October 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: Remove unused function declaration in imx-ssi.h
by Axel Lin 05 Oct '11

05 Oct '11

These functions are removed in commit f0fba2ad "ASoC: multi-component - ASoC Multi-Component Support". Let's remove the leftover function declaration in header file. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/imx/imx-ssi.h | 6 ------ 1 files changed, 0 insertions(+), 6 deletions(-) diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h index 0a84cec..1072dfb 100644 --- a/sound/soc/imx/imx-ssi.h +++ b/sound/soc/imx/imx-ssi.h @@ -218,12 +218,6 @@ struct imx_ssi { struct platform_device *soc_platform_pdev_fiq; }; -struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev, - struct imx_ssi *ssi); -void imx_ssi_fiq_exit(struct platform_device *pdev, struct imx_ssi *ssi); -struct snd_soc_platform *imx_ssi_dma_mx2_init(struct platform_device *pdev, - struct imx_ssi *ssi); - int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); int imx_pcm_new(struct snd_soc_pcm_runtime *rtd); void imx_pcm_free(struct snd_pcm *pcm); -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: Remove impossible case from wm8994_hw_params
by Axel Lin 05 Oct '11

05 Oct '11

We set hw_params callback for wm8994_aif3_dai_ops to wm8994_aif3_hw_params. Thus no need to check wm8994-aif3 in wm8994_hw_params. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8994.c | 9 --------- 1 files changed, 0 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d5b69c0..94a8db2 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2210,7 +2210,6 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - struct wm8994 *control = codec->control_data; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int aif1_reg; int aif2_reg; @@ -2253,14 +2252,6 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, dev_dbg(codec->dev, "AIF2 using split LRCLK\n"); } break; - case 3: - switch (control->type) { - case WM8958: - aif1_reg = WM8958_AIF3_CONTROL_1; - break; - default: - return 0; - } default: return -EINVAL; } -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: Return early with -EINVAL if invalid dai format is detected
by Axel Lin 05 Oct '11

05 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Cc: Timur Tabi <timur(a)freescale.com> Cc: Arnaud Patard <arnaud.patard(a)rtp-net.org> --- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs42l51.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 5830c93..e645d67 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -271,7 +271,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, break; default: dev_err(codec->dev, "invalid dai format\n"); - ret = -EINVAL; + return -EINVAL; } /* set master/slave audio interface */ diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 286878d..779294b 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -298,7 +298,7 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai, break; default: dev_err(codec->dev, "invalid DAI format\n"); - ret = -EINVAL; + return -EINVAL; } switch (format & SND_SOC_DAIFMT_MASTER_MASK) { -- 1.7.4.1

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[alsa-devel] [PATCH v2 00/10] Combining snd_soc_*_volsw/volsw_2r
by Peter Ujfalusi 05 Oct '11

05 Oct '11

Hello Mark, Changes since v1 (RFC series): - rebased on git://opensource.wolfsonmicro.com/linux-2.6-asoc, for-3.2 branch - twl6040: warning added for the custom volsw calls in case of unexpected parameter - snd_soc_volsw_is_stereo returns type changed to bool - patch 06: optimized the if() cases in order to reduce the number of comparison in certain cases - patch 06: Use the (reg == reg2) comparison instead of (shift != rshitf) to differentiate between single or 2r type of stereo controls - patch 07: Use the (reg == reg2) comparison instead of (shift != rshitf) to differentiate between single or 2r type of stereo controls Intro mail from v1: This series will combine the snd_soc_*_volsw, and snd_soc_*_volsw_2r callbacks. We are going to drop the volsw_r2 callbacks, and merge the support to the _volsw functions. In order to achieve this, the macros need to be adjusted, and also the drivers, which were not using the macros to construct the soc_mixer_control needed to be changed, so I'm not going to introduce any breakage, inconsistency. This has been suggested in this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-September/044287.… By Mark in this mail (I hope I did not misunderstood it): http://mailman.alsa-project.org/pipermail/alsa-devel/2011-September/044343.… The use of the existing macros will not change, but underneath we are not going to have two sets of functions for single, or double mixer controls. I think the last 3 patch can be squashed, I did not wanted to touch the core, and a codec driver in a same patch. Regards, Peter --- Peter Ujfalusi (10): ASoC: Consolidate use of controls with custom get/put function ASoC: twl6040: Simplify custom get_volsw callback ASoC: twl6040: Prepare for core put_volsw/volsw_2r merger ASoC: core: Change SOC_SINGLE/DOUBLE_VALUE representation ASoC: core: Combine snd_soc_info_volsw/info_volsw_2r functions ASoC: core: Combine snd_soc_get_volsw/get_volsw_2r functions ASoC: core: Combine snd_soc_put_volsw/put_volsw_2r functions ASoC: twl6040: Simply call snd_soc_put_volsw form the custom code ASoC: core: Remove snd_soc_put_volsw_2r definition ASoC: twl6040: Warn user in twl6040_put_volsw for error case include/sound/soc.h | 33 +++++--- sound/soc/codecs/tlv320aic23.c | 14 +--- sound/soc/codecs/twl4030.c | 48 +++--------- sound/soc/codecs/twl6040.c | 64 ++++------------ sound/soc/codecs/wm8350.c | 39 ++++------ sound/soc/codecs/wm8580.c | 36 +++------ sound/soc/codecs/wm_hubs.c | 18 +--- sound/soc/soc-core.c | 162 +++++++++------------------------------- 8 files changed, 117 insertions(+), 297 deletions(-) -- 1.7.7

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[PATCH] Add "Line In" input jack constants
by David Henningsson 05 Oct '11

05 Oct '11

Similar to Line Out, these constants form the base for future patches enabling input jack reporting for Line in jacks. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/linux/input.h | 1 + include/sound/jack.h | 1 + sound/core/jack.c | 1 + sound/pci/hda/hda_codec.c | 2 ++ 4 files changed, 5 insertions(+), 0 deletions(-) diff --git a/include/linux/input.h b/include/linux/input.h index a637e78..a514fb8 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -814,6 +814,7 @@ struct input_keymap_entry { #define SW_KEYPAD_SLIDE 0x0a /* set = keypad slide out */ #define SW_FRONT_PROXIMITY 0x0b /* set = front proximity sensor active */ #define SW_ROTATE_LOCK 0x0c /* set = rotate locked/disabled */ +#define SW_LINEIN_INSERT 0x0d /* set = inserted */ #define SW_MAX 0x0f #define SW_CNT (SW_MAX+1) diff --git a/include/sound/jack.h b/include/sound/jack.h index c140fc7..63c7907 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -42,6 +42,7 @@ enum snd_jack_types { SND_JACK_MECHANICAL = 0x0008, /* If detected separately */ SND_JACK_VIDEOOUT = 0x0010, SND_JACK_AVOUT = SND_JACK_LINEOUT | SND_JACK_VIDEOOUT, + SND_JACK_LINEIN = 0x0020, /* Kept separate from switches to facilitate implementation */ SND_JACK_BTN_0 = 0x4000, diff --git a/sound/core/jack.c b/sound/core/jack.c index 53b53e9..240a3e1 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -30,6 +30,7 @@ static int jack_switch_types[] = { SW_LINEOUT_INSERT, SW_JACK_PHYSICAL_INSERT, SW_VIDEOOUT_INSERT, + SW_LINEIN_INSERT, }; static int snd_jack_dev_free(struct snd_device *device) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e3db196..8b046a1 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5264,6 +5264,8 @@ static const char *get_jack_default_name(struct hda_codec *codec, hda_nid_t nid, return "Mic"; case SND_JACK_LINEOUT: return "Line-out"; + case SND_JACK_LINEIN: + return "Line-in"; case SND_JACK_HEADSET: return "Headset"; case SND_JACK_VIDEOOUT: -- 1.7.5.4 --------------050706010401040202030702--

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[alsa-devel] [PATCH] ASoC: Use dapm_mark_dirty() for new DAPM widgets for consistency
by Mark Brown 05 Oct '11

05 Oct '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 82d93bf..8711aab 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2185,7 +2185,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) w->new = 1; - list_add(&w->dirty, &(w->dapm->card->dapm_dirty)); + dapm_mark_dirty(w, "new widget"); dapm_debugfs_add_widget(w); } -- 1.7.6.3

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[alsa-devel] [PATCH] ALSA: HDA: Fix DAC assignment for secondary headphone on Sigmatel/IDT
by David Henningsson 05 Oct '11

05 Oct '11

This patch should be safe enough for 3.2, but do you think it is safe enough for stable as well? If we run out of DACs when trying to assign a DAC to a secondary headphone, prefer the DAC of the first headphone to the primary (usually line out) DAC. BugLink: http://bugs.launchpad.net/bugs/845275 (there are codec info at this link as well) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] [RFC][PATCH] ASoC: Fix sync reg_cache with the hardware in wm8990_resume
by Axel Lin 05 Oct '11

05 Oct '11

Current code has off-by-one bug of checking WM8990_RESET. Besides, according to wm8990 datasheet (2-wire serial control mode): A control word consists of 24 bits. The first 8 bits are address bits, the remaining 16 bits are data bits. Current code uses 7 bits address bits and 9 bits data bits which is incorrect. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- I'd appreciate if someone can test it. I'm a little bit confused if this fix is correct, why there is no bug report so far. If I'm wrong, please kindly let me know. Thanks, Axel sound/soc/codecs/wm8990.c | 11 ++++++----- 1 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 100aeee..5546f5a 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1320,16 +1320,17 @@ static int wm8990_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8990_resume(struct snd_soc_codec *codec) { int i; - u8 data[2]; + u8 data[3]; u16 *cache = codec->reg_cache; /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8990_reg); i++) { - if (i + 1 == WM8990_RESET) + if (i == WM8990_RESET) continue; - data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); + data[0] = i; + data[1] = (cache[i] & 0xFF00) >> 8; + data[2] = cache[i] & 0x00FF; + codec->hw_write(codec->control_data, data, 3); } wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- 1.7.4.1

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[PATCH] ALSA: HDA: Fix DAC assignment for secondary headphone on Sigmatel/IDT
by David Henningsson 05 Oct '11

05 Oct '11

If we run out of DACs when trying to assign a DAC to a secondary headphone, prefer the DAC of the first headphone to the primary (usually line out) DAC. BugLink: http://bugs.launchpad.net/bugs/845275 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 18 +++++++++++++++--- 1 files changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dd977b6..1e0b338 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2972,8 +2972,9 @@ static int check_all_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; int j, conn_len; - hda_nid_t conn[HDA_MAX_CONNECTIONS]; + hda_nid_t conn[HDA_MAX_CONNECTIONS], fallback_dac; unsigned int wcaps, wtype; conn_len = snd_hda_get_connections(codec, nid, conn, @@ -3001,10 +3002,21 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) return conn[j]; } } - /* if all DACs are already assigned, connect to the primary DAC */ + + /* if all DACs are already assigned, connect to the primary DAC, + unless we're assigning a secondary headphone */ + fallback_dac = spec->multiout.dac_nids[0]; + if (spec->multiout.hp_nid) { + for (j = 0; j < cfg->hp_outs; j++) + if (cfg->hp_pins[j] == nid) { + fallback_dac = spec->multiout.hp_nid; + break; + } + } + if (conn_len > 1) { for (j = 0; j < conn_len; j++) { - if (conn[j] == spec->multiout.dac_nids[0]) { + if (conn[j] == fallback_dac) { snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, j); break; -- 1.7.5.4 --------------040409070004050505030701--

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[alsa-devel] [PATCH 1/4] ASoC: Add verbose debugging showing why widgets get marked dirty
by Mark Brown 05 Oct '11

05 Oct '11

Help diagnose why we're checking widgets by providing some logging when we first dirty them. This should possibly be a trace point if it's useful but can be absurdly verbose if enabled, we can always change it later if desired. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 26 +++++++++++++++----------- 1 files changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 214a709..e6a0882 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -124,10 +124,13 @@ static bool dapm_dirty_widget(struct snd_soc_dapm_widget *w) return !list_empty(&w->dirty); } -static void dapm_mark_dirty(struct snd_soc_dapm_widget *w) +static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) { - if (!dapm_dirty_widget(w)) + if (!dapm_dirty_widget(w)) { + dev_vdbg(w->dapm->dev, "Marking %s dirty due to %s\n", + w->name, reason); list_add_tail(&w->dirty, &w->dapm->card->dapm_dirty); + } } /* create a new dapm widget */ @@ -1227,7 +1230,7 @@ static void dapm_widget_set_peer_power(struct snd_soc_dapm_widget *peer, /* If the peer is already in the state we're moving to then we * won't have an impact on it. */ if (power != peer->power) - dapm_mark_dirty(peer); + dapm_mark_dirty(peer, "peer state change"); } static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, @@ -1624,16 +1627,17 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, /* we now need to match the string in the enum to the path */ if (!(strcmp(path->name, e->texts[mux]))) { path->connect = 1; /* new connection */ - dapm_mark_dirty(path->source); + dapm_mark_dirty(path->source, "mux connection"); } else { if (path->connect) - dapm_mark_dirty(path->source); + dapm_mark_dirty(path->source, + "mux disconnection"); path->connect = 0; /* old connection must be powered down */ } } if (found) { - dapm_mark_dirty(widget); + dapm_mark_dirty(widget, "mux change"); dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); } @@ -1660,11 +1664,11 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, /* found, now check type */ found = 1; path->connect = connect; - dapm_mark_dirty(path->source); + dapm_mark_dirty(path->source, "mixer connection"); } if (found) { - dapm_mark_dirty(widget); + dapm_mark_dirty(widget, "mixer update"); dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); } @@ -1810,7 +1814,7 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, w->connected = status; if (status == 0) w->force = 0; - dapm_mark_dirty(w); + dapm_mark_dirty(w, "pin configuration"); return 0; } @@ -2699,7 +2703,7 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, dev_vdbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { - dapm_mark_dirty(w); + dapm_mark_dirty(w, "stream event"); switch(event) { case SND_SOC_DAPM_STREAM_START: w->active = 1; @@ -2789,7 +2793,7 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, dev_dbg(w->dapm->dev, "dapm: force enable pin %s\n", pin); w->connected = 1; w->force = 1; - dapm_mark_dirty(w); + dapm_mark_dirty(w, "force enable"); return 0; } -- 1.7.6.3

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