December 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC: rt5616: add devicetree document for rt5616" to the asoc tree
by Mark Brown 23 Dec '15

23 Dec '15

The patch ASoC: rt5616: add devicetree document for rt5616 has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 27af4e488c6a2a8c4b6102e777c29fff467a3895 Mon Sep 17 00:00:00 2001 From: Caesar Wang <wxt(a)rock-chips.com> Date: Tue, 22 Dec 2015 13:45:03 +0800 Subject: [PATCH] ASoC: rt5616: add devicetree document for rt5616 Add the description for rt5616 codec. Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/rt5616.txt | 26 ++++++++++++++++++++++ 1 file changed, 26 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5616.txt diff --git a/Documentation/devicetree/bindings/sound/rt5616.txt b/Documentation/devicetree/bindings/sound/rt5616.txt new file mode 100644 index 000000000000..efc48c65198d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5616.txt @@ -0,0 +1,26 @@ +RT5616 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5616". + +- reg : The I2C address of the device. + +Pins on the device (for linking into audio routes) for RT5616: + + * IN1P + * IN2P + * IN2N + * LOUTL + * LOUTR + * HPOL + * HPOR + +Example: + +codec: rt5616@1b { + compatible = "realtek,rt5616"; + reg = <0x1b>; +}; -- 2.6.2

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[alsa-devel] [PATCH 1/2] ASoC: rt5616: add an of_match table
by Caesar Wang 23 Dec '15

23 Dec '15

Add a device tree match table. This serves to make the driver's support of device tree more explicit. Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> --- sound/soc/codecs/rt5616.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index f4005cb..e01a32d 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1287,6 +1287,14 @@ static const struct i2c_device_id rt5616_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5616_i2c_id); +#if defined(CONFIG_OF) +static const struct of_device_id rt5616_of_match[] = { + { .compatible = "realtek,rt5616", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5616_of_match); +#endif + static int rt5616_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1359,6 +1367,7 @@ static void rt5616_i2c_shutdown(struct i2c_client *client) static struct i2c_driver rt5616_i2c_driver = { .driver = { .name = "rt5616", + .of_match_table = of_match_ptr(rt5616_of_match), }, .probe = rt5616_i2c_probe, .remove = rt5616_i2c_remove, -- 1.9.1

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[alsa-devel] Applied "ASoC: da7218: Enable mic level detection reporting to user-space" to the asoc tree
by Mark Brown 23 Dec '15

23 Dec '15

The patch ASoC: da7218: Enable mic level detection reporting to user-space has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e05c25a1af29d65260ed1458f2cc4a959030ebd2 Mon Sep 17 00:00:00 2001 From: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Date: Thu, 3 Dec 2015 17:10:07 +0000 Subject: [PATCH] ASoC: da7218: Enable mic level detection reporting to user-space This patch adds support to the codec driver to handle mic level detect related IRQs, and report these to user-space using a uevent variable. The uevent variable string "EVENT=MIC_LEVEL_DETECT" is sent to user-space, if the mic level detect feature is enabled, and the audio captured at the chosen mic(s) is above a certain threshold. User-space can then handle the event accordingly (e.g. process audio capture stream). This method was chosen over ALSA control notification for a couple of reasons: 1) There's no requirement here for a control to read state from. The event is the only thing that's required and of interest. 2) tinyalsa support for control notifications does not exist so on platforms using this over alsa-lib there is a need to add code to support this event handling. Another possible option would be to use the standard Jack reporting framework but this really does not fit for this kind of event. Finally, use of the input device framework is not being encouraged, due to difficulties in enabling apps to access input devices, so this has also been avoided. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/da7218.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index eacde128c4d6..72686517ff54 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -2202,6 +2202,16 @@ int da7218_hpldet(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(da7218_hpldet); +static void da7218_micldet_irq(struct snd_soc_codec *codec) +{ + char *envp[] = { + "EVENT=MIC_LEVEL_DETECT", + NULL, + }; + + kobject_uevent_env(&codec->dev->kobj, KOBJ_CHANGE, envp); +} + static void da7218_hpldet_irq(struct snd_soc_codec *codec) { struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); @@ -2232,6 +2242,10 @@ static irqreturn_t da7218_irq_thread(int irq, void *data) if (!status) return IRQ_NONE; + /* Mic level detect */ + if (status & DA7218_LVL_DET_EVENT_MASK) + da7218_micldet_irq(codec); + /* HP detect */ if (status & DA7218_HPLDET_JACK_EVENT_MASK) da7218_hpldet_irq(codec); @@ -2936,11 +2950,6 @@ static int da7218_probe(struct snd_soc_codec *codec) } if (da7218->irq) { - /* Mask off mic level events, currently not handled */ - snd_soc_update_bits(codec, DA7218_EVENT_MASK, - DA7218_LVL_DET_EVENT_MSK_MASK, - DA7218_LVL_DET_EVENT_MSK_MASK); - ret = devm_request_threaded_irq(codec->dev, da7218->irq, NULL, da7218_irq_thread, IRQF_TRIGGER_LOW | IRQF_ONESHOT, -- 2.6.2

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[alsa-devel] [PATCH 0/12] ASoC: rsnd: add Multi channel support
by Kuninori Morimoto 23 Dec '15

23 Dec '15

Hi Mark These are Multi channel support for Renesas R-Car sound. These patches are tested on Salvator-X board, but it can't use Multi channel, because supported format is different between codec chip and R-Car. Thus, it was tested on board which doesn't mount codec chip, with oscilloscope. Kuninori Morimoto (12): 1) ASoC: rsnd: fixup SSIU control timing 2) ASoC: rsnd: tidyup return value of rsnd_get_adinr_bit() 3) ASoC: rsnd: tidyup debug print position on rsnd_dma_attach() 4) ASoC: rsnd: rsnd_dai_connect() returns error if it connect to existing mod 5) ASoC: rsnd: SSI/SSIU use rsnd_get_slot_extend() to check TDM 6) ASoC: rsnd: remove rsnd_get_slot_runtime() 7) ASoC: rsnd: tidyup rsnd_get_slot_xxx() naming 8) ASoC: rsnd: add rsnd_set_slot() / rsnd_get_slot_num() 9) ASoC: rsnd: add rsnd_parse_connect_common() and remove complex macro 10) ASoC: rsnd: add missing DT example for Simple Card 11) ASoC: rsnd: add missing DT example for Simple Card with TDM 12) ASoC: rsnd: add Multi channel support Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 77 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/core.c | 111 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++---------------------------------------- sound/soc/sh/rcar/dma.c | 10 ++++----- sound/soc/sh/rcar/dvc.c | 3 +-- sound/soc/sh/rcar/gen.c | 3 +++ sound/soc/sh/rcar/rsnd.h | 45 +++++++++++++++++++++++++++++++++------ sound/soc/sh/rcar/ssi.c | 118 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++---- sound/soc/sh/rcar/ssiu.c | 57 +++++++++++++++++++++++++++++++++++++++++--------- 8 files changed, 351 insertions(+), 73 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: ssm2518: Use a signed return type for ssm2518_lookup_mcs()
by SF Markus Elfring 23 Dec '15

23 Dec '15

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 20 Dec 2015 10:34:25 +0100 The return type "unsigned int" was used by the ssm2518_lookup_mcs() function even though it will eventually return a negative error code. Improve this implementation detail by deletion of the type modifier then. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/codecs/ssm2518.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 86b81a6..e2e0bfa 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -309,7 +309,7 @@ static const struct snd_pcm_hw_constraint_list ssm2518_constraints_12288000 = { .count = ARRAY_SIZE(ssm2518_rates_12288000), }; -static unsigned int ssm2518_lookup_mcs(struct ssm2518 *ssm2518, +static int ssm2518_lookup_mcs(struct ssm2518 *ssm2518, unsigned int rate) { const unsigned int *sysclks = NULL; -- 2.6.3

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[alsa-devel] [PATCH] ASoC: rt5616: rename some alsa control names
by Bard Liao 23 Dec '15

23 Dec '15

Rename some alsa control name as what they should be. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5616.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index f4005cb..4de8c86 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -323,9 +323,9 @@ static const struct snd_kcontrol_new rt5616_snd_controls[] = { RT5616_L_VOL_SFT, RT5616_R_VOL_SFT, 175, 0, dac_vol_tlv), /* IN1/IN2 Control */ - SOC_SINGLE_TLV("IN1 Boost", RT5616_IN1_IN2, + SOC_SINGLE_TLV("IN1 Boost Volume", RT5616_IN1_IN2, RT5616_BST_SFT1, 8, 0, bst_tlv), - SOC_SINGLE_TLV("IN2 Boost", RT5616_IN1_IN2, + SOC_SINGLE_TLV("IN2 Boost Volume", RT5616_IN1_IN2, RT5616_BST_SFT2, 8, 0, bst_tlv), /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5616_INL1_INR1_VOL, @@ -339,7 +339,7 @@ static const struct snd_kcontrol_new rt5616_snd_controls[] = { 127, 0, adc_vol_tlv), /* ADC Boost Volume Control */ - SOC_DOUBLE_TLV("ADC Boost Gain", RT5616_ADC_BST_VOL, + SOC_DOUBLE_TLV("ADC Boost Volume", RT5616_ADC_BST_VOL, RT5616_ADC_L_BST_SFT, RT5616_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), }; -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: atmel_wm8904: add snd_soc_pm_ops
by Songjun Wu 23 Dec '15

23 Dec '15

Sometimes the audio play can not be resumed after it is suspended. Add snd_soc_pm_ops to execute power management operations, then this issue is fixed. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> --- sound/soc/atmel/atmel_wm8904.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index 1933bcd..fdd28ed 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -183,6 +183,7 @@ static struct platform_driver atmel_asoc_wm8904_driver = { .driver = { .name = "atmel-wm8904-audio", .of_match_table = of_match_ptr(atmel_asoc_wm8904_dt_ids), + .pm = &snd_soc_pm_ops, }, .probe = atmel_asoc_wm8904_probe, .remove = atmel_asoc_wm8904_remove, -- 1.7.9.5

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[alsa-devel] [PATCH v2] ASoC: fsl_asrc: add support for 8-30kHz output sample rate
by Zidan Wang 23 Dec '15

23 Dec '15

Add 8kHz, 11.025kHz, 16kHz, 22.05kHz output sample rate support. According referance menual, "Limited support for the case when output sampling rates is between 8kHz and 30kHz. The limitation is the supported ratio (Fsin/Fsout) range as between 1/24 to 8." Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v1->v2: 1. fix indentation issue. sound/soc/fsl/fsl_asrc.c | 39 +++++++++++++++++++++++---------------- 1 file changed, 23 insertions(+), 16 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index d964016..bd32c30 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -31,21 +31,21 @@ dev_dbg(&asrc_priv->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) /* Sample rates are aligned with that defined in pcm.h file */ -static const u8 process_option[][8][2] = { - /* 32kHz 44.1kHz 48kHz 64kHz 88.2kHz 96kHz 176kHz 192kHz */ - {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 5512Hz */ - {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 8kHz */ - {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 11025Hz */ - {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 16kHz */ - {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 22050Hz */ - {{0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0},}, /* 32kHz */ - {{0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 44.1kHz */ - {{0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 48kHz */ - {{1, 2}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0},}, /* 64kHz */ - {{1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 88.2kHz */ - {{1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 96kHz */ - {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 176kHz */ - {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 192kHz */ +static const u8 process_option[][12][2] = { + /* 8kHz 11.025kHz 16kHz 22.05kHz 32kHz 44.1kHz 48kHz 64kHz 88.2kHz 96kHz 176kHz 192kHz */ + {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 5512Hz */ + {{0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 8kHz */ + {{0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 11025Hz */ + {{1, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 16kHz */ + {{1, 2}, {1, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 22050Hz */ + {{1, 2}, {2, 1}, {2, 1}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0},}, /* 32kHz */ + {{2, 2}, {2, 2}, {2, 1}, {2, 1}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 44.1kHz */ + {{2, 2}, {2, 2}, {2, 1}, {2, 1}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 48kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {1, 2}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0},}, /* 64kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 88.2kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 96kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 176kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 192kHz */ }; /* Corresponding to process_option */ @@ -55,7 +55,7 @@ static int supported_input_rate[] = { }; static int supported_asrc_rate[] = { - 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, + 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, }; /** @@ -287,6 +287,13 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) return -EINVAL; } + if ((outrate > 8000 && outrate < 30000) && + (outrate/inrate > 24 || inrate/outrate > 8)) { + pair_err("exceed supported ratio range [1/24, 8] for \ + inrate/outrate: %d/%d\n", inrate, outrate); + return -EINVAL; + } + /* Validate input and output clock sources */ clk_index[IN] = clk_map[IN][config->inclk]; clk_index[OUT] = clk_map[OUT][config->outclk]; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: fsl-asoc-card: use different route map for AC'97 mode
by Maciej S. Szmigiero 23 Dec '15

23 Dec '15

fsl_ssi uses different stream names ("AC97 Playback" / "AC97 Capture") in AC'97 mode so in this case fsl-asoc-card route map should also be using them. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl-asoc-card.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index c63d89da51f1..562b3bd22d9a 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -107,6 +107,13 @@ static const struct snd_soc_dapm_route audio_map[] = { {"CPU-Capture", NULL, "Capture"}, }; +static const struct snd_soc_dapm_route audio_map_ac97[] = { + {"AC97 Playback", NULL, "ASRC-Playback"}, + {"Playback", NULL, "AC97 Playback"}, + {"ASRC-Capture", NULL, "AC97 Capture"}, + {"AC97 Capture", NULL, "Capture"}, +}; + /* Add all possible widgets into here without being redundant */ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_LINE("Line Out Jack", NULL), @@ -579,7 +586,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.dev = &pdev->dev; priv->card.name = priv->name; priv->card.dai_link = priv->dai_link; - priv->card.dapm_routes = audio_map; + priv->card.dapm_routes = fsl_asoc_card_is_ac97(priv) ? + audio_map_ac97 : audio_map; priv->card.late_probe = fsl_asoc_card_late_probe; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;

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[alsa-devel] [PATCH 0/5] ASoC: Allow topology to create DAI links
by mengdong.lin＠linux.intel.com 23 Dec '15

23 Dec '15

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series allows DAI links to be created from topology info. The components with topology info can come in two ways: - As a component for cpu dai or codec dai of a DAI link predefined by machine driver. - As a auxiliary components defined by the machine driver for a soc card. For this case, the machine driver even need not define any DAI links. The ASoC core will find and probe these components when instantiating the soc card. When probing the components, topology info will be loaded and DAI links can be created by the topology core. And then ASoC core will find and bind new DAI links after probing the components. We implement a DAI link list and define API for adding/removing DAI links from topology. Mengdong Lin (5): ASoC: Implement DAI links in a list & define API to add/remove a link ASoC: Define add/remove_dai_link ops for a soc card ASoC: soc_bind_dai_link() directly returns success for a bound DAI link ASoC: Bind new DAI links after probing components ASoC: The soc card can have auxiliary components include/sound/soc.h | 29 +++++++++- sound/soc/soc-core.c | 153 +++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 180 insertions(+), 2 deletions(-) -- 2.5.0

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