March 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] src/pcm/pcm_ladspa.c: Delay LADSPA plugin activate call
by Matthias Larisch 03 Apr '14

03 Apr '14

Some LADSPA Plugins rely on connected control ports on activate call. While this is not okay by spec, the spec also encourages the activate call happening as late as possible. Signed-off-by: Matthias Larisch <mail(a)matthias-larisch.de> --- src/pcm/pcm_ladspa.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c index 9ce5242..7d1e3df 100644 --- a/src/pcm/pcm_ladspa.c +++ b/src/pcm/pcm_ladspa.c @@ -645,8 +645,6 @@ static int snd_pcm_ladspa_allocate_instances(snd_pcm_t *pcm, snd_pcm_ladspa_t *l return -EINVAL; } list_add_tail(&instance->list, &plugin->instances); - if (plugin->desc->activate) - plugin->desc->activate(instance->handle); if (plugin->policy == SND_PCM_LADSPA_POLICY_DUPLICATE) { err = snd_pcm_ladspa_connect_plugin_duplicate(plugin, &plugin->input, &plugin->output, instance, idx); if (err < 0) { @@ -664,6 +662,8 @@ static int snd_pcm_ladspa_allocate_instances(snd_pcm_t *pcm, snd_pcm_ladspa_t *l assert(err >= 0); err = snd_pcm_ladspa_connect_controls(plugin, &plugin->output, instance); assert(err >= 0); + if (plugin->desc->activate) + plugin->desc->activate(instance->handle); } err = snd_pcm_ladspa_check_connect(plugin, &plugin->input, &instance->input, depth); if (err < 0) -- 1.9.1

2 1

[alsa-devel] [PATCHv2 00/11] Remove set_cache_io entirely from ASoC probe
by Xiubo Li 02 Apr '14

02 Apr '14

Change in v2: - Add .get_regmap() support for struct snd_soc_codec_driver. Xiubo Li (11): ASoC: core: Move the default regmap I/O setting to snd_soc_register_codec() ASoc: 88pm860x: Remove the set_cache_io() entirely from ASoC probe. ASoc: cq93vc: Remove the set_cache_io() entirely from ASoC probe. ASoc: mc13783: Remove the set_cache_io() entirely from ASoC probe. ASoc: si476x: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm5102: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm5110: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm8350: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm8400: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm8994: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm8997: Remove the set_cache_io() entirely from ASoC probe. include/sound/soc.h | 1 + sound/soc/codecs/88pm860x-codec.c | 12 ++++++++---- sound/soc/codecs/cq93vc.c | 10 ++++++++-- sound/soc/codecs/mc13783.c | 14 ++++++-------- sound/soc/codecs/si476x.c | 14 ++++++-------- sound/soc/codecs/wm5102.c | 12 ++++++++---- sound/soc/codecs/wm5110.c | 12 ++++++++---- sound/soc/codecs/wm8350.c | 10 ++++++++-- sound/soc/codecs/wm8400.c | 10 ++++++++-- sound/soc/codecs/wm8994.c | 10 ++++++++-- sound/soc/codecs/wm8997.c | 13 ++++++++----- sound/soc/soc-core.c | 28 ++++++++++++++++++---------- sound/soc/soc-io.c | 9 +++------ 13 files changed, 98 insertions(+), 57 deletions(-) -- 1.8.4

4 14

[alsa-devel] [PATCH 00/13] ASoC: Move IO and kcontrols to the component level
by Lars-Peter Clausen 02 Apr '14

02 Apr '14

Hi, This series is the first step towards full componentisation of the ASoC core. It moves both the IO abstraction layers within ASoC as well as the standard set of kcontrols to the component level. This for example means we can get rid of constructs like if (w->codec) snd_soc_read(....) else if(w->platform) snd_soc_platform_read(...) Moving the kcontrols to the component level means we can use the same implementation also for other non-CODEC components. E.g. there seems to be an increasing amount of CPU components that have basic signal processing and things like volume controls etc. whose register layout is similar to those used in CODECs. Currently each CPU component driver re-implements these controls by hand. The first two patches introduce two new helper functions which hide the actual implementation on how the CODEC or platform struct that register a control can be obtained from the control. This means that when the actual implementation is changed only the two helper functions need to be updated and not every single driver. The patches that follow that are just cleanups removing unused IO stuff and move all IO functions to soc-io.c. The next step is to make platforms also components. And then finally first the IO abstraction layers in ASoC are unified at the component level and then on top of that the kcontrol helpers are moved to the component level. The series depends on quite a few topic branches related to changes to the core and cleanups for individual drivers. It is probably best to place it on top of asoc-v3.15-2. The patch that moves the kcontrols to the component level also has a runtime dependency on the not yet applied patches that move the ams-delta and mfld_machine controls to the card level. - Lars Lars-Peter Clausen (13): ASoC: Add snd_soc_kcontrol_codec() helper function ASoC: Add snd_soc_kcontrol_platform() helper function ASoC: Prepare SOC_SINGLE_XR_SX controls for regmap ASoC: Move IO functions to soc-io.c ASoC: Drop ASoC level caching from hw_write/hw_read ASoC: Remove IO register modifier callbacks ASoC: Add helper function to cast component back to CODEC ASoC: Track which components have been registered with snd_soc_register_component() ASoC: Let snd_soc_platform subclass snd_soc_component ASoC: Move IO abstraction to the component level ASoC: Move standard kcontrol helpers to the component level ASoC: Remove snd_soc_update_bits_locked() ASoC: dapm: Rename soc_widget_update_bits_locked() to soc_widget_update_bits() include/sound/soc-dapm.h | 1 + include/sound/soc.h | 114 +++++++-- sound/soc/codecs/88pm860x-codec.c | 8 +- sound/soc/codecs/ab8500-codec.c | 12 +- sound/soc/codecs/adav80x.c | 4 +- sound/soc/codecs/ak4641.c | 4 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 4 +- sound/soc/codecs/cs42l51.c | 4 +- sound/soc/codecs/da7210.c | 4 +- sound/soc/codecs/da7213.c | 4 +- sound/soc/codecs/da732x.c | 4 +- sound/soc/codecs/da9055.c | 2 +- sound/soc/codecs/lm4857.c | 4 +- sound/soc/codecs/max9768.c | 4 +- sound/soc/codecs/max98088.c | 12 +- sound/soc/codecs/max98090.c | 4 +- sound/soc/codecs/max98095.c | 16 +- sound/soc/codecs/pcm1681.c | 4 +- sound/soc/codecs/rt5631.c | 4 +- sound/soc/codecs/sgtl5000.c | 4 +- sound/soc/codecs/sta32x.c | 4 +- sound/soc/codecs/tas5086.c | 4 +- sound/soc/codecs/tlv320aic23.c | 4 +- sound/soc/codecs/tlv320dac33.c | 4 +- sound/soc/codecs/twl4030.c | 10 +- sound/soc/codecs/twl6040.c | 8 +- sound/soc/codecs/wl1273.c | 12 +- sound/soc/codecs/wm2000.c | 8 +- sound/soc/codecs/wm8350.c | 4 +- sound/soc/codecs/wm8400.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8731.c | 4 +- sound/soc/codecs/wm8753.c | 4 +- sound/soc/codecs/wm8804.c | 4 +- sound/soc/codecs/wm8903.c | 4 +- sound/soc/codecs/wm8904.c | 14 +- sound/soc/codecs/wm8955.c | 4 +- sound/soc/codecs/wm8958-dsp2.c | 32 +-- sound/soc/codecs/wm8960.c | 4 +- sound/soc/codecs/wm8962.c | 8 +- sound/soc/codecs/wm8983.c | 4 +- sound/soc/codecs/wm8985.c | 4 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8994.c | 10 +- sound/soc/codecs/wm8996.c | 4 +- sound/soc/codecs/wm9081.c | 4 +- sound/soc/codecs/wm_adsp.c | 4 +- sound/soc/codecs/wm_hubs.c | 2 +- sound/soc/intel/sst-haswell-pcm.c | 8 +- sound/soc/soc-cache.c | 2 - sound/soc/soc-core.c | 520 ++++++++++++++++---------------------- sound/soc/soc-dapm.c | 91 +------ sound/soc/soc-io.c | 220 +++++++++++++--- 55 files changed, 644 insertions(+), 592 deletions(-) -- 1.8.0

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[alsa-devel] [PATCH] ASoC: core: Fail probe if we fail to add dai widgets
by Nariman Poushin 01 Apr '14

01 Apr '14

Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b322cf2..5cf7107 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1132,8 +1132,15 @@ static int soc_probe_codec(struct snd_soc_card *card, driver->num_dapm_widgets); /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(dai, &codec->component.dai_list, list) - snd_soc_dapm_new_dai_widgets(&codec->dapm, dai); + list_for_each_entry(dai, &codec->component.dai_list, list) { + ret = snd_soc_dapm_new_dai_widgets(&codec->dapm, dai); + + if (ret != 0) { + dev_err(codec->dev, + "Failed to create DAI widgets %d\n", ret); + goto err_probe; + } + } codec->dapm.idle_bias_off = driver->idle_bias_off; -- 1.8.5.3

2 1

[alsa-devel] [PATCH] ASoC: alc5632: fix uninit var in alc5632_probe()
by Stephen Warren 01 Apr '14

01 Apr '14

From: Stephen Warren <swarren(a)nvidia.com> alc5632_probe() returns ret, yet it is not initialized or set anywhere. This ends up causing the function to appear to fail, and audio not to work on the Toshiba AC100, with my compiler at least. This function used to set ret in all cases, but recent cleanup removed that. Cc: Xiubo Li <Li.Xiubo(a)freescale.com> Fixes: 5d6be5aa6bec ("ASoC: codec: Simplify ASoC probe code.") Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/alc5632.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index ec071a6306ef..3ee8d4e41a99 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1061,7 +1061,6 @@ static int alc5632_resume(struct snd_soc_codec *codec) static int alc5632_probe(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); - int ret; /* power on device */ alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1075,7 +1074,7 @@ static int alc5632_probe(struct snd_soc_codec *codec) return -EINVAL; } - return ret; + return 0; } /* power down chip */ -- 1.8.1.5

3 2

Re: [alsa-devel] [PATCH] ASoC: Add support for multi register mux
by Songhee Baek 01 Apr '14

01 Apr '14

> -----Original Message----- > From: Songhee Baek > Sent: Friday, March 28, 2014 11:10 AM > To: 'Lars-Peter Clausen' > Cc: Arun Shamanna Lakshmi; 'lgirdwood(a)gmail.com'; 'broonie(a)kernel.org'; > 'swarren(a)wwwdotorg.org'; 'alsa-devel(a)alsa-project.org'; 'tiwai(a)suse.de'; > 'linux-kernel(a)vger.kernel.org' > Subject: RE: [alsa-devel] [PATCH] ASoC: Add support for multi register mux > > > > > On 03/26/2014 11:41 PM, Songhee Baek wrote: > > > >> On 03/26/2014 01:02 AM, Arun Shamanna Lakshmi wrote: > > > >> > > > >> The way you describe this it seems to me that a value array for > > > >> this kind of mux would look like. > > > >> > > > >> 0x00000000, 0x00000000, 0x00000001 0x00000000, 0x00000000, > > > >> 0x00000002 0x00000000, 0x00000000, 0x00000003 0x00000000, > > > >> 0x00000000, 0x00000004 0x00000000, 0x00000000, 0x00000008 ... > > > >> > > > >> That seems to be extremely tedious. If the MUX uses a one hot > > > >> encoding how about storing the index of the bit in the values > > > >> array and use (1 << value) when writing the value to the register? > > > > > > > > If we store the index of the bit, the value will be duplicated for > > > > each > > > registers inputs since register has 0 to 31bits to shift, then we > > > need to decode the index to interpret value for which registers to > > > set. If we need to interpret the decoded value of index, it is > > > better to have custom put/get function in our driver, isn't it? > > > > > > > > > > I'm not sure I understand. If you use (val / 32) to pick the > > > register and (val % > > > 32) to pick the bit in the register this should work just fine. > > > Maybe I'm missing something. Do you have a real world code example > > > of of the this type of enum is used? > > > > > > > I can use val/32 and val%32 for this multi register mux. > With this logic, put function would be easy however get function becomes tedious due to looping on each bit per register for 3 of them in our case. Rather, it would be easy to identify a unique MUX_OFFSET to distinguish between the 3 registers as shown in the code below. These get/put functions are updated from previous mail, now it works for multi register mux, please review these function whether I can add in current put/get function. #define MULTI_MUX_INPUT_OFFSET(n) (5 * n) int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val, reg_idx; if (e->reg[0] != SND_SOC_NOPM) { for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { reg_val = snd_soc_read(codec, e->reg[reg_idx]); val = (reg_val >> e->shift_l) & e->mask[reg_idx]; if (val) { val += MULTI_MUX_INPUT_OFFSET(reg_idx); break; } } } else { reg_val = dapm_kcontrol_get_value(kcontrol); val = (reg_val >> e->shift_l) & e->mask[0]; } ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); if (e->shift_l != e->shift_r) { val = (reg_val >> e->shift_r) & e->mask[0]; val = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = val; } return 0; } int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int *item = ucontrol->value.enumerated.item; unsigned int change, i, value, val, update_idx = 0; unsigned int mask; struct snd_soc_dapm_update update; int ret = 0, reg_val; if (item[0] >= e->items) return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] > e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l; mask |= e->mask[0] << e->shift_r; } if (e->num_regs < 2) { value = val; goto update_reg; } for (i = 0; i < e->num_regs; i++) { reg_val = val - MULTI_MUX_INPUT_OFFSET(i); /* checking reg_val is power of 2 : one-hot code */ /* if reg_val after subtract MULTI_MUX_INPUT_OFFSET is not power of 2, reg[i] should be zero */ if (reg_val & (reg_val - 1)) { /* clear the current input register */ snd_soc_write(codec, e->reg[i], 0); } else { /* reg_val is power of 2, store updated info */ value = reg_val; mask = e->mask[i]; update_idx = i; } } update_reg: mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); if (e->reg[update_idx] != SND_SOC_NOPM) change = snd_soc_test_bits(codec, e->reg[update_idx], mask, value); else change = dapm_kcontrol_set_value(kcontrol, value); if (change) { if (e->reg[update_idx] != SND_SOC_NOPM) { update.kcontrol = kcontrol; update.reg = e->reg[update_idx]; update.mask= mask; update.val = value; card->update = &update; } ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); card->update = NULL; } mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); return change; } > > > >>> - int reg; > > > >>> + int reg[SOC_ENUM_MAX_REGS]; > > > >>> unsigned char shift_l; > > > >>> unsigned char shift_r; > > > >>> unsigned int items; > > > >>> - unsigned int mask; > > > >>> + unsigned int mask[SOC_ENUM_MAX_REGS]; > > > >> > > > >> If you make mask and reg pointers instead of arrays this should > > > >> be much more flexible and not be limited to 3 registers. > We will make reg* and mask* instead of arrays and since we use the same structure, the plan is to share the get and put function code. Thanks. Songhee.

4 6

[alsa-devel] [PATCH] ALSA: usb-audio - Capture and duplex support for Digidesign Mbox 1 sound card.
by Damien Zammit 01 Apr '14

01 Apr '14

This patch completes the support for Digidesign Mbox 1 sound card by adding capture. The issue with this card was that it has two endpoints on the same interface. This patch creates a dual endpoint quirk. The quirk interface needs a second audioformat struct for this to work which I called ".data2". Signed-off-by: Damien Zammit <damien(a)zamaudio.com>

4 13

[alsa-devel] [PATCH] ASoC: fsl_sai: Fix buggy configurations in trigger()
by Nicolin Chen 01 Apr '14

01 Apr '14

The current trigger() has two crucial problems: 1) The DMA request enabling operations (FSL_SAI_CSR_FRDE) for Tx and Rx are now totally exclusive: It would fail to run simultaneous Tx-Rx cases. 2) The TERE disabling operation depends on an incorrect condition -- active reference count that only gets increased in snd_pcm_open() and decreased in snd_pcm_close(): The TERE would never get cleared. So this patch overwrites the trigger function by following these rules: A) We continue to support tx-async-while-rx-sync-to-tx case alone, which's originally limited by this fsl_sai driver, but we make the code easy to modify for the further support of the opposite case. B) We enable both TE and RE for PLAYBACK stream or CAPTURE stream but only enabling the DMA request bit (FSL_SAI_CSR_FRDE) of the current direction due to the requirement of SAI -- For tx-async-while-rx-sync-to-tx case, the receiver is enabled only when both the transmitter and receiver are enabled. C) We only enable one side interrupt for each stream since over/underrun on the opposite stream would be resulted from what we've done in rule B: enabling TERE but remaining FRDE disabled, even though the xrun on the opposite direction will not break the current stream. Tested cases: a) aplay test.wav -d5 b) arecord -r44100 -c2 -fS16_LE test.wav -d5 c) arecord -r44100 -c2 -fS16_LE -d5 | aplay d) (aplay test2.wav &); sleep 1; arecord -r44100 -c2 -fS16_LE test.wav -d1 e) (arecord -r44100 -c2 -fS16_LE test.wav -d5 &); sleep 1; aplay test.wav -d1 Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 43 +++++++++++++++++++++++-------------------- sound/soc/fsl/fsl_sai.h | 11 +++++++++++ 2 files changed, 34 insertions(+), 20 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index f088545..d64c33f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -365,6 +365,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 tcsr, rcsr; /* @@ -379,14 +380,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_read(sai->regmap, FSL_SAI_TCSR, &tcsr); regmap_read(sai->regmap, FSL_SAI_RCSR, &rcsr); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - tcsr |= FSL_SAI_CSR_FRDE; - rcsr &= ~FSL_SAI_CSR_FRDE; - } else { - rcsr |= FSL_SAI_CSR_FRDE; - tcsr &= ~FSL_SAI_CSR_FRDE; - } - /* * It is recommended that the transmitter is the last enabled * and the first disabled. @@ -395,22 +388,32 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - tcsr |= FSL_SAI_CSR_TERE; - rcsr |= FSL_SAI_CSR_TERE; + if (!(tcsr & FSL_SAI_CSR_FRDE || rcsr & FSL_SAI_CSR_FRDE)) { + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, + FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, + FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); + } - regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr); - regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr); + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), + FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS); + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), + FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (!(cpu_dai->playback_active || cpu_dai->capture_active)) { - tcsr &= ~FSL_SAI_CSR_TERE; - rcsr &= ~FSL_SAI_CSR_TERE; + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), + FSL_SAI_CSR_FRDE, 0); + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), + FSL_SAI_CSR_xIE_MASK, 0); + + if (!(tcsr & FSL_SAI_CSR_FRDE || rcsr & FSL_SAI_CSR_FRDE)) { + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, + FSL_SAI_CSR_TERE, 0); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, + FSL_SAI_CSR_TERE, 0); } - - regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr); - regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr); break; default: return -EINVAL; @@ -464,8 +467,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, FSL_SAI_FLAGS); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, FSL_SAI_FLAGS); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_TX * 2); regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index a264185..be26d46 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -35,6 +35,16 @@ #define FSL_SAI_RFR 0xc0 /* SAI Receive FIFO */ #define FSL_SAI_RMR 0xe0 /* SAI Receive Mask */ +#define FSL_SAI_xCSR(tx) (tx ? FSL_SAI_TCSR : FSL_SAI_RCSR) +#define FSL_SAI_xCR1(tx) (tx ? FSL_SAI_TCR1 : FSL_SAI_RCR1) +#define FSL_SAI_xCR2(tx) (tx ? FSL_SAI_TCR2 : FSL_SAI_RCR2) +#define FSL_SAI_xCR3(tx) (tx ? FSL_SAI_TCR3 : FSL_SAI_RCR3) +#define FSL_SAI_xCR4(tx) (tx ? FSL_SAI_TCR4 : FSL_SAI_RCR4) +#define FSL_SAI_xCR5(tx) (tx ? FSL_SAI_TCR5 : FSL_SAI_RCR5) +#define FSL_SAI_xDR(tx) (tx ? FSL_SAI_TDR : FSL_SAI_RDR) +#define FSL_SAI_xFR(tx) (tx ? FSL_SAI_TFR : FSL_SAI_RFR) +#define FSL_SAI_xMR(tx) (tx ? FSL_SAI_TMR : FSL_SAI_RMR) + /* SAI Transmit/Recieve Control Register */ #define FSL_SAI_CSR_TERE BIT(31) #define FSL_SAI_CSR_FR BIT(25) @@ -48,6 +58,7 @@ #define FSL_SAI_CSR_FWF BIT(17) #define FSL_SAI_CSR_FRF BIT(16) #define FSL_SAI_CSR_xIE_SHIFT 8 +#define FSL_SAI_CSR_xIE_MASK (0x1f << FSL_SAI_CSR_xIE_SHIFT) #define FSL_SAI_CSR_WSIE BIT(12) #define FSL_SAI_CSR_SEIE BIT(11) #define FSL_SAI_CSR_FEIE BIT(10) -- 1.8.4

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[alsa-devel] cloning snd_seq_t (or creating one from client id)
by Tobias Schlemmer 31 Mar '14

31 Mar '14

Hi, Recently, I came upon the following problem with the ALSA sequencer interface: I'm surprised that the upcoming version of RtMidi uses one ALSA client per MIDI port. As an end user I'd expect to see only one ALSA client per program instance (e.g. using aconnect -i - o -l). Asking for the reason of that change I got the answer that it has something to do with broken multithread support. Digging deeper into this problem I found out that sequencer handles (snd_seq_t) are not thread safe. So my question is: Could you provide some way to implement multithreaded ALSA clients? My idea is to provide some function int snd_seq_clone(snd_seq_t * old, snd_seq_t ** new) /* clone a sequencer */ and/or int snd_seq_create_from_client(int clientid, snd_seq_t ** new) /* create a sequencer from a given client id */ and/or int snd_seq_split_port(snd_seq_t * old, snd_seq_t ** new, int port) /* create a sequencer that steels a port from old */ For the semantics I think cloned sequencers should act as if they don't know each other. Each one has its own set of ports. While the generation of port numbers still uses the same counter. I don't expect creating a port to be that time critical that it can't use locks. If it helps to ease the implementation one could use a master sequencer with some slaves. If the master sequencer is deleted the slaves are automatically freed and must not be used anymore. And if a subsequencer tries to access a port of another subsequncer, ALSA could just return with an error. Regards, Tobias P.S.: please, CC to me in answers.

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[alsa-devel] [PATCH] ASoC: rt5640: Change the setting method of idle_bias_off
by Oder Chiou 31 Mar '14

31 Mar '14

The patch moves the idle_bias_off setting to struct "soc_codec_dev_rt5640". Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index b088f14..6b8778f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1895,7 +1895,6 @@ static int rt5640_probe(struct snd_soc_codec *codec) return ret; } - codec->dapm.idle_bias_off = 1; rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); @@ -2004,6 +2003,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5640 = { .suspend = rt5640_suspend, .resume = rt5640_resume, .set_bias_level = rt5640_set_bias_level, + .idle_bias_off = true, .controls = rt5640_snd_controls, .num_controls = ARRAY_SIZE(rt5640_snd_controls), .dapm_widgets = rt5640_dapm_widgets, -- 1.8.1.1.439.g50a6b54

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