From d0db84e713eaaccea2a435e1625fb3150b335f4a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 7 Aug 2012 15:37:47 +0300 Subject: ASoC: omap-mcbsp: Fix 6pin mux configuration The check for the mux_signal callback was wrong which prevents us to configure the 6pin port's FSR/CLKR signal mux. Reported-by: CF Adad Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown Cc: stable@vger.kernel.org (3.4+) --- sound/soc/omap/mcbsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 34835e8a9160..d33c48baaf71 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -745,7 +745,7 @@ int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) { const char *signal, *src; - if (mcbsp->pdata->mux_signal) + if (!mcbsp->pdata->mux_signal) return -EINVAL; switch (mux) { -- cgit v1.2.3 From 48a08bab3066a9452216f8c52e0d6f35566de04d Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 8 Aug 2012 00:47:21 -0300 Subject: ASoC: mxs: Fix the name of the SoC family SND_SOC_MXS_SGTL5000 is used on MXS boards, so fix the SoC family name. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/mxs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index 99a997f19bb9..b6fa77678d97 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -10,7 +10,7 @@ menuconfig SND_MXS_SOC if SND_MXS_SOC config SND_SOC_MXS_SGTL5000 - tristate "SoC Audio support for i.MX boards with sgtl5000" + tristate "SoC Audio support for MXS boards with sgtl5000" depends on I2C select SND_SOC_SGTL5000 help -- cgit v1.2.3 From 0865a75d4166bddc533fd50831829ceefb94f9b0 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 7 Aug 2012 16:51:34 -0300 Subject: ASoC: imx-ssi: Remove mono support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Playing a mono track results in incorrect playback rate, ie, the audio is played at a faster rate. Remove mono support in the driver by setting 'channes_min' to dual-channel and this allows mono tracks to be played correctly. Reported-by: Gaëtan Carlier Tested-by: Gaëtan Carlier Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/imx-ssi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 28dd76c7cb1c..81d7728cf67f 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -380,13 +380,14 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { - .channels_min = 1, + /* The SSI does not support monaural audio. */ + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, -- cgit v1.2.3 From ed36081350d2ca4f692f04c6a2d24d1e3a339da1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Aug 2012 17:12:52 +0200 Subject: ALSA: hda - Add codec->pcm_format_first flag Introduced a new flag to set up the PCM stream format at first before the stream_id and channel tag. Some codecs (e.g. CA0132) seem preferring this over stream_id -> format order. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 69 ++++++++++++++++++++++++++++++----------------- sound/pci/hda/hda_codec.h | 1 + 2 files changed, 46 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 88a9c20eb7a2..598b9e2d85e6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1386,6 +1386,44 @@ int snd_hda_codec_configure(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_codec_configure); +/* update the stream-id if changed */ +static void update_pcm_stream_id(struct hda_codec *codec, + struct hda_cvt_setup *p, hda_nid_t nid, + u32 stream_tag, int channel_id) +{ + unsigned int oldval, newval; + + if (p->stream_tag != stream_tag || p->channel_id != channel_id) { + oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); + newval = (stream_tag << 4) | channel_id; + if (oldval != newval) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, + newval); + p->stream_tag = stream_tag; + p->channel_id = channel_id; + } +} + +/* update the format-id if changed */ +static void update_pcm_format(struct hda_codec *codec, struct hda_cvt_setup *p, + hda_nid_t nid, int format) +{ + unsigned int oldval; + + if (p->format_id != format) { + oldval = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_STREAM_FORMAT, 0); + if (oldval != format) { + msleep(1); + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_STREAM_FORMAT, + format); + } + p->format_id = format; + } +} + /** * snd_hda_codec_setup_stream - set up the codec for streaming * @codec: the CODEC to set up @@ -1400,7 +1438,6 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, { struct hda_codec *c; struct hda_cvt_setup *p; - unsigned int oldval, newval; int type; int i; @@ -1413,29 +1450,13 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, p = get_hda_cvt_setup(codec, nid); if (!p) return; - /* update the stream-id if changed */ - if (p->stream_tag != stream_tag || p->channel_id != channel_id) { - oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - newval = (stream_tag << 4) | channel_id; - if (oldval != newval) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - newval); - p->stream_tag = stream_tag; - p->channel_id = channel_id; - } - /* update the format-id if changed */ - if (p->format_id != format) { - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_STREAM_FORMAT, 0); - if (oldval != format) { - msleep(1); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, - format); - } - p->format_id = format; - } + + if (codec->pcm_format_first) + update_pcm_format(codec, p, nid, format); + update_pcm_stream_id(codec, p, nid, stream_tag, channel_id); + if (!codec->pcm_format_first) + update_pcm_format(codec, p, nid, format); + p->active = 1; p->dirty = 0; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c422d330ca54..7fbc1bcaf1a9 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -861,6 +861,7 @@ struct hda_codec { unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */ unsigned int no_jack_detect:1; /* Machine has no jack-detection */ + unsigned int pcm_format_first:1; /* PCM format must be set first */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ int power_transition; /* power-state in transition */ -- cgit v1.2.3 From 55cf87fe0e9783e25f442be1e48b8319d86131ea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Aug 2012 17:15:55 +0200 Subject: ALSA: hda - Fix superfluous "-in" suffix from CA0132 capture items Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index d0d3540e39e7..2685590925ff 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -988,12 +988,12 @@ static void ca0132_config(struct hda_codec *codec) /* Mic-in */ spec->input_pins[0] = 0x12; - spec->input_labels[0] = "Mic-In"; + spec->input_labels[0] = "Mic"; spec->adcs[0] = 0x07; /* Line-In */ spec->input_pins[1] = 0x11; - spec->input_labels[1] = "Line-In"; + spec->input_labels[1] = "Line"; spec->adcs[1] = 0x08; spec->num_inputs = 2; } -- cgit v1.2.3 From 27ebeb0b1b5bb26908e485a7e8bd2ec30366ffef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Aug 2012 17:20:18 +0200 Subject: ALSA: hda - Use the standard PCM ops for CA0132 Now with the workaround using codec->pcm_format_first flag, we can clean up the home-baked codes in patch_ca0132.c. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 142 +++++++++---------------------------------- 1 file changed, 29 insertions(+), 113 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 2685590925ff..31512a0f1d07 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -464,50 +464,17 @@ exit: } /* - * PCM stuffs + * PCM callbacks */ -static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid, - u32 stream_tag, - int channel_id, int format) +static int ca0132_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { - unsigned int oldval, newval; - - if (!nid) - return; - - snd_printdd("ca0132_setup_stream: " - "NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n", - nid, stream_tag, channel_id, format); - - /* update the format-id if changed */ - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_STREAM_FORMAT, - 0); - if (oldval != format) { - msleep(20); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, - format); - } - - oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - newval = (stream_tag << 4) | channel_id; - if (oldval != newval) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - newval); - } -} - -static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) -{ - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); + struct ca0132_spec *spec = codec->spec; + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } -/* - * PCM callbacks - */ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -515,10 +482,8 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, + stream_tag, format, substream); } static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, @@ -526,92 +491,45 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dacs[0]); - - return 0; + return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } /* * Digital out */ -static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) +static int ca0132_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dig_out, stream_tag, 0, format); - - return 0; -} - -static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dig_out); - - return 0; + return snd_hda_multi_out_dig_open(codec, &spec->multiout); } -/* - * Analog capture - */ -static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo, +static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->adcs[substream->number], - stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, + stream_tag, format, substream); } -static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->adcs[substream->number]); - - return 0; -} - -/* - * Digital capture - */ -static int ca0132_dig_capture_pcm_prepare(struct hda_pcm_stream *hinfo, +static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dig_in, stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); } -static int ca0132_dig_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) +static int ca0132_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dig_in); - - return 0; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); } /* @@ -621,6 +539,7 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = { .channels_min = 2, .channels_max = 2, .ops = { + .open = ca0132_playback_pcm_open, .prepare = ca0132_playback_pcm_prepare, .cleanup = ca0132_playback_pcm_cleanup }, @@ -630,10 +549,6 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .ops = { - .prepare = ca0132_capture_pcm_prepare, - .cleanup = ca0132_capture_pcm_cleanup - }, }; static struct hda_pcm_stream ca0132_pcm_digital_playback = { @@ -641,6 +556,8 @@ static struct hda_pcm_stream ca0132_pcm_digital_playback = { .channels_min = 2, .channels_max = 2, .ops = { + .open = ca0132_dig_playback_pcm_open, + .close = ca0132_dig_playback_pcm_close, .prepare = ca0132_dig_playback_pcm_prepare, .cleanup = ca0132_dig_playback_pcm_cleanup }, @@ -650,10 +567,6 @@ static struct hda_pcm_stream ca0132_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .ops = { - .prepare = ca0132_dig_capture_pcm_prepare, - .cleanup = ca0132_dig_capture_pcm_cleanup - }, }; static int ca0132_build_pcms(struct hda_codec *codec) @@ -961,6 +874,9 @@ static void ca0132_config(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; + codec->pcm_format_first = 1; + codec->no_sticky_stream = 1; + /* line-outs */ cfg->line_outs = 1; cfg->line_out_pins[0] = 0x0b; /* front */ -- cgit v1.2.3 From 8e13fc1c5f694a6ae4032c7f94103c137136733f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Aug 2012 17:26:54 +0200 Subject: ALSA: hda - Add missing SPDIF I/O setup for CA0132 CA0132 driver had some codes to handle the S/PDIF I/O, but the actual setups of pins and converters were missing. Now the pins are added. Also, fixed a few points triggering invalid codec verbs and mixer elements since the digital I/O audio widgets on CA0132 have no amp. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 31512a0f1d07..9c0ec0a55bef 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -246,7 +246,7 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); } - if (dac) + if (dac && (get_wcaps(codec, dac) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, dac, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO); } @@ -261,7 +261,7 @@ static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc) AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)); } - if (adc) + if (adc && (get_wcaps(codec, adc) & AC_WCAP_IN_AMP)) snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)); } @@ -841,18 +841,16 @@ static int ca0132_build_controls(struct hda_codec *codec) spec->dig_out); if (err < 0) return err; - err = add_out_volume(codec, spec->dig_out, "IEC958"); + err = snd_hda_create_spdif_share_sw(codec, &spec->multiout); if (err < 0) return err; + /* spec->multiout.share_spdif = 1; */ } if (spec->dig_in) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in); if (err < 0) return err; - err = add_in_volume(codec, spec->dig_in, "IEC958"); - if (err < 0) - return err; } return 0; } @@ -912,6 +910,16 @@ static void ca0132_config(struct hda_codec *codec) spec->input_labels[1] = "Line"; spec->adcs[1] = 0x08; spec->num_inputs = 2; + + /* SPDIF I/O */ + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + cfg->dig_out_pins[0] = 0x0c; + cfg->dig_outs = 1; + cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; + spec->dig_in = 0x09; + cfg->dig_in_pin = 0x0e; + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; } static void ca0132_init_chip(struct hda_codec *codec) -- cgit v1.2.3 From 94c142a160d63edac0e1fca7848960dcf75dd2a9 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 9 Aug 2012 10:56:12 +0200 Subject: ALSA: hda - Fix pop noise in headphones on S3 for Asus X55A, X55V To turn off pin control for the pin was tested, and helped against this issue. BugLink: https://bugs.launchpad.net/bugs/1034779 Tested-by: Chih-Hsyuan Ho Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 80d90cb42853..430771776915 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1752,6 +1752,14 @@ static int via_suspend(struct hda_codec *codec) { struct via_spec *spec = codec->spec; vt1708_stop_hp_work(spec); + + if (spec->codec_type == VT1802) { + /* Fix pop noise on headphones */ + int i; + for (i = 0; i < spec->autocfg.hp_outs; i++) + snd_hda_set_pin_ctl(codec, spec->autocfg.hp_pins[i], 0); + } + return 0; } #endif -- cgit v1.2.3 From 0d624275720a4b01217693eb80d967a0d5f1f3a3 Mon Sep 17 00:00:00 2001 From: Vaibhav Bedia Date: Wed, 8 Aug 2012 20:40:31 +0530 Subject: ASoC: Davinci: McASP: Flush the FIFO before enabling FIFO should be flushed before it is enabled for the first time. This fixes the I/O errors reported by the ASoC core on a fresh boot Signed-off-by: Vaibhav Bedia Signed-off-by: Hebbar, Gururaja Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 95441bfc8190..ce5e5cd254dd 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -380,14 +380,20 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt) /* enable FIFO */ + if (dev->txnumevt) { /* enable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } mcasp_start_tx(dev); } else { - if (dev->rxnumevt) /* enable FIFO */ + if (dev->rxnumevt) { /* enable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } mcasp_start_rx(dev); } } -- cgit v1.2.3 From 8b5eae137b91cb2db15fe2c5a913cafde4629339 Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Thu, 9 Aug 2012 18:08:40 -0400 Subject: ASoC: bfin: fix memory leak in sport3 controller driver Signed-off-by: Scott Jiang Signed-off-by: Mark Brown --- sound/soc/blackfin/bf6xx-sport.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c index 318c5ba5360f..dfb744381c42 100644 --- a/sound/soc/blackfin/bf6xx-sport.c +++ b/sound/soc/blackfin/bf6xx-sport.c @@ -413,7 +413,14 @@ EXPORT_SYMBOL(sport_create); void sport_delete(struct sport_device *sport) { + if (sport->tx_desc) + dma_free_coherent(NULL, sport->tx_desc_size, + sport->tx_desc, 0); + if (sport->rx_desc) + dma_free_coherent(NULL, sport->rx_desc_size, + sport->rx_desc, 0); sport_free_resource(sport); + kfree(sport); } EXPORT_SYMBOL(sport_delete); -- cgit v1.2.3 From 52c0eee3329b08dfd912a59e0246e21026308301 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 30 Jul 2012 18:23:35 +0100 Subject: ASoC: wm8962: Don't duplicate bias level management in resume The core will bring the bias level up for us since we use idle_bias_off, duplicating this may be harmful. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 15 --------------- 1 file changed, 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index aa9ce9dd7d8a..ce6720073798 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3733,21 +3733,6 @@ static int wm8962_runtime_resume(struct device *dev) regcache_sync(wm8962->regmap); - regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, - WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, - WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); - - /* Bias enable at 2*50k for ramp */ - regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, - WM8962_VMID_SEL_MASK | WM8962_BIAS_ENA, - WM8962_BIAS_ENA | 0x180); - - msleep(5); - - /* VMID back to 2x250k for standby */ - regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, - WM8962_VMID_SEL_MASK, 0x100); - return 0; } -- cgit v1.2.3 From 15676937e6a7e98d37f4c1eaa0e7b3c111627fce Mon Sep 17 00:00:00 2001 From: Chris Rattray Date: Thu, 9 Aug 2012 10:10:54 +0100 Subject: ASoC: wm8994: Add missing dapm routes for WM8958 rev A Signed-off-by: Chris Rattray Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 04ef03175c51..6c9eeca85b95 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4038,6 +4038,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; case WM8958: if (wm8994->revision < 1) { + snd_soc_dapm_add_routes(dapm, wm8994_intercon, + ARRAY_SIZE(wm8994_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, ARRAY_SIZE(wm8994_revd_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, -- cgit v1.2.3 From d34e4e00adbbc91ff9fc96ed9a4e4b65161868da Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Aug 2012 15:47:15 +0200 Subject: ALSA: platform: Check CONFIG_PM_SLEEP instead of CONFIG_PM When CONFIG_PM is set but CONFIG_PM_SLEEP is unset, SIMPLE_DEV_PM_OPS() ignores the given functions, and this leads to compile warnings. For avoiding this, simply check CONFIG_PM_SLEEP instead of CONFIG_PM. Reported-by: Arnd Bergmann Signed-off-by: Takashi Iwai --- sound/arm/pxa2xx-ac97.c | 4 ++-- sound/atmel/abdac.c | 2 +- sound/atmel/ac97c.c | 2 +- sound/drivers/aloop.c | 2 +- sound/drivers/dummy.c | 2 +- sound/drivers/pcsp/pcsp.c | 4 ++-- sound/ppc/powermac.c | 2 +- 7 files changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 0d7b25e81643..4e1fda75c1c9 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -106,7 +106,7 @@ static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = { .prepare = pxa2xx_ac97_pcm_prepare, }; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int pxa2xx_ac97_do_suspend(struct snd_card *card) { @@ -243,7 +243,7 @@ static struct platform_driver pxa2xx_ac97_driver = { .driver = { .name = "pxa2xx-ac97", .owner = THIS_MODULE, -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP .pm = &pxa2xx_ac97_pm_ops, #endif }, diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index eb4ceb71123e..98554f4882b7 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -534,7 +534,7 @@ out_put_pclk: return retval; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int atmel_abdac_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index bf47025bdf45..3c8d3ba7ddfc 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -1134,7 +1134,7 @@ err_snd_card_new: return retval; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int atmel_ac97c_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 1128b35b2b05..5a34355e78e8 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1176,7 +1176,7 @@ static int __devexit loopback_remove(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int loopback_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index f7d3bfc6bca8..54bb6644a598 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -1064,7 +1064,7 @@ static int __devexit snd_dummy_remove(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_dummy_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 6ca59fc6dcb9..ef171295f6d4 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -199,7 +199,7 @@ static void pcsp_stop_beep(struct snd_pcsp *chip) pcspkr_stop_sound(); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int pcsp_suspend(struct device *dev) { struct snd_pcsp *chip = dev_get_drvdata(dev); @@ -212,7 +212,7 @@ static SIMPLE_DEV_PM_OPS(pcsp_pm, pcsp_suspend, NULL); #define PCSP_PM_OPS &pcsp_pm #else #define PCSP_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static void pcsp_shutdown(struct platform_device *dev) { diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index f5ceb6f282de..210cafe04890 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -143,7 +143,7 @@ static int __devexit snd_pmac_remove(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_pmac_driver_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); -- cgit v1.2.3 From 144dad99ef6ad10c8c8ebe787d08157c4a94201f Mon Sep 17 00:00:00 2001 From: James Ralston Date: Thu, 9 Aug 2012 09:38:59 -0700 Subject: ALSA: hda_intel: Add Device IDs for Intel Lynx Point-LP PCH This patch adds the Intel HD Audio Device IDs for the Intel Lynx Point-LP PCH Signed-off-by: James Ralston Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c8aced182fd1..60882c62f180 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -151,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, CPT}," "{Intel, PPT}," "{Intel, LPT}," + "{Intel, LPT_LP}," "{Intel, HPT}," "{Intel, PBG}," "{Intel, SCH}," @@ -3270,6 +3271,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + /* Lynx Point-LP */ + { PCI_DEVICE(0x8086, 0x9c20), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + /* Lynx Point-LP */ + { PCI_DEVICE(0x8086, 0x9c21), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0c0c), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | -- cgit v1.2.3 From fb099cb712e878b9eb4e78dd6b268312a0b2b50f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 9 Aug 2012 18:44:37 +0100 Subject: ASoC: core: Upgrade the severity of probe deferral errors to dev_err() In the past when ASoC had a custom probe deferral mechanism people complained about the logspam it generated and didn't want to know about the fact that we were doing probe deferral so all the error messages for it were at dev_dbg(), making diagnostics hard. Now that we have probe deferral as an accepted thing and it's generating log messages anyway there's no need to worry about this so upgrade the severity of all the probe deferral sources to dev_err() so that they are displayed by default. Also add one for missing aux_devs since there wasn't one. Reported-by: Russell King Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f81c5976b961..c501af6d8dbe 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -826,7 +826,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } if (!rtd->cpu_dai) { - dev_dbg(card->dev, "CPU DAI %s not registered\n", + dev_err(card->dev, "CPU DAI %s not registered\n", dai_link->cpu_dai_name); return -EPROBE_DEFER; } @@ -857,14 +857,14 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } if (!rtd->codec_dai) { - dev_dbg(card->dev, "CODEC DAI %s not registered\n", + dev_err(card->dev, "CODEC DAI %s not registered\n", dai_link->codec_dai_name); return -EPROBE_DEFER; } } if (!rtd->codec) { - dev_dbg(card->dev, "CODEC %s not registered\n", + dev_err(card->dev, "CODEC %s not registered\n", dai_link->codec_name); return -EPROBE_DEFER; } @@ -888,7 +888,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) rtd->platform = platform; } if (!rtd->platform) { - dev_dbg(card->dev, "platform %s not registered\n", + dev_err(card->dev, "platform %s not registered\n", dai_link->platform_name); return -EPROBE_DEFER; } @@ -1481,6 +1481,8 @@ static int soc_check_aux_dev(struct snd_soc_card *card, int num) return 0; } + dev_err(card->dev, "%s not registered\n", aux_dev->codec_name); + return -EPROBE_DEFER; } -- cgit v1.2.3 From de64c0ee7dbcbfbbe63bd9ea45783d87babc6452 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 10 Aug 2012 12:22:58 +0300 Subject: ALSA: cs46xx - signedness bug in snd_cs46xx_codec_read() This function returns its own error codes instead of normal negative error codes. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index f75f5ffdfdfb..a71d1c14a0f6 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -94,7 +94,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip, if (snd_BUG_ON(codec_index != CS46XX_PRIMARY_CODEC_INDEX && codec_index != CS46XX_SECONDARY_CODEC_INDEX)) - return -EINVAL; + return 0xffff; chip->active_ctrl(chip, 1); -- cgit v1.2.3 From 14bc9c6dc694e2d7930802f7afd275de25ef8394 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 10 Aug 2012 13:29:32 +0200 Subject: ALSA: hda - Fix panned "Beep Playback Switch" When "Beep Playback Switch" had a different value on left and right channels (such as muting left but not right, or vice versa), this could result in the right channel being ignored. This patch enables beep to be sounding from right channel only, and also give correct result back to userspace (e g amixer). Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 0bc2315b181d..d26ae65b43b7 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -237,10 +237,9 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep) { + if (beep && !beep->enabled) { ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[1] = - beep->enabled; + ucontrol->value.integer.value[1] = 0; return 0; } return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); @@ -252,9 +251,18 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep) - snd_hda_enable_beep_device(codec, - *ucontrol->value.integer.value); + if (beep) { + u8 chs = get_amp_channels(kcontrol); + int enable = 0; + long *valp = ucontrol->value.integer.value; + if (chs & 1) { + enable |= *valp; + valp++; + } + if (chs & 2) + enable |= *valp; + snd_hda_enable_beep_device(codec, enable); + } return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); -- cgit v1.2.3 From e037cb4a54e26b5f55f856e0e7445cfcfb2f3d31 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Fri, 10 Aug 2012 14:11:58 +0200 Subject: ALSA : hda - bug fix on checking the supported power states of a codec The return value of snd_hda_param_read() is -1 for an error, otherwise it's the supported power states of a codec. The supported power states is a 32-bit value. Bit 31 will be set to 1 if the codec supports EPSS, thus making "sup" negative. And the bit 28:5 is reserved as "0". So a negative value other than -1 shall be further checked. Please refer to High-Definition spec 7.3.4.12 "Supported Power States", thanks! Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 88a9c20eb7a2..629131ad7b8b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3497,7 +3497,7 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg { int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE); - if (sup < 0) + if (sup == -1) return false; if (sup & power_state) return true; -- cgit v1.2.3 From 61f5d61ef94d7082d96494e2a6dd79de2b4437d2 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Wed, 8 Aug 2012 11:34:43 +0530 Subject: ASoC: Samsung: Fix build error MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes the following build error: In file included from arch/arm/mach-exynos/include/mach/dma.h:24:0, from arch/arm/plat-samsung/include/plat/dma-ops.h:17, from arch/arm/plat-samsung/include/plat/dma.h:128, from sound/soc/samsung/pcm.c:23: arch/arm/plat-samsung/include/plat/dma-pl330.h:106:8: error: redefinition of ‘struct s3c2410_dma_client’ arch/arm/plat-samsung/include/plat/dma.h:40:8: note: originally defined here make[3]: *** [sound/soc/samsung/pcm.o] Error 1 Signed-off-by: Sachin Kamat Signed-off-by: Sachin Kamat Acked-by: Kukjin Kim Signed-off-by: Mark Brown --- sound/soc/samsung/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index b7b2a1f91425..89b064650f14 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -20,7 +20,7 @@ #include #include -#include +#include #include "dma.h" #include "pcm.h" -- cgit v1.2.3 From 088c820b732dbfd515fc66d459d5f5777f79b406 Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Mon, 13 Aug 2012 14:11:10 +0800 Subject: ALSA: hda - fix Copyright debug message As spec said, 1 indicates no copyright is asserted. Signed-off-by: Wang Xingchao Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_proc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 7e46258fc700..6894ec66258c 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -412,7 +412,7 @@ static void print_digital_conv(struct snd_info_buffer *buffer, if (digi1 & AC_DIG1_EMPHASIS) snd_iprintf(buffer, " Preemphasis"); if (digi1 & AC_DIG1_COPYRIGHT) - snd_iprintf(buffer, " Copyright"); + snd_iprintf(buffer, " Non-Copyright"); if (digi1 & AC_DIG1_NONAUDIO) snd_iprintf(buffer, " Non-Audio"); if (digi1 & AC_DIG1_PROFESSIONAL) -- cgit v1.2.3 From 14ebd8a8c15e9fed638120bdb93f1a814e13d6a9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 10 Aug 2012 15:40:12 +0100 Subject: ASoC: wm5102: Add missing input PGA routes Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 6537f16d383e..496ce9a9d8be 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -639,6 +639,15 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), -- cgit v1.2.3 From 17c3f7e8f3ef796a9db3b22f3797188d0e7ac88c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 10 Aug 2012 15:40:22 +0100 Subject: ASoC: wm5110: Add missing input PGA routes Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 8033f7065189..01ebbcc5c6a4 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -681,6 +681,18 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + + { "IN4L PGA", NULL, "IN4L" }, + { "IN4R PGA", NULL, "IN4R" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), -- cgit v1.2.3 From 12022a785328fdf61a3e1a4bc34db0098dabe839 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Aug 2012 16:28:36 +0100 Subject: ASoC: jack: Always notify full jack status Don't just notify for the bits we've updated, notify the full state of the jack otherwise users might get confused by misleading reports. Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 7f8b3b7428bb..0c172938b82a 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -103,7 +103,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) } /* Report before the DAPM sync to help users updating micbias status */ - blocking_notifier_call_chain(&jack->notifier, status, jack); + blocking_notifier_call_chain(&jack->notifier, jack->status, jack); snd_soc_dapm_sync(dapm); -- cgit v1.2.3 From 265d931a9e9a7e290faa5e2145f4b2ebf38ea84c Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 13 Aug 2012 17:10:46 +0200 Subject: ALSA: hda - Fix 'Beep Playback Switch' with no underlying mute switch Some Conexant devices (e g CX20590) have no mute capability on their Beep widgets. This patch makes sure we don't try setting mutes on those widgets. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index d26ae65b43b7..0849aac449f2 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -231,15 +231,22 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device); +static bool ctl_has_mute(struct snd_kcontrol *kcontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + return query_amp_caps(codec, get_amp_nid(kcontrol), + get_amp_direction(kcontrol)) & AC_AMPCAP_MUTE; +} + /* get/put callbacks for beep mute mixer switches */ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep && !beep->enabled) { + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[1] = 0; + ucontrol->value.integer.value[1] = beep->enabled; return 0; } return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); @@ -263,6 +270,8 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, enable |= *valp; snd_hda_enable_beep_device(codec, enable); } + if (!ctl_has_mute(kcontrol)) + return 0; return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); -- cgit v1.2.3 From 3bdcff70b6cd049e6f4437b955850f5db83653cc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 14 Aug 2012 17:42:11 +0200 Subject: ALSA: lx6464es: Add a missing error check Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=44541 Signed-off-by: Takashi Iwai --- sound/pci/lx6464es/lx6464es.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index d1ab43706735..5579b08bb35b 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -851,6 +851,8 @@ static int __devinit lx_pcm_create(struct lx6464es *chip) /* hardcoded device name & channel count */ err = snd_pcm_new(chip->card, (char *)card_name, 0, 1, 1, &pcm); + if (err < 0) + return err; pcm->private_data = chip; -- cgit v1.2.3 From e9ba389c5ffc4dd29dfe17e00e48877302111135 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Aug 2012 12:32:00 +0200 Subject: ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream A PCM capture stream on usb-audio causes a scheduling-while-atomic BUG, as reported in the bugzilla entry below. It's because snd_usb_endpoint_start() is called at first at trigger START for a capture stream, and this function contains the left-over EP deactivation codes. The problem doesn't happen for a playback stream because the function is called at PCM prepare time, which can sleep. This patch fixes the BUG by moving the EP deactivation code into the PCM prepare callback. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=46011 Cc: [v3.5+] Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 4 ---- sound/usb/pcm.c | 3 +++ 2 files changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 0f647d22cb4a..c41181202688 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -821,10 +821,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) if (++ep->use_count != 1) return 0; - /* just to be sure */ - deactivate_urbs(ep, 0, 1); - wait_clear_urbs(ep); - ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a1298f379428..62ec808ed792 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -544,6 +544,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->last_frame_number = 0; runtime->delay = 0; + /* clear the pending deactivation on the target EPs */ + deactivate_endpoints(subs); + /* for playback, submit the URBs now; otherwise, the first hwptr_done * updates for all URBs would happen at the same time when starting */ if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) -- cgit v1.2.3 From 5e68fb3cab23b327e9f15803607e697d7eea1966 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 16 Aug 2012 14:11:09 +0200 Subject: ALSA: hda - Don't send invalid volume knob command on IDT 92hd75bxx Instead of blindly initializing a volume knob widget, first check that there actually is a volume knob widget. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 94040ccf8e8f..ea5775a1a7db 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4272,7 +4272,8 @@ static int stac92xx_init(struct hda_codec *codec) unsigned int gpio; int i; - snd_hda_sequence_write(codec, spec->init); + if (spec->init) + snd_hda_sequence_write(codec, spec->init); /* power down adcs initially */ if (spec->powerdown_adcs) @@ -5748,7 +5749,6 @@ again: /* fallthru */ case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5773,7 +5773,6 @@ again: spec->stream_delay = 40; /* 40 milliseconds */ /* disable VSW */ - spec->init = stac92hd71bxx_core_init; unmute_init++; snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); @@ -5788,7 +5787,6 @@ again: /* fallthru */ default: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5796,6 +5794,9 @@ again: break; } + if (get_wcaps_type(get_wcaps(codec, 0x28)) == AC_WID_VOL_KNB) + spec->init = stac92hd71bxx_core_init; + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); -- cgit v1.2.3 From 939d5044b117302cabdd30833685d9f214e9bff6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Aug 2012 13:08:23 +0100 Subject: ASoC: wm5102: Remove DRC2 It will be removed from future device revisions. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 16 ---------------- 1 file changed, 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 496ce9a9d8be..e33d327396ad 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -128,13 +128,9 @@ SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE), SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), -SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5, - ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA), ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), @@ -236,8 +232,6 @@ ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); -ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE); -ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); @@ -349,10 +343,6 @@ SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0, - NULL, 0), -SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0, - NULL, 0), SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, NULL, 0), @@ -466,8 +456,6 @@ ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), -ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"), -ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"), ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), @@ -553,8 +541,6 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"), { name, "EQ4", "EQ4" }, \ { name, "DRC1L", "DRC1L" }, \ { name, "DRC1R", "DRC1R" }, \ - { name, "DRC2L", "DRC2L" }, \ - { name, "DRC2R", "DRC2R" }, \ { name, "LHPF1", "LHPF1" }, \ { name, "LHPF2", "LHPF2" }, \ { name, "LHPF3", "LHPF3" }, \ @@ -684,8 +670,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), - ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"), - ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"), ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), -- cgit v1.2.3 From ccf795847a38235ee4a56a24129ce75147d6ba8f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Aug 2012 22:36:04 +0100 Subject: ASoC: wm9712: Fix microphone source selection Currently the microphone input source is not selectable as while there is a DAPM widget it's not connected to anything so it won't be properly instantiated. Add something more correct for the input structure to get things going, even though it's not hooked into the rest of the routing map and so won't actually achieve anything except allowing the relevant register bits to be written. Reported-by: Christop Fritz Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm9712.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index f16fb361a4eb..fd74b8843d34 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -272,7 +272,7 @@ SOC_DAPM_ENUM("Route", wm9712_enum[9]); /* Mic select */ static const struct snd_kcontrol_new wm9712_mic_src_controls = -SOC_DAPM_ENUM("Route", wm9712_enum[7]); +SOC_DAPM_ENUM("Mic Source Select", wm9712_enum[7]); /* diff select */ static const struct snd_kcontrol_new wm9712_diff_sel_controls = @@ -291,7 +291,9 @@ SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0, &wm9712_capture_selectl_controls), SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0, &wm9712_capture_selectr_controls), -SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0, +SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0, + &wm9712_mic_src_controls), +SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0, &wm9712_mic_src_controls), SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0, &wm9712_diff_sel_controls), @@ -319,6 +321,7 @@ SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0), SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0), SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0), +SND_SOC_DAPM_PGA("Differential Mic", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1), SND_SOC_DAPM_OUTPUT("MONOOUT"), SND_SOC_DAPM_OUTPUT("HPOUTL"), @@ -379,6 +382,18 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = { {"Mic PGA", NULL, "MIC1"}, {"Mic PGA", NULL, "MIC2"}, + /* microphones */ + {"Differential Mic", NULL, "MIC1"}, + {"Differential Mic", NULL, "MIC2"}, + {"Left Mic Select Source", "Mic 1", "MIC1"}, + {"Left Mic Select Source", "Mic 2", "MIC2"}, + {"Left Mic Select Source", "Stereo", "MIC1"}, + {"Left Mic Select Source", "Differential", "Differential Mic"}, + {"Right Mic Select Source", "Mic 1", "MIC1"}, + {"Right Mic Select Source", "Mic 2", "MIC2"}, + {"Right Mic Select Source", "Stereo", "MIC2"}, + {"Right Mic Select Source", "Differential", "Differential Mic"}, + /* left capture selector */ {"Left Capture Select", "Mic", "MIC1"}, {"Left Capture Select", "Speaker Mixer", "Speaker Mixer"}, -- cgit v1.2.3 From 28c42c28309244d0b15d1b385e33429d59997679 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Jul 2012 18:37:28 +0100 Subject: ASoC: wm9712: Fix inverted capture volume The capture volume increases with the register value so it shouldn't be flagged as inverted. Reported-by: Christoph Fritz Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index fd74b8843d34..c6d2076a796b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -148,7 +148,7 @@ SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1), SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1), SOC_ENUM("Capture Volume Steps", wm9712_enum[6]), -SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1), +SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 0), SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0), SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv), -- cgit v1.2.3 From 94f3ec6b2222eb5c0af0c784f0656ff5b909d870 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Sat, 18 Aug 2012 18:55:15 +0300 Subject: sound: oss/sb_audio: prevent divide by zero bug Speed comes from get_user() in audio_ioctl(). We use it to set the "s" variable before clamping it to valid values so it could lead to a divide by zero bug. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/oss/sb_audio.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/sb_audio.c b/sound/oss/sb_audio.c index 733b014ec7d1..b2b3c014221a 100644 --- a/sound/oss/sb_audio.c +++ b/sound/oss/sb_audio.c @@ -575,13 +575,15 @@ static int jazz16_audio_set_speed(int dev, int speed) if (speed > 0) { int tmp; - int s = speed * devc->channels; + int s; if (speed < 5000) speed = 5000; if (speed > 44100) speed = 44100; + s = speed * devc->channels; + devc->tconst = (256 - ((1000000 + s / 2) / s)) & 0xff; tmp = 256 - devc->tconst; -- cgit v1.2.3 From aaf265c22e48f10c94ad04d23b6ab0c88f554d35 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:58 +0200 Subject: ALSA: sound/atmel/abdac.c: fix error return code Initialize retval before returning from a failed call to ioremap. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/atmel/abdac.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index 98554f4882b7..277ebce23a45 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -452,6 +452,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev) dac->regs = ioremap(regs->start, resource_size(regs)); if (!dac->regs) { dev_dbg(&pdev->dev, "could not remap register memory\n"); + retval = -ENOMEM; goto out_free_card; } -- cgit v1.2.3 From 0c23e46eb4878422c25351ff54ab0fe80c643809 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:57 +0200 Subject: ALSA: sound/atmel/ac97c.c: fix error return code In the first case, the second test of whether retval is negative is redundant. It is dropped and the previous and subsequent tests are combined. In the second case, add an initialization of retval on failure of ioremap. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/atmel/ac97c.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 3c8d3ba7ddfc..9052aff37f64 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -278,14 +278,9 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, if (retval < 0) return retval; /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (cpu_is_at32ap7000()) { - if (retval < 0) - return retval; - /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (retval == 1) - if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.rx_chan); - } + if (cpu_is_at32ap7000() && retval == 1) + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); /* Set restrictions to params. */ mutex_lock(&opened_mutex); @@ -980,6 +975,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) if (!chip->regs) { dev_dbg(&pdev->dev, "could not remap register memory\n"); + retval = -ENOMEM; goto err_ioremap; } -- cgit v1.2.3 From 4d8ce1c9966663bad69e738952179f3cc52710bf Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:56 +0200 Subject: ALSA: sound/pci/ctxfi/ctatc.c: fix error return code Initialize err before returning on failure, as done elsewhere in the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 8e40262d4117..2f6e9c762d3f 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1725,8 +1725,10 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, atc_connect_resources(atc); atc->timer = ct_timer_new(atc); - if (!atc->timer) + if (!atc->timer) { + err = -ENOMEM; goto error1; + } err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, atc, &ops); if (err < 0) -- cgit v1.2.3 From ae970eb45d8a1ea4506be23c3f776225b9575d0e Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:55 +0200 Subject: ALSA: sound/pci/sis7019.c: fix error return code Initialize rc before returning on failure, as done elsewhere in the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/pci/sis7019.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 512434efcc31..805ab6e9a78f 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1377,8 +1377,9 @@ static int __devinit sis_chip_create(struct snd_card *card, if (rc) goto error_out_cleanup; - if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, - sis)) { + rc = request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, + sis); + if (rc) { dev_err(&pci->dev, "unable to allocate irq %d\n", sis->irq); goto error_out_cleanup; } -- cgit v1.2.3 From b17cbdd85f84c8323189416da6e9701d2793b0e5 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:54 +0200 Subject: ALSA: sound/pci/rme9652/hdspm.c: fix error return code Convert a nonnegative error return code to a negative one, as returned elsewhere in the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index b8ac8710f47f..b12308b5ba2a 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6585,7 +6585,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card, snd_printk(KERN_ERR "HDSPM: " "unable to kmalloc Mixer memory of %d Bytes\n", (int)sizeof(struct hdspm_mixer)); - return err; + return -ENOMEM; } hdspm->port_names_in = NULL; -- cgit v1.2.3 From c86b93628e5649fd7bb0574b570a51b2b02d586c Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:59 +0200 Subject: ALSA: sound/ppc/snd_ps3.c: fix error return code Initialize ret before returning on failure, as done elsewhere in the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/ppc/snd_ps3.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 1aa52eff526a..9b18b5243a56 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -1040,6 +1040,7 @@ static int __devinit snd_ps3_driver_probe(struct ps3_system_bus_device *dev) GFP_KERNEL); if (!the_card.null_buffer_start_vaddr) { pr_info("%s: nullbuffer alloc failed\n", __func__); + ret = -ENOMEM; goto clean_preallocate; } pr_debug("%s: null vaddr=%p dma=%#llx\n", __func__, -- cgit v1.2.3 From c41999a23929f30808bae6009d8065052d4d73fd Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 20 Aug 2012 11:17:00 +0200 Subject: ALSA: hda - don't create dysfunctional mixer controls for ca0132 It's possible that these amps are settable somehow, e g through secret codec verbs, but for now, don't create the controls (as they won't be working anyway, and cause errors in amixer). Cc: stable@kernel.org BugLink: https://bugs.launchpad.net/bugs/1038651 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 9c0ec0a55bef..49750a96d649 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -275,6 +275,10 @@ static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type); + if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_MUTE) == 0) { + snd_printdd("Skipping '%s %s Switch' (no mute on node 0x%x)\n", pfx, dirstr[dir], nid); + return 0; + } sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } @@ -286,6 +290,10 @@ static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type); + if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_NUM_STEPS) == 0) { + snd_printdd("Skipping '%s %s Volume' (no amp on node 0x%x)\n", pfx, dirstr[dir], nid); + return 0; + } sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]); return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } -- cgit v1.2.3 From 535b6c51fe8293c88ce919cdfc4390c67a1cb6d1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Aug 2012 21:25:22 +0200 Subject: ALSA: hda - Fix leftover codec->power_transition When the codec turn-on operation is canceled by the immediate power-on, the driver left the power_transition flag as is. This caused the persistent avoidance of power-save behavior. Cc: [v3.5+] Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c3077d5dec6e..f560051a949e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4454,6 +4454,8 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) * then there is no need to go through power up here. */ if (codec->power_on) { + if (codec->power_transition < 0) + codec->power_transition = 0; spin_unlock(&codec->power_lock); return; } -- cgit v1.2.3 From 53e1719f3da0f095b8db1461bd12dd79f3246b84 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Mon, 20 Aug 2012 21:50:13 +0200 Subject: ALSA: snd-als100: fix suspend/resume snd_card_als100_probe() does not set pcm field in struct snd_sb. As a result, PCM is not suspended and applications don't know that they need to resume the playback. Tested with Labway A381-F20 card (ALS120). Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/isa/als100.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/isa/als100.c b/sound/isa/als100.c index 2d67c78c9f4b..f7cdaf51512d 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -233,7 +233,7 @@ static int __devinit snd_card_als100_probe(int dev, irq[dev], dma8[dev], dma16[dev]); } - if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) { + if ((error = snd_sb16dsp_pcm(chip, 0, &chip->pcm)) < 0) { snd_card_free(card); return error; } -- cgit v1.2.3 From e93c7d1bc350189511d32cec2f0af79c30e7fa47 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 30 Aug 2012 17:06:15 +0300 Subject: ASoC: omap-mcbsp: Fix compilation error due to leftover code Part of commit (which patches sound/soc/omap/mcbsp.c file): 8fef626 ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code since the tree where it has been applied did not had the earlier patch: d0db84e ASoC: omap-mcbsp: Fix 6pin mux configuration which changed code around omap_mcbsp_6pin_src_mux(). Because of the missing part from 8fef626 the sound/soc/omap/mcbsp.c does not compile in linux-next. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/mcbsp.c | 31 ------------------------------- 1 file changed, 31 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 935ccf633976..bc06175e6367 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -762,37 +762,6 @@ int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) } -int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) -{ - const char *signal, *src; - - if (!mcbsp->pdata->mux_signal) - return -EINVAL; - - switch (mux) { - case CLKR_SRC_CLKR: - signal = "clkr"; - src = "clkr"; - break; - case CLKR_SRC_CLKX: - signal = "clkr"; - src = "clkx"; - break; - case FSR_SRC_FSR: - signal = "fsr"; - src = "fsr"; - break; - case FSR_SRC_FSX: - signal = "fsr"; - src = "fsx"; - break; - default: - return -EINVAL; - } - - return mcbsp->pdata->mux_signal(mcbsp->dev, signal, src); -} - #define max_thres(m) (mcbsp->pdata->buffer_size) #define valid_threshold(m, val) ((val) <= max_thres(m)) #define THRESHOLD_PROP_BUILDER(prop) \ -- cgit v1.2.3