From 2aba76f014a7b56ab4fe75845c5fd57b5590acc2 Mon Sep 17 00:00:00 2001 From: Michael Williamson Date: Fri, 20 May 2011 10:26:06 -0400 Subject: audio: tlv320aic26: fix PLL register configuration The current PLL configuration code for the tlc320aic26 codec appears to assume a hardcoded system clock of 12 MHz. Use the clock value provided by the DAI_OPS API for the calculation. Tested using a MityDSP-L138 platform providing a 24.576 MHz clock. Signed-off-by: Michael Williamson Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320aic26.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index e2a7608d3944..7859bdcc93db 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -161,10 +161,18 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL; } - /* Configure PLL */ + /** + * Configure PLL + * fsref = (mclk * PLLM) / 2048 + * where PLLM = J.DDDD (DDDD register ranges from 0 to 9999, decimal) + */ pval = 1; - jval = (fsref == 44100) ? 7 : 8; - dval = (fsref == 44100) ? 5264 : 1920; + /* compute J portion of multiplier */ + jval = fsref / (aic26->mclk / 2048); + /* compute fractional DDDD component of multiplier */ + dval = fsref - (jval * (aic26->mclk / 2048)); + dval = (10000 * dval) / (aic26->mclk / 2048); + dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval); qval = 0; reg = 0x8000 | qval << 11 | pval << 8 | jval << 2; aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg); -- cgit v1.2.3 From 508b76864c18f34f8d6ba08d192f5817f8dc8ead Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 20 May 2011 16:52:37 +0300 Subject: ASoC: tlv320aic3x: Don't sync first two registers from register cache There is no need to sync first two registers from cache to hw after a reset. First one is used to select page for register access and this driver is normally accessing page 0 only. Second one does a software reset which is obviously unneeded after hardware or previous software reset command. Signed-off-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index c3d96fc8c267..9047bb173c6b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1114,7 +1114,7 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) /* Sync reg_cache with the hardware */ codec->cache_only = 0; - for (i = 0; i < ARRAY_SIZE(aic3x_reg); i++) + for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++) snd_soc_write(codec, i, cache[i]); if (aic3x->model == AIC3X_MODEL_3007) aic3x_init_3007(codec); -- cgit v1.2.3 From 9fb352b18b11124ed1ddebc0d74ebbd7ba8defd7 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 20 May 2011 16:52:38 +0300 Subject: ASoC: tlv320aic3x: Do soft reset to codec when going to bias off state TLV320AIC33, TLV320AIC34 and I believe others too in this family have some hw bugs that cause that analogue and digital VDD supplies remain leaking up to a few mA of current after certain use cases even the hw blocks inside codec are driven to off. Highest leakages occur after using the bypass paths inside codec but it is possible to get smaller leakages just by toggling mute switches in unused audio paths (i.e. no DAPM changes) while codec is on due another active audio path. While some cases are able to workaroud by making sure that e.g. output mixer switches are muted before powering down the output stage this doesn't help all the cases. Therefore use the software reset command to clear possible leakage currents since that works in every cases and affects only this codec instance. Only drawback is that now cache sync is required everytime when codec bias comes out from bias off state, not only when supply regulators were off. Signed-off-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320aic3x.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 9047bb173c6b..789453d44ec5 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1120,6 +1120,13 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) aic3x_init_3007(codec); codec->cache_sync = 0; } else { + /* + * Do soft reset to this codec instance in order to clear + * possible VDD leakage currents in case the supply regulators + * remain on + */ + snd_soc_write(codec, AIC3X_RESET, SOFT_RESET); + codec->cache_sync = 1; aic3x->power = 0; /* HW writes are needless when bias is off */ codec->cache_only = 1; -- cgit v1.2.3 From e999dc50404d401150a5429b6459473a691fd1a0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Jun 2011 12:14:07 +0100 Subject: ASoC: Fix Blackfin I2S _pointer() implementation return in bounds values The Blackfin DMA controller can report one frame beyond the end of the buffer in the wraparound case but ALSA requires that the pointer always be in the buffer. Do the wraparound to handle this. A similar bug is likely to apply to the other Blackfin PCM drivers but the code is less obvious to inspection and I don't have a user to test. Reported-by: Kieran O'Leary Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/blackfin/bf5xx-i2s-pcm.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index b5101efd1c87..f1fd95bb6416 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -138,11 +138,20 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) pr_debug("%s enter\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { diff = sport_curr_offset_tx(sport); - frames = bytes_to_frames(substream->runtime, diff); } else { diff = sport_curr_offset_rx(sport); - frames = bytes_to_frames(substream->runtime, diff); } + + /* + * TX at least can report one frame beyond the end of the + * buffer if we hit the wraparound case - clamp to within the + * buffer as the ALSA APIs require. + */ + if (diff == snd_pcm_lib_buffer_bytes(substream)) + diff = 0; + + frames = bytes_to_frames(substream->runtime, diff); + return frames; } -- cgit v1.2.3 From 713d1369789f2a2336c3431b15276c968862bdb7 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 1 Jul 2011 13:56:13 -0600 Subject: ASoC: Tegra: I2S: Ensure clock is enabled when writing regs The I2S controller needs a clock to respond to register writes. Without this, register writes will at worst hang the CPU. In practice, I've only observed writes being dropped. Luckily, the dropped register writes historically had no effect: TEGRA_I2S_TIMING: The value we wrote was the reset default. TEGRA_I2S_FIFO_SCR: The default was for the FIFOs to request more data when one slot was empty. The requested value was for the FIFOs to request when four slots were empty. The DMA controller in the mainline kernel is configured to burst a single entry at a time into the FIFO, hence there was no issue. The only negative effect was on bus efficiency losses due to an increased number of arbitration attempts. However, in various non-upstream changes, the DMA controller now bursts four entries at a time into the FIFO. If there is only space for one entry, the data is simply dropped. In practice, this resulted in 3/4 of samples being dropped, and playback at 4x the expected rate and pitch. By fixing the clocking issue, this is solved. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_i2s.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 6b817e20548c..95f03c10b4f7 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -222,12 +222,18 @@ static int tegra_i2s_hw_params(struct snd_pcm_substream *substream, if (i2sclock % (2 * srate)) reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE; + if (!i2s->clk_refs) + clk_enable(i2s->clk_i2s); + tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg); tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR, TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); + if (!i2s->clk_refs) + clk_disable(i2s->clk_i2s); + return 0; } -- cgit v1.2.3 From 8e9ddf811ba021506d2316fcfe619faa0ab3f567 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 1 Jul 2011 17:24:46 -0700 Subject: ASoC: Ensure we delay long enough for WM8994 FLL to lock when starting This delay is very conservative. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 970a95c5360b..c2fc0356c2a4 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1713,6 +1713,8 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, WM8994_FLL1_ENA | WM8994_FLL1_FRAC, reg); + + msleep(5); } wm8994->fll[id].in = freq_in; -- cgit v1.2.3 From 873bd4cb4fbba6a3e99f750e17ef2ba6ef96e9d3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Jul 2011 09:25:59 +0200 Subject: ASoC: Don't set invalid name string to snd_card->driver field The snd_card->driver field contains a driver name string, and in general it shouldn't contain space or special letters. The commit 2b39535b9e54888649923beaab443af212b6c0fd changed the string copy from card->name, but the long name string may contain such letters, thus it may still lead to a segfault. A temporary fix is not to copy the long name string but just keep it empty as the earlier version did. Reported-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/soc/soc-core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d75043ed7fc0..b194be09e74d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1929,8 +1929,9 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), "%s", card->long_name ? card->long_name : card->name); - snprintf(card->snd_card->driver, sizeof(card->snd_card->driver), - "%s", card->driver_name ? card->driver_name : card->name); + if (card->driver_name) + strlcpy(card->snd_card->driver, card->driver_name, + sizeof(card->snd_card->driver)); if (card->late_probe) { ret = card->late_probe(card); -- cgit v1.2.3 From 4c7c5374ce6876d3d2c7013ef815051df4258952 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 4 Jul 2011 10:27:51 -0700 Subject: ASoC: Manage WM8731 ACTIVE bit as a supply widget Now we have supply widgets there's no need to open code the handling of the ACTIVE bit. Signed-off-by: Mark Brown Tested-by: Nicolas Ferre Acked-by: Liam Girdwood --- sound/soc/codecs/wm8731.c | 29 +++-------------------------- 1 file changed, 3 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 2dc964b55e4f..76b4361e9b80 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -175,6 +175,7 @@ static const struct snd_kcontrol_new wm8731_input_mux_controls = SOC_DAPM_ENUM("Input Select", wm8731_insel_enum); static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("ACTIVE",WM8731_ACTIVE, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OSC", WM8731_PWR, 5, 1, NULL, 0), SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, &wm8731_output_mixer_controls[0], @@ -204,6 +205,8 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source, static const struct snd_soc_dapm_route wm8731_intercon[] = { {"DAC", NULL, "OSC", wm8731_check_osc}, {"ADC", NULL, "OSC", wm8731_check_osc}, + {"DAC", NULL, "ACTIVE"}, + {"ADC", NULL, "ACTIVE"}, /* output mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, @@ -315,29 +318,6 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8731_pcm_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - - /* set active */ - snd_soc_write(codec, WM8731_ACTIVE, 0x0001); - - return 0; -} - -static void wm8731_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - - /* deactivate */ - if (!codec->active) { - udelay(50); - snd_soc_write(codec, WM8731_ACTIVE, 0x0); - } -} - static int wm8731_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; @@ -480,7 +460,6 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: - snd_soc_write(codec, WM8731_ACTIVE, 0x0); snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -496,9 +475,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S24_LE) static struct snd_soc_dai_ops wm8731_dai_ops = { - .prepare = wm8731_pcm_prepare, .hw_params = wm8731_hw_params, - .shutdown = wm8731_shutdown, .digital_mute = wm8731_mute, .set_sysclk = wm8731_set_dai_sysclk, .set_fmt = wm8731_set_dai_fmt, -- cgit v1.2.3 From 9c7a083d94656ad6d6f2e03ba90194f2cc5bced5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Jul 2011 09:25:54 +0200 Subject: ALSA: hda - Change all ADCs for dual-adc switching mode for Realtek When the dual-adc switching mode is active in Realtek auto-parser, we need to couple all ADCs as a single capture-volume. Currently, the volume control changes only the first ADC, thus others may remain silent. This patch fixes the problem. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 33 +++++++++++++++++++++++---------- 1 file changed, 23 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d21191dcfe88..7d492713c1c1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2715,17 +2715,30 @@ typedef int (*getput_call_t)(struct snd_kcontrol *kcontrol, static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol, - getput_call_t func) + getput_call_t func, bool check_adc_switch) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - int err; + int i, err; mutex_lock(&codec->control_mutex); - kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[adc_idx], - 3, 0, HDA_INPUT); - err = func(kcontrol, ucontrol); + if (check_adc_switch && spec->dual_adc_switch) { + for (i = 0; i < spec->num_adc_nids; i++) { + kcontrol->private_value = + HDA_COMPOSE_AMP_VAL(spec->adc_nids[i], + 3, 0, HDA_INPUT); + err = func(kcontrol, ucontrol); + if (err < 0) + goto error; + } + } else { + i = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + kcontrol->private_value = + HDA_COMPOSE_AMP_VAL(spec->adc_nids[i], + 3, 0, HDA_INPUT); + err = func(kcontrol, ucontrol); + } + error: mutex_unlock(&codec->control_mutex); return err; } @@ -2734,14 +2747,14 @@ static int alc_cap_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { return alc_cap_getput_caller(kcontrol, ucontrol, - snd_hda_mixer_amp_volume_get); + snd_hda_mixer_amp_volume_get, false); } static int alc_cap_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { return alc_cap_getput_caller(kcontrol, ucontrol, - snd_hda_mixer_amp_volume_put); + snd_hda_mixer_amp_volume_put, true); } /* capture mixer elements */ @@ -2751,14 +2764,14 @@ static int alc_cap_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { return alc_cap_getput_caller(kcontrol, ucontrol, - snd_hda_mixer_amp_switch_get); + snd_hda_mixer_amp_switch_get, false); } static int alc_cap_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { return alc_cap_getput_caller(kcontrol, ucontrol, - snd_hda_mixer_amp_switch_put); + snd_hda_mixer_amp_switch_put, true); } #define _DEFINE_CAPMIX(num) \ -- cgit v1.2.3 From bd7fdbcaa2d06d446577fd3c9b81847b04469e01 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 6 Jul 2011 17:58:56 -0700 Subject: ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2 mask didn't cover update-data Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/ak4642.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 4be0570e3f1f..65f46047b1cb 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -357,7 +357,7 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) default: return -EINVAL; } - snd_soc_update_bits(codec, PW_MGMT2, MS, data); + snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data); snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); /* format type */ -- cgit v1.2.3 From abaead6ac55dbda8b4bae40251d69dc3f0c49b1c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 9 Jul 2011 11:55:28 +0200 Subject: ALSA: hda - Fix a copmile warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit It's harmless but annyoing. sound/pci/hda/patch_realtek.c: In function ‘alc_cap_getput_caller’: sound/pci/hda/patch_realtek.c:2722:9: warning: ‘err’ may be used uninitialized in this function Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7d492713c1c1..b48fb43b5448 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2719,7 +2719,7 @@ static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - int i, err; + int i, err = 0; mutex_lock(&codec->control_mutex); if (check_adc_switch && spec->dual_adc_switch) { -- cgit v1.2.3