_j"; + simple-audio-routing = + "MIC_IN", "Mic Jack", + "Headphone Jack", "HP_OUT", + "Ext Spk", "LINE_OUT"; simple-audio-card,cpu { sound-dai = <&sh_fsi2 0>; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 1d87db8..643c9ed 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -116,12 +116,18 @@ static int asoc_simple_card_parse_of(struct device_node *node, { struct device_node *np; char *name; - int ret = 0; + int ret; /* get CPU/CODEC common format via simple-audio-card,format */ info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); + /* DAPM routes */ + ret = snd_soc_of_parse_audio_routing(&info->snd_card, + "simple-audio-routing"); + if (ret) + return ret; + /* CPU sub-node */ ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,cpu"); @@ -185,6 +191,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); if (cinfo) { int ret; + cinfo->snd_card.dev = &pdev->dev; ret = asoc_simple_card_parse_of(np, cinfo, dev, &of_cpu, &of_codec, @@ -196,6 +203,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } } } else { + cinfo->snd_card.dev = &pdev->dev; cinfo = pdev->dev.platform_data; } @@ -235,7 +243,6 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo->snd_card.owner = THIS_MODULE; cinfo->snd_card.dai_link = &cinfo->snd_link; cinfo->snd_card.num_links = 1; - cinfo->snd_card.dev = &pdev->dev; return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); } -- 1.8.4

3 2

[alsa-devel] [PATCH] ASoC: add snd_soc_of_parse_master_clkdir() for dt
by Xiubo Li 02 Jan '14

02 Jan '14

This patch adds snd_soc_of_parse_master_clkdir() and supports below style on dt: [prefix]master-clkdir; If this property is absent 0 will be returned, 1 otherwise. 0 : SND_SOC_CLOCK_IN 1 : SND_SOC_CLOCK_OUT Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- I'm not very sure whether this patch is need for dt. For the CPU and CODEC devices' sysclk directions maybe different of the same DAI link. include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 19 +++++++++++++++++++ 2 files changed, 21 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5a049d9..1d6396a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1170,6 +1170,8 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname); int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); +unsigned int snd_soc_of_parse_master_clkdir(struct device_node *np, + const char *prefix); unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix); int snd_soc_of_get_dai_name(struct device_node *of_node, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index be88df5..faf102f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4472,6 +4472,25 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_routing); +unsigned int snd_soc_of_parse_master_clkdir(struct device_node *np, + const char *prefix) +{ + char prop[128]; + + if (!prefix) + prefix = ""; + + /* + * check "[prefix]master-clkdir" + * true means SND_SOC_CLOCK_OUT + * false means SND_SOC_CLOCK_IN + */ + snprintf(prop, sizeof(prop), "%smaster-clkdir", prefix); + + return !!of_property_read_bool(np, prop); +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_master_clkdir); + unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix) { -- 1.8.4

3 2

[alsa-devel] [PATCH] ASoC: simple-card: Add device's module clock selection.
by Xiubo Li 02 Jan '14

02 Jan '14

Try to get the device's module clock if the dt has no clocks and system-clock-frequency properties. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- a.out | 0 sound/soc/generic/simple-card.c | 21 ++++++++++++++++++--- 2 files changed, 18 insertions(+), 3 deletions(-) create mode 100644 a.out diff --git a/a.out b/a.out new file mode 100644 index 0000000..e69de29 diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 6ce9745..c0d92813 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -90,14 +90,29 @@ asoc_simple_card_sub_parse_of(struct device_node *np, * dai->sysclk come from * "clocks = <&xxx>" (if system has common clock) * or "system-clock-frequency = <xxx>" + * or device's module clock. */ - clk = of_clk_get(np, 0); - if (IS_ERR(clk)) + if (of_property_read_bool(np, "clocks")) { + clk = of_clk_get(np, 0); + if (IS_ERR(clk)) { + ret = PTR_ERR(clk); + goto parse_error; + } + + dai->sysclk = clk_get_rate(clk); + } else if (of_property_read_bool(np, "system-clock-frequency")) { of_property_read_u32(np, "system-clock-frequency", &dai->sysclk); - else + } else { + clk = of_clk_get(*node, 0); + if (IS_ERR(clk)) { + ret = PTR_ERR(clk); + goto parse_error; + } + dai->sysclk = clk_get_rate(clk); + } ret = 0; -- 1.8.4

4 5

[alsa-devel] [PATCH] ASoC: ad1836: Reject unsupported bit sizes
by Mark Brown 01 Jan '14

01 Jan '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ad1836.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 9a92b79..d7c9838 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -179,6 +179,8 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S32_LE: word_len = AD1836_WORD_LEN_24; break; + default: + return -EINVAL; } regmap_update_bits(ad1836->regmap, AD1836_DAC_CTRL1, -- 1.8.5.2

2 1

[alsa-devel] [PATCH] ASoC: make snd_soc_dai_link more symmetrical
by Jean-Francois Moine 01 Jan '14

01 Jan '14

This patch renders optional the CODEC name / OF node on sound card register, as it is done for the CPU side, in the case the CODEC DAI name is unique. This simplifies the definition of CODECs with multiple DAIs, especially in DTs, while keeping compatibility. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- include/sound/soc.h | 6 ++++-- sound/soc/soc-core.c | 73 ++++++++++++++++++++++++++++++++++++++++---------------------- 2 files changed, 51 insertions(+), 28 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 1f741cb..67c382e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -852,8 +852,10 @@ struct snd_soc_dai_link { */ const char *cpu_dai_name; /* - * You MUST specify the link's codec, either by device name, or by - * DT/OF node, but not both. + * You MAY specify the link's codec, either by device name, or by + * DT/OF node, but not both. If this information is omitted, + * the CODEC DAI is matched using .codec_dai_name only, which hence + * must be globally unique. */ const char *codec_name; const struct device_node *codec_of_node; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e53d87..11b729c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -883,28 +883,54 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) return -EPROBE_DEFER; } - /* Find CODEC from registered CODECs */ - list_for_each_entry(codec, &codec_list, list) { - if (dai_link->codec_of_node) { - if (codec->dev->of_node != dai_link->codec_of_node) - continue; - } else { - if (strcmp(codec->name, dai_link->codec_name)) - continue; + if (dai_link->codec_of_node || dai_link->codec_name) { + + /* Find CODEC from registered CODECs */ + list_for_each_entry(codec, &codec_list, list) { + if (dai_link->codec_of_node) { + if (codec->dev->of_node != dai_link->codec_of_node) + continue; + } else { + if (strcmp(codec->name, dai_link->codec_name)) + continue; + } + + rtd->codec = codec; + + /* + * CODEC found, so find CODEC DAI from registered DAIs from + * this CODEC + */ + list_for_each_entry(codec_dai, &dai_list, list) { + if (codec->dev == codec_dai->dev && + !strcmp(codec_dai->name, + dai_link->codec_dai_name)) { + + rtd->codec_dai = codec_dai; + } + } + + if (!rtd->codec_dai) { + dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", + dai_link->codec_dai_name); + return -EPROBE_DEFER; + } } - rtd->codec = codec; + if (!rtd->codec) { + dev_err(card->dev, "ASoC: CODEC %s not registered\n", + dai_link->codec_name); + return -EPROBE_DEFER; + } + } else { - /* - * CODEC found, so find CODEC DAI from registered DAIs from - * this CODEC - */ + /* find CODEC DAI from registered DAIs */ list_for_each_entry(codec_dai, &dai_list, list) { - if (codec->dev == codec_dai->dev && - !strcmp(codec_dai->name, + if (!strcmp(codec_dai->name, dai_link->codec_dai_name)) { - rtd->codec_dai = codec_dai; + rtd->codec = codec_dai->codec; + break; } } @@ -915,12 +941,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } } - if (!rtd->codec) { - dev_err(card->dev, "ASoC: CODEC %s not registered\n", - dai_link->codec_name); - return -EPROBE_DEFER; - } - /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; if (!platform_name && !dai_link->platform_of_node) @@ -3710,12 +3730,13 @@ int snd_soc_register_card(struct snd_soc_card *card) struct snd_soc_dai_link *link = &card->dai_link[i]; /* - * Codec must be specified by 1 of name or OF node, - * not both or neither. + * Codec may be specified by either name or OF node, + * but can be left unspecified + * and will be matched based on CODEC DAI name alone. */ - if (!!link->codec_name == !!link->codec_of_node) { + if (link->codec_name && link->codec_of_node) { dev_err(card->dev, - "ASoC: Neither/both codec name/of_node are set for %s\n", + "ASoC: Both codec name/of_node are set for %s\n", link->name); return -EINVAL; } -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 3

[alsa-devel] [PATCH] ASoC: imx-spdif: Use snd-soc-dummy CODEC driver to link card
by Nicolin Chen 01 Jan '14

01 Jan '14

This is a quick fix for the below two issues when building spdif as modules. 1) If modprobing modules in order: (Step 1) snd-soc-fsl-spdif -> (Step 2) snd-soc-imx-spdif -> (Step 3) snd-soc-spdif-tx/rx, we will fail to create imx-spdif card and dai link unless we rmmod snd-soc-imx-spdif and modprobe it again due to the execution platform_driver_unregister() in probe() when meeting -EPROBE_DEFER at Step 2. 2) After "imx-spdif sound-spdif.17: dit-hifi <-> 2004000.spdif mapping ok", 'rmmod snd-soc-imx-spdif' would cause kernel dump with warning: WARNING: CPU: 0 PID: 1301 at /home/rmk/git/linux-rmk/fs/sysfs/dir.c:915 sysfs_hash_and_remove+0x84/0x90() sysfs: can not remove 'dapm_widget', no directory This should be caused by disordered resourse releasing of the whole link. And trying to unregister the card and then CODEC dev can't fix this issue. Thus this patch just provides a simple fix to these two bugs by using the snd-soc-dummy in the core instead of seperate snd-soc-spdif-tx/rx so that there's no need to handle the registering and unregistering of CODEC or CODEC dai any more. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/Kconfig | 1 - sound/soc/fsl/imx-spdif.c | 84 ++++++++++++----------------------------------- 2 files changed, 21 insertions(+), 64 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index bb4ab96..1c7b3b7 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -201,7 +201,6 @@ config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF - select SND_SOC_SPDIF select REGMAP_MMIO help SoC Audio support for i.MX boards with S/PDIF diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 980dd1f..e1dc401 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -14,17 +14,15 @@ #include <sound/soc.h> struct imx_spdif_data { - struct snd_soc_dai_link dai[2]; + struct snd_soc_dai_link dai; struct snd_soc_card card; - struct platform_device *txdev; - struct platform_device *rxdev; }; static int imx_spdif_audio_probe(struct platform_device *pdev) { struct device_node *spdif_np, *np = pdev->dev.of_node; struct imx_spdif_data *data; - int ret = 0, num_links = 0; + int ret = 0; spdif_np = of_parse_phandle(np, "spdif-controller", 0); if (!spdif_np) { @@ -39,69 +37,42 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) goto end; } - if (of_property_read_bool(np, "spdif-out")) { - data->dai[num_links].name = "S/PDIF TX"; - data->dai[num_links].stream_name = "S/PDIF PCM Playback"; - data->dai[num_links].codec_dai_name = "dit-hifi"; - data->dai[num_links].codec_name = "spdif-dit"; - data->dai[num_links].cpu_of_node = spdif_np; - data->dai[num_links].platform_of_node = spdif_np; - num_links++; - - data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0); - if (IS_ERR(data->txdev)) { - ret = PTR_ERR(data->txdev); - dev_err(&pdev->dev, "register dit failed: %d\n", ret); - goto end; - } - } + data->dai.name = "S/PDIF PCM"; + data->dai.stream_name = "S/PDIF PCM"; + data->dai.codec_dai_name = "snd-soc-dummy-dai"; + data->dai.codec_name = "snd-soc-dummy"; + data->dai.cpu_of_node = spdif_np; + data->dai.platform_of_node = spdif_np; + data->dai.playback_only = true; + data->dai.capture_only = true; - if (of_property_read_bool(np, "spdif-in")) { - data->dai[num_links].name = "S/PDIF RX"; - data->dai[num_links].stream_name = "S/PDIF PCM Capture"; - data->dai[num_links].codec_dai_name = "dir-hifi"; - data->dai[num_links].codec_name = "spdif-dir"; - data->dai[num_links].cpu_of_node = spdif_np; - data->dai[num_links].platform_of_node = spdif_np; - num_links++; - - data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0); - if (IS_ERR(data->rxdev)) { - ret = PTR_ERR(data->rxdev); - dev_err(&pdev->dev, "register dir failed: %d\n", ret); - goto error_dit; - } - } + if (of_property_read_bool(np, "spdif-out")) + data->dai.capture_only = false; + + if (of_property_read_bool(np, "spdif-in")) + data->dai.playback_only = false; - if (!num_links) { + if (data->dai.playback_only && data->dai.capture_only) { dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n"); - goto error_dir; + goto end; } data->card.dev = &pdev->dev; - data->card.num_links = num_links; - data->card.dai_link = data->dai; + data->card.dai_link = &data->dai; + data->card.num_links = 1; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto error_dir; + goto end; ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret); - goto error_dir; + goto end; } platform_set_drvdata(pdev, data); - goto end; - -error_dir: - if (data->rxdev) - platform_device_unregister(data->rxdev); -error_dit: - if (data->txdev) - platform_device_unregister(data->txdev); end: if (spdif_np) of_node_put(spdif_np); @@ -109,18 +80,6 @@ end: return ret; } -static int imx_spdif_audio_remove(struct platform_device *pdev) -{ - struct imx_spdif_data *data = platform_get_drvdata(pdev); - - if (data->rxdev) - platform_device_unregister(data->rxdev); - if (data->txdev) - platform_device_unregister(data->txdev); - - return 0; -} - static const struct of_device_id imx_spdif_dt_ids[] = { { .compatible = "fsl,imx-audio-spdif", }, { /* sentinel */ } @@ -134,7 +93,6 @@ static struct platform_driver imx_spdif_driver = { .of_match_table = imx_spdif_dt_ids, }, .probe = imx_spdif_audio_probe, - .remove = imx_spdif_audio_remove, }; module_platform_driver(imx_spdif_driver); -- 1.8.4

2 2

[alsa-devel] [PATCH] ASoC: simple-card: Add cpu_dai and codec_dai names NULL check
by Xiubo Li 31 Dec '13

31 Dec '13

The name of cpu DAI maybe omitted, and then strlen() will lead kernel panic. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 3d190d0..1d87db8 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -142,6 +142,9 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (ret < 0) return ret; + if (!info->cpu_dai.name || !info->codec_dai.name) + return -EINVAL; + /* card name is created from CPU/CODEC dai name */ name = devm_kzalloc(dev, strlen(info->cpu_dai.name) + -- 1.8.4

2 3

[alsa-devel] [PATCH] ASoC: simple-card: Cleanup __asoc_simple_card_dai_init() ret check
by Xiubo Li 31 Dec '13

31 Dec '13

The ret parameter is always equal to zero till here. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 643c9ed..6ce9745 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -25,7 +25,7 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, daifmt |= set->fmt; - if (!ret && daifmt) + if (daifmt) ret = snd_soc_dai_set_fmt(dai, daifmt); if (ret == -ENOTSUPP) { -- 1.8.4

2 1

[alsa-devel] [Patch v2 1/2] ASoC: atmel: sam9x5_wm8731: fix oops when unload module
by Bo Shen 31 Dec '13

31 Dec '13

As the priv is not assigned to card->drvdata, it is NULL, so when unload module, it will cause NULL pointer oops. Assign priv to card->drvdata to fix this issue. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Change in v2: - using snd_soc_card_set_drvdata() --- sound/soc/atmel/sam9x5_wm8731.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 992ae38..1b37228 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -97,6 +97,8 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) goto out; } + snd_soc_card_set_drvdata(card, priv); + card->dev = &pdev->dev; card->owner = THIS_MODULE; card->dai_link = dai; -- 1.7.9.5

3 6

[alsa-devel] [PATCH v2] ASoC: fsl_sai: Fix one bug for hardware limitation.
by Xiubo Li 31 Dec '13

31 Dec '13

This is maybe one bug or a limitation of the hardware that the {T,R}CR2's Synchronous Mode bits must be set as late as possible, or the SAI device maybe hanged up, and there has not any explaination about this limitation in the SAI Data Sheet. And the {T,R}CR2's Synchronous Mode bits must be set at the same time whether for Tx or Rx stream. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index af80246..2ece147 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -145,7 +145,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: val_cr4 |= FSL_SAI_CR4_FSE; - val_cr4 |= FSL_SAI_CR4_FSP; break; default: return -EINVAL; @@ -185,9 +184,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - if (fsl_dir == FSL_FMT_RECEIVER) - val_cr2 |= FSL_SAI_CR2_SYNC; - sai_writel(sai, val_cr2, sai->base + reg_cr2); sai_writel(sai, val_cr4, sai->base + reg_cr4); @@ -253,6 +249,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); + val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; if (sai->big_endian_data) val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); else @@ -272,7 +269,15 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 tcsr, rcsr, val_cr3, reg_cr3; + u32 tcsr, rcsr, val_cr2, val_cr3, reg_cr3; + + val_cr2 = sai_readl(sai, sai->base + FSL_SAI_TCR2); + val_cr2 &= ~FSL_SAI_CR2_SYNC; + sai_writel(sai, val_cr2, sai->base + FSL_SAI_TCR2); + + val_cr2 = sai_readl(sai, sai->base + FSL_SAI_RCR2); + val_cr2 |= FSL_SAI_CR2_SYNC; + sai_writel(sai, val_cr2, sai->base + FSL_SAI_RCR2); tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR); rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR); -- 1.8.4

2 3

[alsa-devel] HD audio problem (fwd)
by Russ Whitaker 31 Dec '13

31 Dec '13

Have not heard back from you. What should I do? ---------- Forwarded message ---------- Date: Thu, 19 Dec 2013 23:48:19 -0800 (PST) From: Russ Whitaker <russ(a)ashlandhome.net> To: alsa-devel(a)alsa-project.org Subject: HD audio problem Hi Bought a new mini-itx mb: ECS KBN-I On board sound chip: VT1705CF Installed kernel linux-3.10.24 Otherwise distro is Slackware64 14.1 mplayer plays .mp3 files just fine. No sound from video clips that use Adobe flashplayer, and I get this error message: ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave I think the problem is: card0 and card1 are interchanged as card0 is listed as the HDMI video output. How to fixit, please. Thank you, Russ

1 0

[alsa-devel] [PATCH] ASoC: fsl_sai: Fix one bug for hardware limitation.
by Xiubo Li 31 Dec '13

31 Dec '13

This is maybe one bug or a limitation of the hardware that the {T,R}CR2's Synchronous Mode bits must be set as late as possible, or the SAI device maybe hanged up, and there has not any explaination about this limitation in the SAI Data Sheet. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 31 ++++++++++++++++++++++--------- 1 file changed, 22 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index af80246..59228a10 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -62,6 +62,7 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, reg_cr2 = FSL_SAI_RCR2; val_cr2 = sai_readl(sai, sai->base + reg_cr2); + switch (clk_id) { case FSL_SAI_CLK_BUS: val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; @@ -82,6 +83,7 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, default: return -EINVAL; } + sai_writel(sai, val_cr2, sai->base + reg_cr2); return 0; @@ -138,14 +140,13 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, val_cr4 = sai_readl(sai, sai->base + reg_cr4); if (sai->big_endian_data) - val_cr4 |= FSL_SAI_CR4_MF; - else val_cr4 &= ~FSL_SAI_CR4_MF; + else + val_cr4 |= FSL_SAI_CR4_MF; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: val_cr4 |= FSL_SAI_CR4_FSE; - val_cr4 |= FSL_SAI_CR4_FSP; break; default: return -EINVAL; @@ -185,9 +186,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - if (fsl_dir == FSL_FMT_RECEIVER) - val_cr2 |= FSL_SAI_CR2_SYNC; - sai_writel(sai, val_cr2, sai->base + reg_cr2); sai_writel(sai, val_cr4, sai->base + reg_cr4); @@ -253,10 +251,11 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); + val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; if (sai->big_endian_data) - val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); - else val_cr5 |= FSL_SAI_CR5_FBT(0); + else + val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); val_cr4 |= FSL_SAI_CR4_FRSZ(channels); val_mr = ~0UL - ((1 << channels) - 1); @@ -327,8 +326,22 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + u32 val_cr2; + int ret; + + ret = clk_prepare_enable(sai->clk); + if (ret) + return ret; + + val_cr2 = sai_readl(sai, sai->base + FSL_SAI_TCR2); + val_cr2 &= ~FSL_SAI_CR2_SYNC; + sai_writel(sai, val_cr2, sai->base + FSL_SAI_TCR2); - return clk_prepare_enable(sai->clk); + val_cr2 = sai_readl(sai, sai->base + FSL_SAI_RCR2); + val_cr2 |= FSL_SAI_CR2_SYNC; + sai_writel(sai, val_cr2, sai->base + FSL_SAI_RCR2); + + return 0; } static void fsl_sai_shutdown(struct snd_pcm_substream *substream, -- 1.8.4

3 2

[alsa-devel] [PATCH] ASoC: fsl_sai: Add disable operation for the corresponding data channel.
by Xiubo Li 30 Dec '13

30 Dec '13

Enables/Disables the corresponding data channel for tx/rx operation. A channel must be enabled before its FIFO is accessed, and then disable it when tx/rx is stopped or idle. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 596aabb..af80246 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -124,20 +124,17 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, unsigned int fmt, int fsl_dir) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 val_cr2, val_cr3, val_cr4, reg_cr2, reg_cr3, reg_cr4; + u32 val_cr2, val_cr4, reg_cr2, reg_cr4; if (fsl_dir == FSL_FMT_TRANSMITTER) { reg_cr2 = FSL_SAI_TCR2; - reg_cr3 = FSL_SAI_TCR3; reg_cr4 = FSL_SAI_TCR4; } else { reg_cr2 = FSL_SAI_RCR2; - reg_cr3 = FSL_SAI_RCR3; reg_cr4 = FSL_SAI_RCR4; } val_cr2 = sai_readl(sai, sai->base + reg_cr2); - val_cr3 = sai_readl(sai, sai->base + reg_cr3); val_cr4 = sai_readl(sai, sai->base + reg_cr4); if (sai->big_endian_data) @@ -188,13 +185,10 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - val_cr3 |= FSL_SAI_CR3_TRCE; - if (fsl_dir == FSL_FMT_RECEIVER) val_cr2 |= FSL_SAI_CR2_SYNC; sai_writel(sai, val_cr2, sai->base + reg_cr2); - sai_writel(sai, val_cr3, sai->base + reg_cr3); sai_writel(sai, val_cr4, sai->base + reg_cr4); return 0; @@ -278,7 +272,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - unsigned int tcsr, rcsr; + u32 tcsr, rcsr, val_cr3, reg_cr3; tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR); rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR); @@ -286,17 +280,24 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { tcsr |= FSL_SAI_CSR_FRDE; rcsr &= ~FSL_SAI_CSR_FRDE; + reg_cr3 = FSL_SAI_TCR3; } else { rcsr |= FSL_SAI_CSR_FRDE; tcsr &= ~FSL_SAI_CSR_FRDE; + reg_cr3 = FSL_SAI_RCR3; } + val_cr3 = sai_readl(sai, sai->base + reg_cr3); + switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: tcsr |= FSL_SAI_CSR_TERE; rcsr |= FSL_SAI_CSR_TERE; + val_cr3 |= FSL_SAI_CR3_TRCE; + + sai_writel(sai, val_cr3, sai->base + reg_cr3); sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); break; @@ -308,8 +309,12 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, tcsr &= ~FSL_SAI_CSR_TERE; rcsr &= ~FSL_SAI_CSR_TERE; } + + val_cr3 &= ~FSL_SAI_CR3_TRCE; + sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); + sai_writel(sai, val_cr3, sai->base + reg_cr3); break; default: return -EINVAL; -- 1.8.4

2 1

[alsa-devel] [PATCH 1/1] ASoC: Samsung: Trivial cleanups in header files
by Sachin Kamat 30 Dec '13

30 Dec '13

commit 5d229ce569be ("ASoC: samsung: move plat/ headers to local directory") moved the header files but forgot to clean the pointers to their old locaton. Remove them now. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/regs-ac97.h | 9 ++++----- sound/soc/samsung/regs-iis.h | 9 ++++----- 2 files changed, 8 insertions(+), 10 deletions(-) diff --git a/sound/soc/samsung/regs-ac97.h b/sound/soc/samsung/regs-ac97.h index c3878f7acb83..a71be45bbffc 100644 --- a/sound/soc/samsung/regs-ac97.h +++ b/sound/soc/samsung/regs-ac97.h @@ -1,5 +1,4 @@ -/* arch/arm/mach-s3c2410/include/mach/regs-ac97.h - * +/* * Copyright (c) 2006 Simtec Electronics <linux(a)simtec.co.uk> * http://www.simtec.co.uk/products/SWLINUX/ * @@ -10,8 +9,8 @@ * S3C2440 AC97 Controller */ -#ifndef __ASM_ARCH_REGS_AC97_H -#define __ASM_ARCH_REGS_AC97_H __FILE__ +#ifndef __SAMSUNG_REGS_AC97_H__ +#define __SAMSUNG_REGS_AC97_H__ #define S3C_AC97_GLBCTRL (0x00) @@ -64,4 +63,4 @@ #define S3C_AC97_PCM_DATA (0x18) #define S3C_AC97_MIC_DATA (0x1C) -#endif /* __ASM_ARCH_REGS_AC97_H */ +#endif /* __SAMSUNG_REGS_AC97_H__ */ diff --git a/sound/soc/samsung/regs-iis.h b/sound/soc/samsung/regs-iis.h index a18d35e7a735..dc6cbbe9c4f0 100644 --- a/sound/soc/samsung/regs-iis.h +++ b/sound/soc/samsung/regs-iis.h @@ -1,5 +1,4 @@ -/* arch/arm/plat-samsung/include/plat/regs-iis.h - * +/* * Copyright (c) 2003 Simtec Electronics <linux(a)simtec.co.uk> * http://www.simtec.co.uk/products/SWLINUX/ * @@ -10,8 +9,8 @@ * S3C2410 IIS register definition */ -#ifndef __ASM_ARCH_REGS_IIS_H -#define __ASM_ARCH_REGS_IIS_H +#ifndef __SAMSUNG_REGS_IIS_H__ +#define __SAMSUNG_REGS_IIS_H__ #define S3C2410_IISCON (0x00) @@ -67,4 +66,4 @@ #define S3C2410_IISFIFO (0x10) -#endif /* __ASM_ARCH_REGS_IIS_H */ +#endif /* __SAMSUNG_REGS_IIS_H__ */ -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: fsl_sai: Move the global registers setting to _dai_probe()
by Xiubo Li 30 Dec '13

30 Dec '13

Because we cannot make sure which one of _dai_fmt() and _dai_sysclk() will be firstly called. So move the RCSR/TCSR and TCR1/RCR1's initialization to _dai_probe(), and this can make sure that before any of {T,R}CR{1~5} register to be set the RCSR/TCSR's RE/TE bit has been cleared for the hareware limitation. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index b72132f..596aabb 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -100,11 +100,6 @@ static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, if (ret) return ret; - sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR); - sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR); - sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1); - sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1); - ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, FSL_FMT_TRANSMITTER); if (ret) { @@ -351,6 +346,18 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); + int ret; + + ret = clk_prepare_enable(sai->clk); + if (ret) + return ret; + + sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR); + sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR); + sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1); + sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1); + + clk_disable_unprepare(sai->clk); snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx, &sai->dma_params_rx); -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: mcbsp: Trivial cleanup in asoc-ti-mcbsp.h
by Sachin Kamat 30 Dec '13

30 Dec '13

Commit 2203747c9771 ("ARM: omap: move platform_data definitions") moved the file to the current location but forgot to remove the pointer to its previous location. Clean it up. While at it also change the header file protection macros appropriately. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- include/linux/platform_data/asoc-ti-mcbsp.h | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/include/linux/platform_data/asoc-ti-mcbsp.h b/include/linux/platform_data/asoc-ti-mcbsp.h index c78d90b28b19..3c73c045f8da 100644 --- a/include/linux/platform_data/asoc-ti-mcbsp.h +++ b/include/linux/platform_data/asoc-ti-mcbsp.h @@ -1,6 +1,4 @@ /* - * arch/arm/plat-omap/include/mach/mcbsp.h - * * Defines for Multi-Channel Buffered Serial Port * * Copyright (C) 2002 RidgeRun, Inc. @@ -21,8 +19,8 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * */ -#ifndef __ASM_ARCH_OMAP_MCBSP_H -#define __ASM_ARCH_OMAP_MCBSP_H +#ifndef __ASOC_TI_MCBSP_H +#define __ASOC_TI_MCBSP_H #include <linux/spinlock.h> #include <linux/clk.h> -- 1.7.9.5

3 2

[alsa-devel] [PATCHv4 1/4] ALSA: Add SAI SoC Digital Audio Interface driver.
by Xiubo Li 27 Dec '13

27 Dec '13

Fix the description of one comment. This adds Freescale SAI ASoC Audio support. This implementation is only compatible with device tree definition. Features: o Supports playback/capture o Supports 16/20/24 bit PCM o Supports 8k~96k sample rates o Supports master and slave mode. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> Signed-off-by: Alison Wang <b18965(a)freescale.com --- sound/soc/fsl/Kconfig | 14 ++ sound/soc/fsl/Makefile | 4 +- sound/soc/fsl/fsl_sai.c | 467 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_sai.h | 114 ++++++++++++ 4 files changed, 598 insertions(+), 1 deletion(-) create mode 100644 sound/soc/fsl/fsl_sai.c create mode 100644 sound/soc/fsl/fsl_sai.h diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index b7ab71f..21db1b0 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,3 +1,7 @@ +config SND_SOC_FSL_SAI + tristate + select SND_SOC_GENERIC_DMAENGINE_PCM + config SND_SOC_FSL_SSI tristate @@ -213,3 +217,13 @@ config SND_SOC_IMX_MC13783 select SND_SOC_IMX_PCM_DMA endif # SND_IMX_SOC + +menuconfig SND_VF610_SOC + tristate "SoC Audio for Freescale VF610 CPUs" + select DMA_ENGINE + help + Say Y or M if you want to add support for codecs attached to + the VF610 CPUs. + + This will enable Freeacale SAI and SGTL5000 codec, and an extra + TWR-AUDIO-SGTL sub-board is needed for SGTL5000. diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 8db705b..aaccbee 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -10,11 +10,13 @@ obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o -# Freescale PowerPC SSI/DMA Platform Support +# Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-objs := fsl_ssi.o snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o +obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c new file mode 100644 index 0000000..aee0a2c --- /dev/null +++ b/sound/soc/fsl/fsl_sai.c @@ -0,0 +1,467 @@ +/* + * Freescale ALSA SoC Digital Audio Interface (SAI) driver. + * + * Copyright 2012-2013 Freescale Semiconductor, Inc. + * + * This program is free software, you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation, either version 2 of the License, or(at your + * option) any later version. + * + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/dmaengine.h> +#include <linux/module.h> +#include <linux/of_address.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> + +#include "fsl_sai.h" + +static inline u32 sai_readl(struct fsl_sai *sai, + const void __iomem *addr) +{ + u32 val; + + val = __raw_readl(addr); + + if (likely(sai->big_endian_regs)) + val = be32_to_cpu((__force __be32)val); + else + val = le32_to_cpu((__force __le32)val); + rmb(); + + return val; +} + +static inline void sai_writel(struct fsl_sai *sai, + u32 val, void __iomem *addr) +{ + wmb(); + if (likely(sai->big_endian_regs)) + val = (__force u32)cpu_to_be32(val); + else + val = (__force u32)cpu_to_le32(val); + + __raw_writel(val, addr); +} + +static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int fsl_dir) +{ + u32 val_cr2, reg_cr2; + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + if (fsl_dir == FSL_FMT_TRANSMITTER) + reg_cr2 = FSL_SAI_TCR2; + else + reg_cr2 = FSL_SAI_RCR2; + + val_cr2 = sai_readl(sai, sai->base + reg_cr2); + + switch (clk_id) { + case FSL_SAI_CLK_BUS: + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; + val_cr2 |= FSL_SAI_CR2_MSEL_BUS; + break; + case FSL_SAI_CLK_MAST1: + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; + val_cr2 |= FSL_SAI_CR2_MSEL_MCLK1; + break; + case FSL_SAI_CLK_MAST2: + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; + val_cr2 |= FSL_SAI_CR2_MSEL_MCLK2; + break; + case FSL_SAI_CLK_MAST3: + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; + val_cr2 |= FSL_SAI_CR2_MSEL_MCLK3; + break; + default: + return -EINVAL; + } + + sai_writel(sai, val_cr2, sai->base + reg_cr2); + + return 0; +} + +static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + int ret; + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + if (dir == SND_SOC_CLOCK_IN) + return 0; + + ret = clk_prepare_enable(sai->clk); + if (ret) + return ret; + + sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR); + sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR); + sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1); + sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1); + + ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, + FSL_FMT_TRANSMITTER); + if (ret) { + dev_err(cpu_dai->dev, + "Cannot set SAI's transmitter sysclk: %d\n", + ret); + return ret; + } + + ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, + FSL_FMT_RECEIVER); + if (ret) { + dev_err(cpu_dai->dev, + "Cannot set SAI's receiver sysclk: %d\n", + ret); + return ret; + } + + clk_disable_unprepare(sai->clk); + + return 0; +} + +static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, + unsigned int fmt, int fsl_dir) +{ + u32 val_cr2, val_cr3, val_cr4, reg_cr2, reg_cr3, reg_cr4; + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + if (fsl_dir == FSL_FMT_TRANSMITTER) { + reg_cr2 = FSL_SAI_TCR2; + reg_cr3 = FSL_SAI_TCR3; + reg_cr4 = FSL_SAI_TCR4; + } else { + reg_cr2 = FSL_SAI_RCR2; + reg_cr3 = FSL_SAI_RCR3; + reg_cr4 = FSL_SAI_RCR4; + } + + val_cr2 = sai_readl(sai, sai->base + reg_cr2); + val_cr3 = sai_readl(sai, sai->base + reg_cr3); + val_cr4 = sai_readl(sai, sai->base + reg_cr4); + + if (sai->big_endian_data) + val_cr4 |= FSL_SAI_CR4_MF; + else + val_cr4 &= ~FSL_SAI_CR4_MF; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val_cr4 |= FSL_SAI_CR4_FSE; + val_cr4 |= FSL_SAI_CR4_FSP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + val_cr4 |= FSL_SAI_CR4_FSP; + val_cr2 &= ~FSL_SAI_CR2_BCP; + break; + case SND_SOC_DAIFMT_IB_NF: + val_cr4 &= ~FSL_SAI_CR4_FSP; + val_cr2 &= ~FSL_SAI_CR2_BCP; + break; + case SND_SOC_DAIFMT_NB_IF: + val_cr4 |= FSL_SAI_CR4_FSP; + val_cr2 |= FSL_SAI_CR2_BCP; + break; + case SND_SOC_DAIFMT_NB_NF: + val_cr4 &= ~FSL_SAI_CR4_FSP; + val_cr2 |= FSL_SAI_CR2_BCP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + val_cr2 |= FSL_SAI_CR2_BCD_MSTR; + val_cr4 |= FSL_SAI_CR4_FSD_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFM: + val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR; + val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR; + break; + default: + return -EINVAL; + } + + val_cr3 |= FSL_SAI_CR3_TRCE; + + if (fsl_dir == FSL_FMT_RECEIVER) + val_cr2 |= FSL_SAI_CR2_SYNC; + + sai_writel(sai, val_cr2, sai->base + reg_cr2); + sai_writel(sai, val_cr3, sai->base + reg_cr3); + sai_writel(sai, val_cr4, sai->base + reg_cr4); + + return 0; +} + +static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + int ret; + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + ret = clk_prepare_enable(sai->clk); + if (ret) + return ret; + + ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_TRANSMITTER); + if (ret) { + dev_err(cpu_dai->dev, + "Cannot set SAI's transmitter format: %d\n", + ret); + return ret; + } + + ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER); + if (ret) { + dev_err(cpu_dai->dev, + "Cannot set SAI's receiver format: %d\n", + ret); + return ret; + } + + clk_disable_unprepare(sai->clk); + + return 0; +} + +static int fsl_sai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + u32 val_cr4, val_cr5, val_mr, reg_cr4, reg_cr5, reg_mr, word_width; + unsigned int channels = params_channels(params); + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + reg_cr4 = FSL_SAI_TCR4; + reg_cr5 = FSL_SAI_TCR5; + reg_mr = FSL_SAI_TMR; + } else { + reg_cr4 = FSL_SAI_RCR4; + reg_cr5 = FSL_SAI_RCR5; + reg_mr = FSL_SAI_RMR; + } + + val_cr4 = sai_readl(sai, sai->base + reg_cr4); + val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK; + val_cr4 &= ~FSL_SAI_CR4_FRSZ_MASK; + + val_cr5 = sai_readl(sai, sai->base + reg_cr5); + val_cr5 &= ~FSL_SAI_CR5_WNW_MASK; + val_cr5 &= ~FSL_SAI_CR5_W0W_MASK; + val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + word_width = 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + word_width = 20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + word_width = 24; + break; + default: + return -EINVAL; + } + + val_cr4 |= FSL_SAI_CR4_SYWD(word_width); + val_cr5 |= FSL_SAI_CR5_WNW(word_width); + val_cr5 |= FSL_SAI_CR5_W0W(word_width); + + if (sai->big_endian_data) + val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); + else + val_cr5 |= FSL_SAI_CR5_FBT(0); + + val_cr4 |= FSL_SAI_CR4_FRSZ(channels); + if (channels == 2 || channels == 1) + val_mr = ~0UL - ((1 << channels) - 1); + else + return -EINVAL; + + sai_writel(sai, val_cr4, sai->base + reg_cr4); + sai_writel(sai, val_cr5, sai->base + reg_cr5); + sai_writel(sai, val_mr, sai->base + reg_mr); + + return 0; +} + +static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *cpu_dai) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + unsigned int tcsr, rcsr; + + tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR); + rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + tcsr |= FSL_SAI_CSR_FRDE; + rcsr &= ~FSL_SAI_CSR_FRDE; + } else { + rcsr |= FSL_SAI_CSR_FRDE; + tcsr &= ~FSL_SAI_CSR_FRDE; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + tcsr |= FSL_SAI_CSR_TERE; + rcsr |= FSL_SAI_CSR_TERE; + sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); + sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!(cpu_dai->playback_active || cpu_dai->capture_active)) { + tcsr &= ~FSL_SAI_CSR_TERE; + rcsr &= ~FSL_SAI_CSR_TERE; + } + sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); + sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int fsl_sai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + return clk_prepare_enable(sai->clk); +} + +static void fsl_sai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + + clk_disable_unprepare(sai->clk); +} + +static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { + .set_sysclk = fsl_sai_set_dai_sysclk, + .set_fmt = fsl_sai_set_dai_fmt, + .hw_params = fsl_sai_hw_params, + .trigger = fsl_sai_trigger, + .startup = fsl_sai_startup, + .shutdown = fsl_sai_shutdown, +}; + +static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) +{ + struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); + + snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx, + &sai->dma_params_rx); + + snd_soc_dai_set_drvdata(cpu_dai, sai); + + return 0; +} + +static struct snd_soc_dai_driver fsl_sai_dai = { + .probe = fsl_sai_dai_probe, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_SAI_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_SAI_FORMATS, + }, + .ops = &fsl_sai_pcm_dai_ops, +}; + +static const struct snd_soc_component_driver fsl_component = { + .name = "fsl-sai", +}; + +static int fsl_sai_probe(struct platform_device *pdev) +{ + int ret; + struct fsl_sai *sai; + struct resource *res; + struct device_node *np = pdev->dev.of_node; + + sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); + if (!sai) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + sai->base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(sai->base)) + return PTR_ERR(sai->base); + + sai->clk = devm_clk_get(&pdev->dev, "sai"); + if (IS_ERR(sai->clk)) { + dev_err(&pdev->dev, "Cannot get SAI's clock\n"); + return PTR_ERR(sai->clk); + } + + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; + sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; + sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; + sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX; + + sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); + sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); + + platform_set_drvdata(pdev, sai); + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, + &fsl_sai_dai, 1); + if (ret) + return ret; + + return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); +} + +static const struct of_device_id fsl_sai_ids[] = { + { .compatible = "fsl,vf610-sai", }, + { /* sentinel */ } +}; + +static struct platform_driver fsl_sai_driver = { + .probe = fsl_sai_probe, + .driver = { + .name = "fsl-sai", + .owner = THIS_MODULE, + .of_match_table = fsl_sai_ids, + }, +}; +module_platform_driver(fsl_sai_driver); + +MODULE_DESCRIPTION("Freescale Soc SAI Interface"); +MODULE_AUTHOR("Xiubo Li, <Li.Xiubo(a)freescale.com>"); +MODULE_ALIAS("platform:fsl-sai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h new file mode 100644 index 0000000..41bb62e --- /dev/null +++ b/sound/soc/fsl/fsl_sai.h @@ -0,0 +1,114 @@ +/* + * Copyright 2012-2013 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __FSL_SAI_H +#define __FSL_SAI_H + +#include <sound/dmaengine_pcm.h> + +#define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +/* SAI Transmit/Recieve Control Register */ +#define FSL_SAI_TCSR 0x00 +#define FSL_SAI_RCSR 0x80 +#define FSL_SAI_CSR_TERE BIT(31) +#define FSL_SAI_CSR_FWF BIT(17) +#define FSL_SAI_CSR_FRIE BIT(8) +#define FSL_SAI_CSR_FRDE BIT(0) + +/* SAI Transmit Data/FIFO/MASK Register */ +#define FSL_SAI_TDR 0x20 +#define FSL_SAI_TFR 0x40 +#define FSL_SAI_TMR 0x60 + +/* SAI Recieve Data/FIFO/MASK Register */ +#define FSL_SAI_RDR 0xa0 +#define FSL_SAI_RFR 0xc0 +#define FSL_SAI_RMR 0xe0 + +/* SAI Transmit and Recieve Configuration 1 Register */ +#define FSL_SAI_TCR1 0x04 +#define FSL_SAI_RCR1 0x84 + +/* SAI Transmit and Recieve Configuration 2 Register */ +#define FSL_SAI_TCR2 0x08 +#define FSL_SAI_RCR2 0x88 +#define FSL_SAI_CR2_SYNC BIT(30) +#define FSL_SAI_CR2_MSEL_MASK (0xff << 26) +#define FSL_SAI_CR2_MSEL_BUS 0 +#define FSL_SAI_CR2_MSEL_MCLK1 BIT(26) +#define FSL_SAI_CR2_MSEL_MCLK2 BIT(27) +#define FSL_SAI_CR2_MSEL_MCLK3 (BIT(26) | BIT(27)) +#define FSL_SAI_CR2_BCP BIT(25) +#define FSL_SAI_CR2_BCD_MSTR BIT(24) + +/* SAI Transmit and Recieve Configuration 3 Register */ +#define FSL_SAI_TCR3 0x0c +#define FSL_SAI_RCR3 0x8c +#define FSL_SAI_CR3_TRCE BIT(16) +#define FSL_SAI_CR3_WDFL(x) (x) +#define FSL_SAI_CR3_WDFL_MASK 0x1f + +/* SAI Transmit and Recieve Configuration 4 Register */ +#define FSL_SAI_TCR4 0x10 +#define FSL_SAI_RCR4 0x90 +#define FSL_SAI_CR4_FRSZ(x) (((x) - 1) << 16) +#define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) +#define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) +#define FSL_SAI_CR4_SYWD_MASK (0x1f << 8) +#define FSL_SAI_CR4_MF BIT(4) +#define FSL_SAI_CR4_FSE BIT(3) +#define FSL_SAI_CR4_FSP BIT(1) +#define FSL_SAI_CR4_FSD_MSTR BIT(0) + +/* SAI Transmit and Recieve Configuration 5 Register */ +#define FSL_SAI_TCR5 0x14 +#define FSL_SAI_RCR5 0x94 +#define FSL_SAI_CR5_WNW(x) (((x) - 1) << 24) +#define FSL_SAI_CR5_WNW_MASK (0x1f << 24) +#define FSL_SAI_CR5_W0W(x) (((x) - 1) << 16) +#define FSL_SAI_CR5_W0W_MASK (0x1f << 16) +#define FSL_SAI_CR5_FBT(x) ((x) << 8) +#define FSL_SAI_CR5_FBT_MASK (0x1f << 8) + +/* SAI type */ +#define FSL_SAI_DMA BIT(0) +#define FSL_SAI_USE_AC97 BIT(1) +#define FSL_SAI_NET BIT(2) +#define FSL_SAI_TRA_SYN BIT(3) +#define FSL_SAI_REC_SYN BIT(4) +#define FSL_SAI_USE_I2S_SLAVE BIT(5) + +#define FSL_FMT_TRANSMITTER 0 +#define FSL_FMT_RECEIVER 1 + +/* SAI clock sources */ +#define FSL_SAI_CLK_BUS 0 +#define FSL_SAI_CLK_MAST1 1 +#define FSL_SAI_CLK_MAST2 2 +#define FSL_SAI_CLK_MAST3 3 + +/* SAI data transfer numbers per DMA request */ +#define FSL_SAI_MAXBURST_TX 6 +#define FSL_SAI_MAXBURST_RX 6 + +struct fsl_sai { + struct clk *clk; + + void __iomem *base; + + bool big_endian_regs; + bool big_endian_data; + + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; +}; + +#endif /* __FSL_SAI_H */ -- 1.8.4

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[alsa-devel] microphone recording got lower or higher tune than standard - 2
by 刘伟丰 27 Dec '13

27 Dec '13

part two: these are my /proc/asound/cardXXX: ls /proc/asound/card* /proc/asound/cards /proc/asound/card0: id pcm0c pcm0p ---------------------------------------------------------------- ls /proc/asound/card0/pcm0c info oss sub0 cat /proc/asound/card0/pcm0c/oss #nothing cat /proc/asound/card0/pcm0c/info card: 0 device: 0 subdevice: 0 stream: CAPTURE id: CS42L52 PCM cs42l52-0 name: subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 ls /proc/asound/card0/pcm0c/sub0/ hw_params info status sw_params cat /proc/asound/card0/pcm0c/sub0/hw_params closed cat /proc/asound/card0/pcm0c/sub0/status closed cat /proc/asound/card0/pcm0c/sub0/sw_params closed cat /proc/asound/card0/pcm0c/sub0/info card: 0 device: 0 subdevice: 0 stream: CAPTURE id: CS42L52 PCM cs42l52-0 name: subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 ---------------------------------------------------------------- cat /proc/asound/card0/pcm0p/oss #nothing cat /proc/asound/card0/pcm0p/info card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: CS42L52 PCM cs42l52-0 name: subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 ls /proc/asound/card0/pcm0p/sub0/ hw_params info status sw_params cat /proc/asound/card0/pcm0p/sub0/hw_params closed cat /proc/asound/card0/pcm0p/sub0/status closed cat /proc/asound/card0/pcm0p/sub0/sw_params closed cat /proc/asound/card0/pcm0p/sub0/info card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: CS42L52 PCM cs42l52-0 name: subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 I'd like post the ouput informations coming from when Audio Tuninf Wizard start and, ones when I selected the first option above in Audio Tuninf Wizard before alsa-info as I think the things maybe necessarily needed also: #run mumble in a terminal: administrator@sdn:~$ mumble QGtkStyle was unable to detect the current GTK+ theme. PulseAudio: Connection failure: Connection refused G15LCDEngineUnix: Unable to connect to G15Daemon. CELT bitstream 8000000b from libcelt0.so.0.0.0 Locale is en_US TextToSpeech: Failed to contact speech dispatcher. Database SQLite: "3.7.9" WARNING The program 'mumble' uses the Apple Bonjour compatibility layer of Avahi. WARNING Please fix your application to use the native API of Avahi! WARNING For more information see http://0pointer.de/avahi-compat?s=libdns_sd&e=mumble Overlay: Listening on "/home/administrator/.MumbleOverlayPipe" GlobalShortcutX: Unable to open any keyboard input devices under /dev/input, falling back to XInput GlobalShortcutX: Using XInput 2.2 GlobalShortcutX: XInput 4:Virtual core XTEST pointer GlobalShortcutX: XInput 5:Virtual core XTEST keyboard GlobalShortcutX: XInput 9:key_input GlobalShortcutX: XInput 10:aml_keypad GlobalShortcutX: XInput 6:Logitech USB Optical Mouse GlobalShortcutX: XInput 7:Logitech USB Keyboard GlobalShortcutX: XInput 8:Logitech USB Keyboard No systemtrayicon available AudioInput: 40000 bits/s, 48000 hz, 480 sample CELT ALSAAudioOutput: Initialized ALSAAudioOutput: ALSA reports 10000 output channels. Clamping to 2. ALSAAudioOutput: Actual buffer 48000 hz, 2 channel 3072 samples [512 per period] ALSAAudioOutput: Initializing 2 channel, 48000 hz mixer AudioOutput: Initialized 2 channel 48000 hz mixer ALSAAudioInput: Initing audiocapture default. ALSAAudioInput: Actual buffer 48000 hz, 1 channel 4096 samples [512 per period] AudioInput: Initialized mixer for 1 channel 48000 hz mic and 0 channel 48000 hz echo warning: The VAD has been replaced by a hack pending a complete rewrite #when the first option is selected: ALSAAudioInput: Releasing ALSA Mic. ALSAAudioOutput: Destroyed AudioInput: 40000 bits/s, 48000 hz, 480 sample CELT ALSAAudioOutput: Initialized ALSAAudioInput: Initing audiocapture default. ALSAAudioInput: Actual buffer 48000 hz, 1 channel 4096 samples [512 per period] AudioInput: Initialized mixer for 1 channel 48000 hz mic and 0 channel 48000 hz echo ALSAAudioInput: Releasing ALSA Mic. ALSAAudioOutput: ALSA reports 10000 output channels. Clamping to 2. ALSAAudioOutput: Actual buffer 48000 hz, 2 channel 3072 samples [512 per period] ALSAAudioOutput: Initializing 2 channel, 48000 hz mixer AudioOutput: Initialized 2 channel 48000 hz mixer ALSAAudioOutput: Destroyed AudioInput: 40000 bits/s, 48000 hz, 480 sample CELT ALSAAudioOutput: Initialized ALSAAudioInput: Initing audiocapture default. ALSAAudioInput: Actual buffer 48000 hz, 1 channel 4096 samples [512 per period] AudioInput: Initialized mixer for 1 channel 48000 hz mic and 0 channel 48000 hz echo ALSAAudioOutput: ALSA reports 10000 output channels. Clamping to 2. ALSAAudioOutput: Actual buffer 48000 hz, 2 channel 3072 samples [512 per period] ALSAAudioOutput: Initializing 2 channel, 48000 hz mixer AudioOutput: Initialized 2 channel 48000 hz mixer warning: The VAD has been replaced by a hack pending a complete rewrite ALSAAudioInput: Releasing ALSA Mic. AudioInput: 40000 bits/s, 48000 hz, 480 sample CELT ALSAAudioInput: Initing audiocapture hw:CARD=AMLCS42L52,DEV=0. ALSAAudioInput: Actual buffer 48000 hz, 1 channel 4096 samples [512 per period] AudioInput: Initialized mixer for 1 channel 48000 hz mic and 0 channel 48000 hz echo warning: The VAD has been replaced by a hack pending a complete rewrite thanks a lot Liuwf

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