From 2b30a55d4d09254d6b25814bf6ac0b7843afdc99 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Tue, 15 Jul 2008 15:07:19 +0200 Subject: ALSA: Au1xpsc: psc not disabled when TX is idle TX idleness isn't tested, but RX twice. PSC is not disabled when TX is idle Signed-off-by: Roel Kluin Acked-by: Manuel Lauss Signed-off-by: Takashi Iwai --- sound/soc/au1x/psc-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index ba4b5c199f2..9384702c7eb 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -231,7 +231,7 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype) /* if both TX and RX are idle, disable PSC */ stat = au_readl(I2S_STAT(pscdata)); - if (!(stat & (PSC_I2SSTAT_RB | PSC_I2SSTAT_RB))) { + if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) { au_writel(0, I2S_CFG(pscdata)); au_sync(); au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata)); -- cgit v1.2.3 From 2927d6eeca0a5004d81fa5bedbdf3f2b1b842903 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Jul 2008 15:06:50 +0100 Subject: ALSA: ASoC: Refactor DAPM event handler The DAPM event callback code has many layers of indentation, taking it over 80 columns. Refactor the code to give less indentation in order to avoid checkpatch issues on further changes and exploding indentation. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/soc-dapm.c | 79 +++++++++++++++++++++++++++------------------------- 1 file changed, 41 insertions(+), 38 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c87061c2a6..17698ef58df 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -586,45 +586,48 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) power_change = (w->power == power) ? 0: 1; w->power = power; + if (!power_change) + continue; + /* call any power change event handlers */ - if (power_change) { - if (w->event) { - pr_debug("power %s event for %s flags %x\n", - w->power ? "on" : "off", w->name, w->event_flags); - if (power) { - /* power up event */ - if (w->event_flags & SND_SOC_DAPM_PRE_PMU) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - return ret; - } - dapm_update_bits(w); - if (w->event_flags & SND_SOC_DAPM_POST_PMU){ - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - return ret; - } - } else { - /* power down event */ - if (w->event_flags & SND_SOC_DAPM_PRE_PMD) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - return ret; - } - dapm_update_bits(w); - if (w->event_flags & SND_SOC_DAPM_POST_PMD) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - return ret; - } - } - } else - /* no event handler */ - dapm_update_bits(w); + if (w->event) + pr_debug("power %s event for %s flags %x\n", + w->power ? "on" : "off", + w->name, w->event_flags); + + /* power up pre event */ + if (power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); + if (ret < 0) + return ret; + } + + /* power down pre event */ + if (!power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); + if (ret < 0) + return ret; + } + + dapm_update_bits(w); + + /* power up post event */ + if (power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMU)) { + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); + if (ret < 0) + return ret; + } + + /* power down post event */ + if (!power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); + if (ret < 0) + return ret; } } } -- cgit v1.2.3 From 9dd8d812d3b4d208a769ca3cf23a7f9294632d0d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Jul 2008 15:06:51 +0100 Subject: ALSA: ASoC: Factor PGA DAPM handling into main This allows pre and post event hooks to be provided for PGA widgets. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/soc-dapm.c | 26 ++++++++------------------ 1 file changed, 8 insertions(+), 18 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 17698ef58df..820347c9ae4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -523,24 +523,6 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) continue; } - /* programmable gain/attenuation */ - if (w->id == snd_soc_dapm_pga) { - int on; - in = is_connected_input_ep(w); - dapm_clear_walk(w->codec); - out = is_connected_output_ep(w); - dapm_clear_walk(w->codec); - w->power = on = (out != 0 && in != 0) ? 1 : 0; - - if (!on) - dapm_set_pga(w, on); /* lower volume to reduce pops */ - dapm_update_bits(w); - if (on) - dapm_set_pga(w, on); /* restore volume from zero */ - - continue; - } - /* pre and post event widgets */ if (w->id == snd_soc_dapm_pre) { if (!w->event) @@ -611,8 +593,16 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) return ret; } + /* Lower PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && !power) + dapm_set_pga(w, power); + dapm_update_bits(w); + /* Raise PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && power) + dapm_set_pga(w, power); + /* power up post event */ if (power && w->event && (w->event_flags & SND_SOC_DAPM_POST_PMU)) { -- cgit v1.2.3 From 9432484110263e9418f380faf05fa9e2e7fb87a0 Mon Sep 17 00:00:00 2001 From: Marek Vasut Date: Sun, 20 Jul 2008 17:36:20 +0200 Subject: ALSA: soc - wm9712 mono mixer this fixes typo in wm9712 codec which prevents it from registering all audio routes (and thus working correctly). Please consider applying. (Tested and works on palmtx, palmld and palmt5) Signed-off-by: Marek Vasut Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm9712.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 9fc8edd8222..1fb7f9a7aec 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -427,20 +427,20 @@ static const struct snd_soc_dapm_route audio_map[] = { {"HPOUTR", NULL, "Headphone PGA"}, {"Headphone PGA", NULL, "Right HP Mixer"}, - /* mono hp mixer */ - {"Mono HP Mixer", NULL, "Left HP Mixer"}, - {"Mono HP Mixer", NULL, "Right HP Mixer"}, + /* mono mixer */ + {"Mono Mixer", NULL, "Left HP Mixer"}, + {"Mono Mixer", NULL, "Right HP Mixer"}, /* Out3 Mux */ {"Out3 Mux", "Left", "Left HP Mixer"}, {"Out3 Mux", "Mono", "Phone Mixer"}, - {"Out3 Mux", "Left + Right", "Mono HP Mixer"}, + {"Out3 Mux", "Left + Right", "Mono Mixer"}, {"Out 3 PGA", NULL, "Out3 Mux"}, {"OUT3", NULL, "Out 3 PGA"}, /* speaker Mux */ {"Speaker Mux", "Speaker Mix", "Speaker Mixer"}, - {"Speaker Mux", "Headphone Mix", "Mono HP Mixer"}, + {"Speaker Mux", "Headphone Mix", "Mono Mixer"}, {"Speaker PGA", NULL, "Speaker Mux"}, {"LOUT2", NULL, "Speaker PGA"}, {"ROUT2", NULL, "Speaker PGA"}, -- cgit v1.2.3