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

[alsa-devel] [PATCH v2] ASoC: fsl_ssi: Add monaural audio support for non-ac97 interface
by Nicolin Chen 02 Dec '13

02 Dec '13

The normal mode of SSI allows it to send/receive data to/from the first slot of each period. So we can use this normal mode to trick I2S signal by puting/getting data to/from the first slot only (the left channel) so as to support monaural audio playback and recording. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v2: * Moved i2s_mode to ssi_private so that we can save and retore it as needed * And dropped the horrible static thing. sound/soc/fsl/fsl_ssi.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f43be6d..cba5c4d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -144,6 +144,7 @@ struct fsl_ssi_private { bool imx_ac97; bool use_dma; bool use_dual_fifo; + u8 i2s_mode; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -325,14 +326,13 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u8 i2s_mode; u8 wm; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; if (ssi_private->imx_ac97) - i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; else - i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; /* * Section 16.5 of the MPC8610 reference manual says that the SSI needs @@ -349,7 +349,7 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, CCSR_SSI_SCR_TFR_CLK_DIS | - i2s_mode | + ssi_private->i2s_mode | (synchronous ? CCSR_SSI_SCR_SYN : 0)); write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | @@ -517,6 +517,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int channels = params_channels(hw_params); unsigned int sample_size = snd_pcm_format_width(params_format(hw_params)); u32 wl = CCSR_SSI_SxCCR_WL(sample_size); @@ -546,6 +547,11 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, else write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + if (!ssi_private->imx_ac97) + write_ssi_mask(&ssi->scr, + CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, + channels == 1 ? 0 : ssi_private->i2s_mode); + return 0; } @@ -658,14 +664,13 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { static struct snd_soc_dai_driver fsl_ssi_dai_template = { .probe = fsl_ssi_dai_probe, .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, -- 1.8.4

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[alsa-devel] HD-Audio: How to reduce driver initializaton time if multiple codecs present on the bus?
by Lin, Mengdong 02 Dec '13

02 Dec '13

Hi Takashi, We're trying to reduce the HD-A driver initialization time when more than one codecs are connected to the bus, but are blocked. Would you please share some advices on this? Usually, there is one HD-A controller connecting to two codecs: one on-board codec and one integrated display codec. During initialization, the codecs are created and configured in a serial way. Creating a codec may cost 6~20ms, and then building controls make cost about 15~30ms. So I had thought it's helpful to build controls for different codecs in parallel in function snd_hda_build_controls(), as we did in driver suspend/resume. But the test shows this doesn't work: (1) Using a workqueue without WQ_UNBOUND, the work items, which calls snd_hda_codec_build_controls(), are processed one after one (I think because there is no explicit sleep in the work function). So the total time does not change. (2) Using an unbouned work queue for the bus or using separate kthreads for each codec, the work items for different codecs are processed in parallel. However, the total time is still the sum of two codec as that without parallel. I think that because snd_hda_codec_read() are used during initialization, the command execution on bus is serialized in the low level. So even if we parallel initializing two codecs in high level, commands on bus are still processed one by one so the total time cannot reduce. We're blocked here. Is there any other means to reduce the time? Thanks Mengdong

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[alsa-devel] [PATCH] ASoC: fsl: imx-ssi: omit ssi counter to avoid harm in unbalanced situation
by Oskar Schirmer 02 Dec '13

02 Dec '13

Unbalanced calls to imx_ssi_trigger() may result in endless SSI activity and thus provoke eternal sound. While on the first glance, the switch statement looks pretty symmetric, the SUSPEND/RESUME pair is not: the suspend case comes along snd_pcm_suspend_all(), which for fsl/imx-pcm-fiq is called only at snd_soc_suspend(), but the resume case originates straight from the SNDRV_PCM_IOCTL_RESUME. This way userland may provoke an unbalanced resume, which might cause the ssi->enabled counter to increase and never return to zero again, so eventually SSI_SCR_SSIEN is never disabled. Simply removing the ssi->enabled will solve the problem, as long as one never goes play and capture game simultaneously, but beware trying both at once, the early TRIGGER_STOP will cut off the other activity prematurely. So now playing and capturing is scrutinized separately, instead of by counting. This is essentially the same stuff as in sound/soc/fsl/imx-pcm-fiq.c which I send a patch for three days ago. Astonishing enough this highly fragile scheme is used twice in parallel to serve the very same control function, synchronously: Once out of sync you are lost until reboot. Note, that these fixes wont prevent state machine distortion on alsa level to cut sound or the like. It just makes sure we have a chance to synchronise again later on. Signed-off-by: Oskar Schirmer <oskar(a)scara.com> --- sound/soc/fsl/imx-ssi.c | 19 ++++++++++++------- sound/soc/fsl/imx-ssi.h | 3 ++- 2 files changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index f5f248c..c68499b 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -298,27 +298,32 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { scr |= SSI_SCR_TE; - else + ssi->playing = 1; + } else { scr |= SSI_SCR_RE; + ssi->capturing = 1; + } sier |= sier_bits; - if (++ssi->enabled == 1) - scr |= SSI_SCR_SSIEN; + scr |= SSI_SCR_SSIEN; break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { scr &= ~SSI_SCR_TE; - else + ssi->playing = 0; + } else { scr &= ~SSI_SCR_RE; + ssi->capturing = 0; + } sier &= ~sier_bits; - if (--ssi->enabled == 0) + if (!ssi->playing && !ssi->capturing) scr &= ~SSI_SCR_SSIEN; break; diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index 560c40f..a3c90d5 100644 --- a/sound/soc/fsl/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h @@ -213,7 +213,8 @@ struct imx_ssi { int fiq_init; int dma_init; - int enabled; + int playing; + int capturing; }; #endif /* _IMX_SSI_H */ -- 1.7.9.5

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[alsa-devel] [PATCH 1/1] ASoC: soc_compress: Add set_metadata
by Richard Fitzgerald 02 Dec '13

02 Dec '13

Pass the set_metadata() calls through to the codec driver. Signed-off-by: Zhao Weijia <weijia.zhao(a)capelabs.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 18 +++++++++++++++++- 1 files changed, 17 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 53c9ecd..186802b 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -318,6 +318,21 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, mutex_unlock(&rtd->pcm_mutex); return 0; } +static int soc_compr_set_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + + if (platform->driver->compr_ops && platform->driver->compr_ops->set_metadata) + platform->driver->compr_ops->set_metadata(cstream, metadata); + + mutex_unlock(&rtd->pcm_mutex); + return 0; +} + static int soc_compr_copy(struct snd_compr_stream *cstream, char __user *buf, size_t count) @@ -372,7 +387,8 @@ static struct snd_compr_ops soc_compr_ops = { .pointer = soc_compr_pointer, .ack = soc_compr_ack, .get_caps = soc_compr_get_caps, - .get_codec_caps = soc_compr_get_codec_caps + .get_codec_caps = soc_compr_get_codec_caps, + .set_metadata = soc_compr_set_metadata, }; /* create a new compress */ -- 1.7.2.5

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[alsa-devel] Using devm_snd_soc_register_card() causes problem with imx-sgtl5000
by Shawn Guo 02 Dec '13

02 Dec '13

I'm trying to test Lars' devm_snd_dmaengine_pcm_register patches with imx-sgtl5000 machine driver, but running into the rmmod problem below on v3.13-rc1. Reverting commit 01984a4 (ASoC: imx-sgtl5000: Use devm_snd_soc_register_card) seems to fix the problem for me. Any hint on what's going wrong here? Shawn root@arm:~# lsmod Module Size Used by snd_soc_imx_sgtl5000 3506 0 snd_soc_sgtl5000 13677 2 snd_soc_imx_audmux 5324 1 snd_soc_imx_sgtl5000 snd_soc_fsl_ssi 8139 2 imx_pcm_dma 1380 1 snd_soc_fsl_ssi root@arm:~# rmmod snd_soc_imx_sgtl5000 Unable to handle kernel paging request at virtual address e594025c pgd = be134000 [e594025c] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: snd_soc_imx_sgtl5000(-) snd_soc_sgtl5000 snd_soc_imx_audmux snd_soc_fsl_ssi imx_pcm_dma CPU: 0 PID: 1793 Comm: rmmod Not tainted 3.13.0-rc1 #1570 task: bee28900 ti: bfbec000 task.ti: bfbec000 PC is at debugfs_remove_recursive+0x28/0x154 LR is at snd_soc_unregister_card+0xa0/0xcc pc : [<80252b38>] lr : [<80496ac4>] psr: a0000013 sp : bfbede00 ip : bfbede28 fp : bfbede24 r10: 803281d4 r9 : bfbec000 r8 : 803271ac r7 : bef54440 r6 : 00000004 r5 : bf9a4010 r4 : bf9a4010 r3 : e5940224 r2 : 00000000 r1 : bef54450 r0 : 803271ac Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c53c7d Table: 4e13404a DAC: 00000015 Process rmmod (pid: 1793, stack limit = 0xbfbec240) Stack: (0xbfbede00 to 0xbfbee000) de00: 00000000 bf9a4010 bf9a4010 00000004 bef54440 bec89000 bfbede44 bfbede28 de20: 80496ac4 80252b1c 804a4b60 bfbede60 bf9a4010 00000004 bfbede54 bfbede48 de40: 804a4b74 80496a30 bfbede94 bfbede58 80328728 804a4b6c bfbede94 a0000013 de60: bf1b5800 bef54440 00000002 bf9a4010 7f0169f8 bf9a4044 00000081 8000e9c4 de80: bfbec000 00000000 bfbedeac bfbede98 80328cb0 80328618 7f016000 bf9a4010 dea0: bfbedec4 bfbedeb0 8032561c 80328c84 bf9a4010 7f0169f8 bfbedee4 bfbedec8 dec0: 80325e84 803255a8 bee28900 7f0169f8 00000000 78208d30 bfbedefc bfbedee8 dee0: 80325410 80325dd4 beca8100 7f0169f8 bfbedf14 bfbedf00 803264f8 803253c8 df00: 7f01635c 7f016a3c bfbedf24 bfbedf18 80327098 803264d4 bfbedf34 bfbedf28 df20: 7f016370 80327090 bfbedfa4 bfbedf38 80085ef0 7f016368 bfbedf54 5f646e73 df40: 5f636f73 5f786d69 6c746773 30303035 00000000 78208008 bfbedf84 bfbedf68 df60: 800613b0 80061194 fffffffe 78208d00 7efc2f07 00000081 7f016a3c 00000800 df80: bfbedf84 00000000 00000000 fffffffe 78208d00 7efc2f07 00000000 bfbedfa8 dfa0: 8000e800 80085dcc fffffffe 78208d00 78208d30 00000800 a8c82400 a8c82400 dfc0: fffffffe 78208d00 7efc2f07 00000081 00000002 00000000 78208008 00000800 dfe0: 7efc2e1c 7efc2ba8 76f5ca47 76edec7c 80000010 78208d30 00000000 00000000 Backtrace: [<80252b10>] (debugfs_remove_recursive+0x0/0x154) from [<80496ac4>] (snd_soc_unregister_card+0xa0/0xcc) r8:bec89000 r7:bef54440 r6:00000004 r5:bf9a4010 r4:bf9a4010 r3:00000000 [<80496a24>] (snd_soc_unregister_card+0x0/0xcc) from [<804a4b74>] (devm_card_release+0x14/0x18) r6:00000004 r5:bf9a4010 r4:bfbede60 r3:804a4b60 [<804a4b60>] (devm_card_release+0x0/0x18) from [<80328728>] (release_nodes+0x11c/0x1dc) [<8032860c>] (release_nodes+0x0/0x1dc) from [<80328cb0>] (devres_release_all+0x38/0x54) [<80328c78>] (devres_release_all+0x0/0x54) from [<8032561c>] (__device_release_driver+0x80/0xd4) r4:bf9a4010 r3:7f016000 [<8032559c>] (__device_release_driver+0x0/0xd4) from [<80325e84>] (driver_detach+0xbc/0xc0) r5:7f0169f8 r4:bf9a4010 [<80325dc8>] (driver_detach+0x0/0xc0) from [<80325410>] (bus_remove_driver+0x54/0x98) r6:78208d30 r5:00000000 r4:7f0169f8 r3:bee28900 [<803253bc>] (bus_remove_driver+0x0/0x98) from [<803264f8>] (driver_unregister+0x30/0x50) r4:7f0169f8 r3:beca8100 [<803264c8>] (driver_unregister+0x0/0x50) from [<80327098>] (platform_driver_unregister+0x14/0x18) r4:7f016a3c r3:7f01635c [<80327084>] (platform_driver_unregister+0x0/0x18) from [<7f016370>] (imx_sgtl5000_driver_exit+0x14/0x1c [snd_soc_imx_sgtl5000]) [<7f01635c>] (imx_sgtl5000_driver_exit+0x0/0x1c [snd_soc_imx_sgtl5000]) from [<80085ef0>] (SyS_delete_module+0x130/0x18c) [<80085dc0>] (SyS_delete_module+0x0/0x18c) from [<8000e800>] (ret_fast_syscall+0x0/0x48) r6:7efc2f07 r5:78208d00 r4:fffffffe Code: 889da9f8 e5983020 e3530000 089da9f8 (e5933038) ---[ end trace 825e7e125251a225 ]---

2 3

[alsa-devel] [PATCH 1/6] ASoC: Add resource managed snd_dmaengine_pcm_register()
by Lars-Peter Clausen 02 Dec '13

02 Dec '13

For many drivers using the generic dmaengine PCM driver one of the few (or the only) things left to do in the drivers remove function is to unregister the PCM device. This patch adds a resource managed version of snd_dmaengine_pcm_register() which makes it possible to simplify the remove function as well as the error path in the probe function for those drivers. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Moved snd_dmaengine_pcm_register() to soc-devres.c --- include/sound/dmaengine_pcm.h | 4 ++++ sound/soc/soc-devres.c | 41 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 45 insertions(+) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 1501731..4ef986c 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -140,6 +140,10 @@ int snd_dmaengine_pcm_register(struct device *dev, unsigned int flags); void snd_dmaengine_pcm_unregister(struct device *dev); +int devm_snd_dmaengine_pcm_register(struct device *dev, + const struct snd_dmaengine_pcm_config *config, + unsigned int flags); + int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config); diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index b1d7322..9998619 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> static void devm_component_release(struct device *dev, void *res) { @@ -84,3 +85,43 @@ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card) return ret; } EXPORT_SYMBOL_GPL(devm_snd_soc_register_card); + +#ifdef CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM + +static void devm_dmaengine_pcm_release(struct device *dev, void *res) +{ + snd_dmaengine_pcm_unregister(*(struct device **)res); +} + +/** + * devm_snd_dmaengine_pcm_register - resource managed dmaengine PCM registration + * @dev: The parent device for the PCM device + * @config: Platform specific PCM configuration + * @flags: Platform specific quirks + * + * Register a dmaengine based PCM device with automatic unregistration when the + * device is unregistered. + */ +int devm_snd_dmaengine_pcm_register(struct device *dev, + const struct snd_dmaengine_pcm_config *config, unsigned int flags) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_dmaengine_pcm_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_dmaengine_pcm_register(dev, config, flags); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_dmaengine_pcm_register); + +#endif -- 1.8.0

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[alsa-devel] [PATCH] ASoC: generic-dmaengine-pcm: Set BATCH flag when residue reporting is not supported
by Lars-Peter Clausen 02 Dec '13

02 Dec '13

For dmaengine drivers which do not support transfer residue reporting we update the PCM pointer with period granularity. Set the SNDRV_PCM_INFO_BATCH flag in this case to let userspace know about this. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-generic-dmaengine-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index cbc9c96..87e8635 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -137,6 +137,9 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea hw.buffer_bytes_max = SIZE_MAX; hw.fifo_size = dma_data->fifo_size; + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) + hw.info |= SNDRV_PCM_INFO_BATCH; + ret = dma_get_slave_caps(chan, &dma_caps); if (ret == 0) { if (dma_caps.cmd_pause) -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: Set SNDRV_PCM_INFO_JOINT_DUPLEX for PCMs with symmetry constraints
by Lars-Peter Clausen 02 Dec '13

02 Dec '13

If there are symmetry constraints between the playback and the capture channel set the SNDRV_PCM_INFO_JOINT_DUPLEX flag to let userspace know about this. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-pcm.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 22af038..eb340a8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -180,6 +180,21 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, return 0; } +static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver; + struct snd_soc_dai_driver *codec_driver = rtd->codec_dai->driver; + struct snd_soc_dai_link *link = rtd->dai_link; + + return cpu_driver->symmetric_rates || codec_driver->symmetric_rates || + link->symmetric_rates || cpu_driver->symmetric_channels || + codec_driver->symmetric_channels || link->symmetric_channels || + cpu_driver->symmetric_samplebits || + codec_driver->symmetric_samplebits || + link->symmetric_samplebits; +} + /* * List of sample sizes that might go over the bus for parameter * application. There ought to be a wildcard sample size for things @@ -316,6 +331,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) &cpu_dai_drv->capture); } + if (soc_pcm_has_symmetry(substream)) + runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; + ret = -EINVAL; if (!runtime->hw.rates) { printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n", -- 1.8.0

2 1

[alsa-devel] [PATCH] alsa-lib: dlmisc: Add a fake dl lib head file when dl lib is not supported by the toolchain.
by Sonic Zhang 02 Dec '13

02 Dec '13

From: sonic <sonic(a)htpc.(none)> The FLAT GNU toolchain doesn't include the dlfcn.h header file. Signed-off-by: Sonic Zhang <sonic.adi(a)gmail.com> --- include/dlmisc.h | 24 ++++++++++++++++++++++++ modules/mixer/simple/sbasedl.c | 4 ++++ src/mixer/simple_abst.c | 4 ++++ 3 files changed, 32 insertions(+) create mode 100644 include/dlmisc.h diff --git a/include/dlmisc.h b/include/dlmisc.h new file mode 100644 index 0000000..3be8d43 --- /dev/null +++ b/include/dlmisc.h @@ -0,0 +1,24 @@ +/* + * This library is free software; you can redistribute it and/or modify + * it under the terms of the GNU Lesser General Public License as + * published by the Free Software Foundation; either version 2.1 of + * the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#ifndef __ALSA_DLMISC_H +#define __ALSA_DLMISC_H + +#define RTLD_NOW 0x00002 +#define RTLD_GLOBAL 0x00100 + +#endif diff --git a/modules/mixer/simple/sbasedl.c b/modules/mixer/simple/sbasedl.c index 494802f..511bebf 100644 --- a/modules/mixer/simple/sbasedl.c +++ b/modules/mixer/simple/sbasedl.c @@ -27,7 +27,11 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <math.h> +#ifdef HAVE_LIBDL #include <dlfcn.h> +#else +#include <dlmisc.h> +#endif #include "config.h" #include "asoundlib.h" #include "mixer_abst.h" diff --git a/src/mixer/simple_abst.c b/src/mixer/simple_abst.c index 9e9aaf5..dd951fe 100644 --- a/src/mixer/simple_abst.c +++ b/src/mixer/simple_abst.c @@ -34,7 +34,11 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <math.h> +#ifdef HAVE_LIBDL #include <dlfcn.h> +#else +#include <dlmisc.h> +#endif #include "config.h" #include "asoundlib.h" #include "mixer_simple.h" -- 1.7.9.5

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[alsa-devel] [RFC 0/4] DT support for rx51-audio
by Sebastian Reichel 30 Nov '13

30 Nov '13

Hi, This patchset adds DT support in rx51-audio. I tested it on the Nokia N900 and was able to play sound with aplay using earphones and earspeaker. The Loudspeakers did not work. I don't know the reason. The patchset consists of 4 patches: 1. convert rx51-audio to snd_soc_register_card() 2. init audio in rx51 platform file 3. add some DT helpers to ASoC core 4. add DT support to rx51-audio Please note, that the patch is RFC level for now. For example non DT boot does not work, because I did not yet add pdata in boardcode. Proposed DTS node would look like this: sound: rx51-audio { compatible = "nokia,rx51-audio"; ti,mcbsp = <&mcbsp2>; ti,codec = <&tlv320aic3x>; ti,codec_aux = <&tlv320aic3x_aux>; ti,headset_amp = <&tpa6130a2>; tvout-selection-gpio = <&gpio2 8 GPIO_ACTIVE_HIGH>; /* 40 */ jack-detection-gpio = <&gpio6 17 GPIO_ACTIVE_HIGH>; /* 177 */ eci-sw-gpio = <&gpio6 22 GPIO_ACTIVE_HIGH>; /* 182 */ speaker-amp-gpio = <&twl_gpio 7 GPIO_ACTIVE_HIGH>; }; -- Sebastian Pali Rohár (2): ASoC: omap: rx51: Use snd_soc_register_card ARM: OMAP: rx51: Register audio device Sebastian Reichel (2): ASoC: Allow Aux Codecs to be specified using DT ASoC: RX-51: Add DT support to sound driver .../devicetree/bindings/sound/nokia,rx51.txt | 28 +++ .../devicetree/bindings/vendor-prefixes.txt | 1 + arch/arm/mach-omap2/board-rx51-peripherals.c | 11 ++ include/sound/rx51.h | 18 ++ include/sound/soc.h | 13 +- sound/soc/omap/omap-mcbsp.c | 5 +- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/rx51.c | 212 +++++++++++++++------ sound/soc/soc-core.c | 68 +++++-- 9 files changed, 277 insertions(+), 81 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nokia,rx51.txt create mode 100644 include/sound/rx51.h -- 1.8.4.rc3

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