September 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [TINYCOMPRESS][PATCH 1/1] compress: Must check for POLLERR before POLLOUT/POLLIN
by Richard Fitzgerald 25 Sep '13

25 Sep '13

In the case of error the ALSA compressed driver sets revents as (POLLOUT | POLLWRNORM | POLLERR) or (POLLIN | POLLWRNORM | POLLERR). So we can't assume that POLLOUT or POLLIN indicate success, we must check for POLLERR first. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- compress.c | 18 +++++++++--------- 1 files changed, 9 insertions(+), 9 deletions(-) diff --git a/compress.c b/compress.c index 0e71c28..5cd0966 100644 --- a/compress.c +++ b/compress.c @@ -378,6 +378,9 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size return total; ret = poll(&fds, 1, compress->max_poll_wait_ms); + if (fds.revents & POLLERR) { + return oops(compress, EIO, "poll returned error!"); + } /* A pause will cause -EBADFD or zero. * This is not an error, just stop writing */ if ((ret == 0) || (ret == -EBADFD)) @@ -387,9 +390,6 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size if (fds.revents & POLLOUT) { continue; } - if (fds.revents & POLLERR) { - return oops(compress, EIO, "poll returned error!"); - } } /* write avail bytes */ if (size > avail.avail) @@ -438,6 +438,9 @@ int compress_read(struct compress *compress, void *buf, unsigned int size) return total; ret = poll(&fds, 1, compress->max_poll_wait_ms); + if (fds.revents & POLLERR) { + return oops(compress, EIO, "poll returned error!"); + } /* A pause will cause -EBADFD or zero. * This is not an error, just stop reading */ if ((ret == 0) || (ret == -EBADFD)) @@ -447,9 +450,6 @@ int compress_read(struct compress *compress, void *buf, unsigned int size) if (fds.revents & POLLIN) { continue; } - if (fds.revents & POLLERR) { - return oops(compress, EIO, "poll returned error!"); - } } /* read avail bytes */ if (size > avail.avail) @@ -616,15 +616,15 @@ int compress_wait(struct compress *compress, int timeout_ms) fds.events = POLLOUT | POLLIN; ret = poll(&fds, 1, timeout_ms); + if (fds.revents & POLLERR) { + return oops(compress, EIO, "poll returned error!"); + } /* A pause will cause -EBADFD or zero. */ if ((ret < 0) && (ret != -EBADFD)) return oops(compress, errno, "poll error"); if (fds.revents & (POLLOUT | POLLIN)) { return 0; } - if (fds.revents & POLLERR) { - return oops(compress, EIO, "poll returned error!"); - } return ret; } -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: kirkwood: simplify clock handling
by Uwe Kleine-König 24 Sep '13

24 Sep '13

There is no need to not use extclk if it is identical to the main clk. The main motivation for this patch is dropping devm_clk_put which is used in a wrong way by all other users. While at it also extend the comments. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- Hello, there might be some further optimisations possible. I only note them here because I don't have the hardware to test: - only enable extclk if it a clock rate used that makes use of the external clock. Not sure if that works; hardware docs reading necessary. - only provide extclk if it's != the internal clock. - The code uses: priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL); i.e. provides a con_id in the dt-case. I think that using NULL unconditionally should also work, i.e. return the first clk associated to the device. OTOH the current code might make things clearer because it's more explicit. Best regards Uwe --- sound/soc/kirkwood/kirkwood-i2s.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 0f3d73d..8224f93 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -104,16 +104,25 @@ static void kirkwood_set_rate(struct snd_soc_dai *dai, uint32_t clks_ctrl; if (rate == 44100 || rate == 48000 || rate == 96000) { - /* use internal dco for the supported rates - * defined in kirkwood_i2s_dai */ + /* + * use internal dco for the supported rates + * defined in kirkwood_i2s_dai + * Note: For these specific rates the dco is also used if + * kirkwood_i2s_dai_extclk is in use. + */ dev_dbg(dai->dev, "%s: dco set rate = %lu\n", __func__, rate); kirkwood_set_dco(priv->io, rate); clks_ctrl = KIRKWOOD_MCLK_SOURCE_DCO; } else { - /* use the external clock for the other rates - * defined in kirkwood_i2s_dai_extclk */ + /* + * use the external clock for the other rates + * defined in kirkwood_i2s_dai_extclk + * This case isn't reached if kirkwood_i2s_dai is in use (and so + * extclk isn't valid) because it only supports the three rates + * above. + */ dev_dbg(dai->dev, "%s: extclk set rate = %lu -> %lu\n", __func__, rate, 256 * rate); clk_set_rate(priv->extclk, 256 * rate); @@ -497,12 +506,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->extclk = devm_clk_get(&pdev->dev, "extclk"); if (!IS_ERR(priv->extclk)) { - if (priv->extclk == priv->clk) { - devm_clk_put(&pdev->dev, priv->extclk); - priv->extclk = ERR_PTR(-EINVAL); - } else { + clk_prepare_enable(priv->extclk); + if (priv->extclk != priv->clk) { dev_info(&pdev->dev, "found external clock\n"); - clk_prepare_enable(priv->extclk); soc_dai = &kirkwood_i2s_dai_extclk; } } -- 1.8.4.rc3

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Re: [alsa-devel] [PATCH v2 4/4] ASoC: rsnd: use regmap instead of original register mapping method
by Mark Brown 24 Sep '13

24 Sep '13

On Tue, Sep 17, 2013 at 10:23:58PM +0900, Kuninori Morimoto wrote: > For example, SSI device have many ports (0-8), > but ADG is only 1 device. > On this driver, all devices are using rsnd_write/read() to > access register. OK, this sounds like the module should have this information - ie check something in the module structure rather than asking regmap.

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[alsa-devel] [PATCH 1/3] ASoC: tlv320aic23: Remove #defines for I2C
by Mark Brown 24 Sep '13

24 Sep '13

From: Mark Brown <broonie(a)linaro.org> The only control interface supported by this driver is I2C so there is no need for conditional compilation around the control interface. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic23.c | 25 +------------------------ 1 file changed, 1 insertion(+), 24 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 31762eb..3299459 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -613,7 +613,6 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon), }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * If the i2c layer weren't so broken, we could pass this kind of data * around @@ -660,29 +659,7 @@ static struct i2c_driver tlv320aic23_i2c_driver = { .id_table = tlv320aic23_id, }; -#endif - -static int __init tlv320aic23_modinit(void) -{ - int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - ret = i2c_add_driver(&tlv320aic23_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register TLV320AIC23 I2C driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(tlv320aic23_modinit); - -static void __exit tlv320aic23_exit(void) -{ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_del_driver(&tlv320aic23_i2c_driver); -#endif -} -module_exit(tlv320aic23_exit); +module_i2c_driver(tlv320aic23_i2c_driver); MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); -- 1.8.4.rc3

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[alsa-devel] [PATCH v3 1/2] ASoC: kirkwood: clk: probe defer when clock not yet ready
by Jean-Francois Moine 24 Sep '13

24 Sep '13

At probe time, a clock device may not be ready when some other device wants to use it. This patch lets the functions clk_get/devm_clk_get return a probe defer when the clock is defined in the DT but not yet available. It also fixes an erroneous call to clk_get_sys() when __clk_get() fails. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v2: fix __clk_get() failure (from Russell King) Note: the associated patch v2 2/2 is unchanged and not resent --- drivers/clk/clk.c | 2 +- drivers/clk/clkdev.c | 7 ++++++- 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/drivers/clk/clk.c b/drivers/clk/clk.c index a004769..582abc8 100644 --- a/drivers/clk/clk.c +++ b/drivers/clk/clk.c @@ -2181,7 +2181,7 @@ EXPORT_SYMBOL_GPL(of_clk_del_provider); struct clk *of_clk_get_from_provider(struct of_phandle_args *clkspec) { struct of_clk_provider *provider; - struct clk *clk = ERR_PTR(-ENOENT); + struct clk *clk = ERR_PTR(-EPROBE_DEFER); /* Check if we have such a provider in our array */ mutex_lock(&of_clk_lock); diff --git a/drivers/clk/clkdev.c b/drivers/clk/clkdev.c index 442a313..1f402b9 100644 --- a/drivers/clk/clkdev.c +++ b/drivers/clk/clkdev.c @@ -157,7 +157,12 @@ struct clk *clk_get(struct device *dev, const char *con_id) if (dev) { clk = of_clk_get_by_name(dev->of_node, con_id); - if (!IS_ERR(clk) && __clk_get(clk)) + if (!IS_ERR(clk)) { + if (!__clk_get(clk)) + clk = ERR_PTR(-ENOENT); + return clk; + } + if (PTR_ERR(clk) == -EPROBE_DEFER) return clk; } -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] "ASoC: ab8500: Convert register I/O to regmap" breaks snowball
by Olof Johansson 24 Sep '13

24 Sep '13

Mark, Lee, [I wasn't subscribed to alsa-devel when the patch was posted, so I'm sending this as a standalone email instead of as a reply to that] linux-next 20130924 fails to boot on snowball (u8500_defconfig, non-DT boot): Unable to handle kernel NULL pointer dereference at virtual address 00000000 pgd = c0004000 [00000000] *pgd=00000000 Internal error: Oops: 80000005 [#1] PREEMPT SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.12.0-rc1-00003-gff795d6 #135 task: ef060000 ti: ef064000 task.ti: ef064000 PC is at 0x0 LR is at snd_soc_update_bits+0x4c/0x88 pc : [<00000000>] lr : [<c02d4f98>] psr: 60000013 sp : ef065d48 ip : 00000020 fp : ef0d054c r10: 00000000 r9 : ef3c0810 r8 : 00000000 r7 : 00000014 r6 : 00000000 r5 : 00000007 r4 : ef36ea00 r3 : 00000000 r2 : 00000007 r1 : 00000014 r0 : ef36ea00 Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c5787d Table: 0000404a DAC: 00000015 Process swapper/0 (pid: 1, stack limit = 0xef064240) Stack: (0xef065d48 to 0xef066000) 5d40: ef153410 c0550bb8 ef153410 ef36ea00 c053dc64 ef36ea00 5d60: c053dc64 c03d5e0c ef36eaac c02e4144 00000000 02000000 ef36ea00 00000000 5d80: c055a688 ef17bc10 ef0a7b00 c02e4358 00000000 00000000 00000000 ef36ea00 5da0: c055a5dc c055a688 c055daa4 c055aa0c ef36eaac c02d74fc c03d592c c055a678 5dc0: ee8115c8 c055daa4 ef0d00c0 00000000 00000000 c055a668 c03d5bf4 c02d8258 5de0: c055dab4 00000000 00000004 eec2d990 c055db7c ef0d0780 c055dbac c055db9c 5e00: c055a6a8 ef36ea30 ef36ea00 ef0d087c ef0d084c ef36ea30 c04a98f4 c055dac4 5e20: c021ce54 c055daa4 000005b8 c053f7f0 c055da64 c051dde8 00000069 ef064028 5e40: 00000000 c02d8ef0 00000000 c055da50 00000000 c02e684c c05924fc c053f7f0 5e60: 00000000 c055da64 c051dde8 c021bc88 c021bc70 c021aa94 00000000 c053f7f0 5e80: c055da64 c053f824 00000000 c021ac80 00000000 c055da64 c021abf4 c0218f6c 5ea0: ef03265c ef1b6734 c055da64 ef192500 c054fea8 c021a220 c0475770 c055da64 5ec0: 00000006 c055da64 00000006 c0563c80 c0563c80 c021b25c 00000000 c052c5f4 5ee0: 00000006 c0008798 ef010900 c04787e4 ef09c100 c03ac5e0 00000000 00000001 5f00: 000014dc c0540b9c 60000013 00000001 00000003 00000000 c0da19bd c03bf6b0 5f20: 00000069 c0036aa4 c04de34c 00000006 c0da19c8 00000006 00000000 c052c5f4 5f40: 00000006 c0563c80 c0563c80 c05014dc 00000069 c0523fec c0523fe0 c0501bc4 5f60: 00000006 00000006 c05014dc 00000000 88000000 00000000 00000000 00000000 5f80: c03a14ec 00000000 00000000 00000000 00000000 00000000 00000000 c03a14f4 5fa0: 00000000 00000000 c03a14ec c000e438 00000000 00000000 00000000 00000000 5fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 5fe0: 00000000 00000000 00000000 00000000 00000013 00000000 00000000 20000000 [<c02d4f98>] (snd_soc_update_bits+0x4c/0x88) from [<c02e4144>] (ab8500_audio_setup_mics+0xc8/0x118) [<c02e4144>] (ab8500_audio_setup_mics+0xc8/0x118) from [<c02e4358>] (ab8500_codec_probe+0x180/0x374) [<c02e4358>] (ab8500_codec_probe+0x180/0x374) from [<c02d74fc>] (soc_probe_codec+0x178/0x2f4) [<c02d74fc>] (soc_probe_codec+0x178/0x2f4) from [<c02d8258>] (snd_soc_instantiate_card+0x794/0x11c0) [<c02d8258>] (snd_soc_instantiate_card+0x794/0x11c0) from [<c02d8ef0>] (snd_soc_register_card+0x26c/0x324) [<c02d8ef0>] (snd_soc_register_card+0x26c/0x324) from [<c02e684c>] (mop500_probe+0xd0/0x118) [<c02e684c>] (mop500_probe+0xd0/0x118) from [<c021bc88>] (platform_drv_probe+0x18/0x1c) [<c021bc88>] (platform_drv_probe+0x18/0x1c) from [<c021aa94>] (driver_probe_device+0x10c/0x228) ... I bisected it down to: commit ff795d614bfa62a3c6fc0bcb75cb8842e5a87892 Author: Mark Brown <broonie(a)linaro.org> Date: Fri Sep 20 10:38:16 2013 +0100 ASoC: ab8500: Convert register I/O to regmap As part of a general push to eliminate the duplicated register I/O support in ASoC convert ab8500 to use regmap. Signed-off-by: Mark Brown <broonie(a)linaro.org> Acked-by: Lee Jones <lee.jones(a)linaro.org> ... but I didn't look closer at what was actually broken. -Olof

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[alsa-devel] [PATCH 1/3] ASoC: max98095: Remove custom hw_write() implementation
by Mark Brown 24 Sep '13

24 Sep '13

From: Mark Brown <broonie(a)linaro.org> The registers that are being kept uncached are marked as volatile anyway so the call has no practical impact. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max98095.c | 25 ++++--------------------- 1 file changed, 4 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 8dbcacd..29c266b 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -612,23 +612,6 @@ static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) } /* - * Filter coefficients are in a separate register segment - * and they share the address space of the normal registers. - * The coefficient registers do not need or share the cache. - */ -static int max98095_hw_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - int ret; - - codec->cache_bypass = 1; - ret = snd_soc_write(codec, reg, value); - codec->cache_bypass = 0; - - return ret ? -EIO : 0; -} - -/* * Load equalizer DSP coefficient configurations registers */ static void m98095_eq_band(struct snd_soc_codec *codec, unsigned int dai, @@ -648,8 +631,8 @@ static void m98095_eq_band(struct snd_soc_codec *codec, unsigned int dai, /* Step through the registers and coefs */ for (i = 0; i < M98095_COEFS_PER_BAND; i++) { - max98095_hw_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); - max98095_hw_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); + snd_soc_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); + snd_soc_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); } } @@ -673,8 +656,8 @@ static void m98095_biquad_band(struct snd_soc_codec *codec, unsigned int dai, /* Step through the registers and coefs */ for (i = 0; i < M98095_COEFS_PER_BAND; i++) { - max98095_hw_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); - max98095_hw_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); + snd_soc_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); + snd_soc_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); } } -- 1.8.4.rc3

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[alsa-devel] [PATCH 1/5] ASoC: Docs: Update codec documentation
by Liam Girdwood 24 Sep '13

24 Sep '13

Update the codec class driver documentation and bring it up to date with the current code base. This includes API changes, regmap and multi component. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Documentation/sound/alsa/soc/codec.txt | 46 +++++++++++----------------------- 1 file changed, 15 insertions(+), 31 deletions(-) diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt index bce23a4..db5f9c9 100644 --- a/Documentation/sound/alsa/soc/codec.txt +++ b/Documentation/sound/alsa/soc/codec.txt @@ -1,22 +1,23 @@ -ASoC Codec Driver -================= +ASoC Codec Class Driver +======================= -The codec driver is generic and hardware independent code that configures the -codec to provide audio capture and playback. It should contain no code that is -specific to the target platform or machine. All platform and machine specific -code should be added to the platform and machine drivers respectively. +The codec class driver is generic and hardware independent code that configures +the codec, FM, MODEM, BT or external DSP to provide audio capture and playback. +It should contain no code that is specific to the target platform or machine. +All platform and machine specific code should be added to the platform and +machine drivers respectively. -Each codec driver *must* provide the following features:- +Each codec class driver *must* provide the following features:- 1) Codec DAI and PCM configuration - 2) Codec control IO - using I2C, 3 Wire(SPI) or both APIs + 2) Codec control IO - using RegMap API 3) Mixers and audio controls 4) Codec audio operations + 5) DAPM description. + 6) DAPM event handler. Optionally, codec drivers can also provide:- - 5) DAPM description. - 6) DAPM event handler. 7) DAC Digital mute control. Its probably best to use this guide in conjunction with the existing codec @@ -64,26 +65,9 @@ struct snd_soc_dai_driver wm8731_dai = { 2 - Codec control IO -------------------- The codec can usually be controlled via an I2C or SPI style interface -(AC97 combines control with data in the DAI). The codec drivers provide -functions to read and write the codec registers along with supplying a -register cache:- - - /* IO control data and register cache */ - void *control_data; /* codec control (i2c/3wire) data */ - void *reg_cache; - -Codec read/write should do any data formatting and call the hardware -read write below to perform the IO. These functions are called by the -core and ALSA when performing DAPM or changing the mixer:- - - unsigned int (*read)(struct snd_soc_codec *, unsigned int); - int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); - -Codec hardware IO functions - usually points to either the I2C, SPI or AC97 -read/write:- - - hw_write_t hw_write; - hw_read_t hw_read; +(AC97 combines control with data in the DAI). The codec driver should use the +Regmap API for all codec IO. Please see include/linux/regmap.h and existing +codec drivers for example regmap usage. 3 - Mixers and audio controls @@ -127,7 +111,7 @@ Defines a stereo enumerated control 4 - Codec Audio Operations -------------------------- -The codec driver also supports the following ALSA operations:- +The codec driver also supports the following ALSA PCM operations:- /* SoC audio ops */ struct snd_soc_ops { -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: fsl: fsl_ssi: Fix simultaneous capture and playback
by Fabio Estevam 24 Sep '13

24 Sep '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> When doing simultaneous capture and playback on a mx6 board we get the following error: $ arecord -f cd | aplay -f cd imx-sgtl5000 sound.13: set sample size in capture stream first fsl-ssi-dai 2028000.ssi: ASoC: can't open interface 2028000.ssi: -11 ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_ open) unable to open slave aplay: main:660: audio open error: Device or resource busy Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo The 'arecord -f cd | aplay -f cd' always trigger cause the 'if (!first_runtime->sample_bits)' block to be true which returns an error. Adjust the logic inside fsl_ssi_startup(), so that we do not always hit the error when playing 'arecord | aplay' line for the first time. Reported-by: Chris Clepper <cgclepper(a)gmail.com> Suggested-by: Nicolin Chen <b42378(a)freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6ac8730..cdbb641 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -469,19 +469,12 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * parameters, then the second stream may be * constrained to the wrong sample rate or size. */ - if (!first_runtime->sample_bits) { - dev_err(substream->pcm->card->dev, - "set sample size in %s stream first\n", - substream->stream == - SNDRV_PCM_STREAM_PLAYBACK - ? "capture" : "playback"); - return -EAGAIN; - } - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + if (first_runtime->sample_bits) { + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, first_runtime->sample_bits, first_runtime->sample_bits); + } } ssi_private->second_stream = substream; -- 1.8.1.2

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[alsa-devel] Audio Miniconference 2013 - Edinburgh 21st October
by Mark Brown 24 Sep '13

24 Sep '13

As in previous years we're planning to have an audio mini-summit to bring everyone together to discuss the current state of audio in Linux, try to resolve outstanding design issues and do some hacking. This year we will be meeting in Edinburgh on 21st October as part of the Linux Foundation conferences being held that week. What topics would people like to discuss? There are a number of ongoing issues like DSP integration and related topics such as compressed audio and I know that there are some issues which have come up while people have been working on deploying UCM especially with PulseAudio integration. What other areas are of interest to people? To help with planning could you please fill out the form at: https://docs.google.com/forms/d/105uav9tkwwkDU4P1T3IOWjfsa3kgtDbFAJOlHgBQR5… if you're planning to attend, ideally by 21st September. Since the facilities are being provided by the Linux Foundation attendees will need to be registered for at least one of the other conferences that week, if that's an issue please let us know. There's a space for topics of interest on the form but please also bring things up on the list so other people can get a feel for what's going on and ideally so we can resolve them all via e-mail!

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