From 3d15aacbb802af72b4ff0c3ba576536cdb3bace0 Mon Sep 17 00:00:00 2001 From: Andrew Bresticker Date: Sun, 19 May 2013 22:58:07 -0700 Subject: ASoC: max98090: request IRQF_ONESHOT interrupt request_threaded_irq() rejects calls which both do not specify a handler (indicating that the primary IRQ handler should be used) and do not set IRQF_ONESHOT because the combination is unsafe with level-triggered interrupts. It is safe in this case, though, since max98090 IRQs are edge-triggered and the interrupts aren't ACK'ed until the codec's IRQ status register is read. Because of this, an IRQF_ONESHOT interrupt doesn't really make a difference, but request one anyway in order to make request_threaded_irq() happy. Signed-off-by: Andrew Bresticker Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index ce0d36412c97..8d14a76c7249 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2233,7 +2233,7 @@ static int max98090_probe(struct snd_soc_codec *codec) dev_dbg(codec->dev, "irq = %d\n", max98090->irq); ret = request_threaded_irq(max98090->irq, NULL, - max98090_interrupt, IRQF_TRIGGER_FALLING, + max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "max98090_interrupt", codec); if (ret < 0) { dev_err(codec->dev, "request_irq failed: %d\n", -- cgit v1.2.3 From 2c071ed7c3660992951abe4b560359058ce41f68 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 20 May 2013 08:33:54 +0100 Subject: ASoC: soc-compress: Send correct stream event for capture start Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3853f7eb3f28..06a8000aa07b 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -220,8 +220,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, goto err; } - snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_START); + if (cstream->direction == SND_COMPRESS_PLAYBACK) + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + SND_SOC_DAPM_STREAM_START); + else + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, + SND_SOC_DAPM_STREAM_START); /* cancel any delayed stream shutdown that is pending */ rtd->pop_wait = 0; -- cgit v1.2.3 From 62cc4d595fe96106ff793cbebbff051179d7619e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 20 May 2013 11:28:35 -0500 Subject: ASoC: wm5110: Add missing speaker initialisation Add callback to initialise the speaker in the core following the recent changes to handling of integration with the thermal interrupts. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 731884e04776..c00480bdf824 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -976,6 +976,8 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; + arizona_init_spk(codec); + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); priv->core.arizona->dapm = &codec->dapm; -- cgit v1.2.3 From 796718925159523919a589ecbd6d1811c22ef55f Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 16 May 2013 15:25:01 +0200 Subject: ASoC: davinci: fix sample rotation McASP serial audio engine needs different rotation values on TX and RX channels. Commit dde109fb462 ("ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback") changed the calculation to fix the playback format, but broke the capture stream by doing it for both TXFMT and RXFMT. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown Cc: stable@vger.kernel.org [3.9 only] --- sound/soc/davinci/davinci-mcasp.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 56ecfc72f2e9..81490febac6d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -631,7 +631,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int word_length) { u32 fmt; - u32 rotate = (word_length / 4) & 0x7; + u32 tx_rotate = (word_length / 4) & 0x7; + u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; /* @@ -655,9 +656,9 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt), TXSSZ(0x0F)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXROT(rotate), TXROT(7)); + TXROT(tx_rotate), TXROT(7)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXROT(rotate), RXROT(7)); + RXROT(rx_rotate), RXROT(7)); mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); } -- cgit v1.2.3 From 0b6e81d1658e2aa6b3acc942088c529fee5aa62e Mon Sep 17 00:00:00 2001 From: Nicolas Schichan Date: Wed, 22 May 2013 19:19:25 +0200 Subject: ASoC: cs42l52: microphone bias is controlled by IFACE_CTL2 register. Signed-off-by: Nicolas Schichan Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0f6f481cec09..1a17c8575886 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -225,7 +225,7 @@ static const char * const mic_bias_level_text[] = { }; static const struct soc_enum mic_bias_level_enum = - SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0, + SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0, ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text); static const char * const cs42l52_mic_text[] = { "Single", "Differential" }; -- cgit v1.2.3 From 99674c721fd9393030365b66cbbceaa193b0c0fd Mon Sep 17 00:00:00 2001 From: Nicolas Schichan Date: Wed, 22 May 2013 19:19:26 +0200 Subject: ASoC: cs42l52: fix bogus shifts in "Speaker Volume" and "PCM Mixer Volume" controls. Signed-off-by: Nicolas Schichan Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 1a17c8575886..8465c1fa6bf9 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -413,7 +413,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Headphone Analog Gain", hp_gain_enum), SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, - CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv), + CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv), @@ -441,7 +441,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, - 6, 0x7f, 0x19, hl_tlv), + 0, 0x7f, 0x19, hl_tlv), SOC_DOUBLE_R("PCM Mixer Switch", CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1), -- cgit v1.2.3 From 40e2516acb426f349c70e3bada821f3203b69de2 Mon Sep 17 00:00:00 2001 From: Nicolas Schichan Date: Wed, 22 May 2013 19:19:27 +0200 Subject: ASoC: cs42l52: fix master playback mute mask. The mask should define the bits to change in the register, not the bits to preserve. This fixes the inadvertent changes of the "Headphone Analog Gain" value during mute/unmute. Signed-off-by: Nicolas Schichan Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 60985c059071..4277012c4719 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -157,7 +157,7 @@ #define CS42L52_PB_CTL1_INV_PCMA (1 << 2) #define CS42L52_PB_CTL1_MSTB_MUTE (1 << 1) #define CS42L52_PB_CTL1_MSTA_MUTE (1 << 0) -#define CS42L52_PB_CTL1_MUTE_MASK 0xFFFD +#define CS42L52_PB_CTL1_MUTE_MASK 0x03 #define CS42L52_PB_CTL1_MUTE 3 #define CS42L52_PB_CTL1_UNMUTE 0 -- cgit v1.2.3 From d47333ddb234dbc661ab2a4fe019758bd33ba33b Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Thu, 23 May 2013 13:38:29 +0200 Subject: ALSA: usb-6fire: Modify firmware version check Check only the uppermost 16 bits instead of the whole 32 bits of the version information. Do this because all firmware version tested with this version information worked correctly and the strict check causes problems for several users. Signed-off-by: Torsten Schenk Signed-off-by: Takashi Iwai --- sound/usb/6fire/firmware.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index a1d9b0792a1e..b9defcdeb7ef 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -42,8 +42,8 @@ static const u8 ep_w_max_packet_size[] = { 0x94, 0x01, 0x5c, 0x02 /* alt 3: 404 EP2 and 604 EP6 (25 fpp) */ }; -static const u8 known_fw_versions[][4] = { - { 0x03, 0x01, 0x0b, 0x00 } +static const u8 known_fw_versions[][2] = { + { 0x03, 0x01 } }; struct ihex_record { @@ -343,7 +343,7 @@ static int usb6fire_fw_check(u8 *version) int i; for (i = 0; i < ARRAY_SIZE(known_fw_versions); i++) - if (!memcmp(version, known_fw_versions + i, 4)) + if (!memcmp(version, known_fw_versions + i, 2)) return 0; snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. " -- cgit v1.2.3 From 39d4ecdb711ba44e0aa0b2f3db74ed5ac97abe21 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 24 May 2013 11:38:24 +0100 Subject: ASoC: wm5110: Correct DSP4R Mixer control name Signed-off-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm5110.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c00480bdf824..ba38f0679662 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -190,7 +190,7 @@ ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP3R", ARIZONA_DSP3RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP4L", ARIZONA_DSP4LMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DSP5R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DSP4R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), -- cgit v1.2.3 From d3134e211e8db7fa833c40b5879fc022693e16c2 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 28 May 2013 15:41:57 +0530 Subject: ASoC: wm8994: use the correct pointer to get the control value Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1eb152cb1097..62dc30598084 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1031,7 +1031,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = codec->control_data; + struct wm8994 *control = wm8994->wm8994; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; int i; int dac; -- cgit v1.2.3 From 9767a58b8b2a0b153c246fb6306c7d48d51bb379 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 28 May 2013 12:52:08 +0100 Subject: ASoC: wm8994: Fix reporting of accessory removal on WM8958 During recent refactoring the code to report removal when MICDET reports an absent microphone was removed, causing problems for systems which rely solely on the MICDET for this functionality. Restore it. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 62dc30598084..b38382cc4b59 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3833,6 +3833,11 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) dev_dbg(codec->dev, "Ignoring removed jack\n"); return IRQ_HANDLED; } + } else if (!(reg & WM8958_MICD_STS)) { + snd_soc_jack_report(wm8994->micdet[0].jack, 0, + SND_JACK_MECHANICAL | SND_JACK_HEADSET | + wm8994->btn_mask); + goto out; } if (wm8994->mic_detecting) -- cgit v1.2.3 From 7d6898be8db92450ce7a0afcc4238680b9703e2b Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 28 May 2013 15:06:42 +0530 Subject: ASoC: wm8994: check for array index returned The array 'drc_cfg' of size 3 may use index value -22 (EINVAL) The array 'retune_mobile_cfg' of size 3 may use index value -22 (EINVAL) Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b38382cc4b59..dfd997aaadfc 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -383,6 +383,8 @@ static int wm8994_get_drc_enum(struct snd_kcontrol *kcontrol, struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int drc = wm8994_get_drc(kcontrol->id.name); + if (drc < 0) + return drc; ucontrol->value.enumerated.item[0] = wm8994->drc_cfg[drc]; return 0; @@ -488,6 +490,9 @@ static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol, struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int block = wm8994_get_retune_mobile_block(kcontrol->id.name); + if (block < 0) + return block; + ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block]; return 0; -- cgit v1.2.3 From 04d245b7899c020559402841d2f70ddd740a7704 Mon Sep 17 00:00:00 2001 From: Nicolas Schichan Date: Thu, 23 May 2013 16:53:02 +0200 Subject: ASoC: cs42l52: fix default value for MASTERA_VOL. The default register value for MASTERA_VOL is 0x00, the same as MASTERB_VOL. Signed-off-by: Nicolas Schichan Acked-by: Brian Austin Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 8465c1fa6bf9..030f53c96ec0 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -86,7 +86,7 @@ static const struct reg_default cs42l52_reg_defaults[] = { { CS42L52_BEEP_VOL, 0x00 }, /* r1D Beep Volume off Time */ { CS42L52_BEEP_TONE_CTL, 0x00 }, /* r1E Beep Tone Cfg. */ { CS42L52_TONE_CTL, 0x00 }, /* r1F Tone Ctl */ - { CS42L52_MASTERA_VOL, 0x88 }, /* r20 Master A Volume */ + { CS42L52_MASTERA_VOL, 0x00 }, /* r20 Master A Volume */ { CS42L52_MASTERB_VOL, 0x00 }, /* r21 Master B Volume */ { CS42L52_HPA_VOL, 0x00 }, /* r22 Headphone A Volume */ { CS42L52_HPB_VOL, 0x00 }, /* r23 Headphone B Volume */ -- cgit v1.2.3