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

[alsa-devel] [PATCH] ASoC: fsl-sai: Remove fsl_sai_remove()
by Xiubo Li 21 Dec '13

21 Dec '13

There is no need of this function and makes the code slightly shorter Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 11 ----------- 1 file changed, 11 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 50a797e..1868ec3 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -385,19 +385,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) return 0; } -static int fsl_sai_dai_remove(struct snd_soc_dai *cpu_dai) -{ - cpu_dai->playback_dma_data = NULL; - cpu_dai->capture_dma_data = NULL; - - snd_soc_dai_set_drvdata(cpu_dai, NULL); - - return 0; -} - static struct snd_soc_dai_driver fsl_sai_dai = { .probe = fsl_sai_dai_probe, - .remove = fsl_sai_dai_remove, .playback = { .channels_min = 1, .channels_max = 2, -- 1.8.4

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[alsa-devel] [PATCH 23/38] sound: soc: add missing put_device call
by Levente Kurusa 21 Dec '13

21 Dec '13

This is required so that we give up the last reference to the device. Signed-off-by: Levente Kurusa <levex(a)linux.com> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e53d87..21caa09 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -536,6 +536,7 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) err = device_register(&codec->ac97->dev); if (err < 0) { dev_err(codec->dev, "ASoC: Can't register ac97 bus\n"); + put_device(&codec->ac97->dev); codec->ac97->dev.bus = NULL; return err; } -- 1.8.3.1

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[alsa-devel] [PATCH 1/1] ASoC: pcm1792a: Remove redundant of_match_ptr helper
by Sachin Kamat 21 Dec '13

21 Dec '13

'pcm1792a_of_match' is always compiled in. Hence the helper macro is not needed. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/pcm1792a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 7146653a8e16..f4d9b769bcda 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -244,7 +244,7 @@ static struct spi_driver pcm1792a_codec_driver = { .driver = { .name = "pcm1792a", .owner = THIS_MODULE, - .of_match_table = of_match_ptr(pcm1792a_of_match), + .of_match_table = pcm1792a_of_match, }, .id_table = pcm1792a_spi_ids, .probe = pcm1792a_spi_probe, -- 1.7.9.5

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[alsa-devel] [PATCH 1/1] ASoC: sam9x5_wm8731: Remove redundant of_match_ptr helper
by Sachin Kamat 21 Dec '13

21 Dec '13

'sam9x5_wm8731_of_match' is always compiled in. Hence the helper macro is not needed. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/atmel/sam9x5_wm8731.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 7d6a9055874b..59df542afc7d 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -195,7 +195,7 @@ static struct platform_driver sam9x5_wm8731_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, - .of_match_table = of_match_ptr(sam9x5_wm8731_of_match), + .of_match_table = sam9x5_wm8731_of_match, }, .probe = sam9x5_wm8731_driver_probe, .remove = sam9x5_wm8731_driver_remove, -- 1.7.9.5

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[alsa-devel] snd_hda_intel and Beats Audio
by Dan McGhee 21 Dec '13

21 Dec '13

I have an HP Envy m6 Sleekbook model k010 with Beats Audio. My system is LFS-7.4 and BLFS-7.4. I have subscribed to this list in order to get the sound on my laptop working and also, hopefully, to provide you with info on what might need to be done in the hda_intel driver. If this list is not the appropriate one in which to do this, I will be more than happy to unsubscribe to it and get on the "users" list. In the hope that this situation is acceptable for this list--and please let me know if it's not--let me continue. Except for video replay, sound and printing my build is complete. I do not have any other sound applications or "helpers" installed, particularly PulseAudio, except gstreamer which is for dependencies. I'm using xcfe4 desktop. In my troubleshooting and research I discovered "hda-jack-retask." I installed it and have been "poking around" for awhile unsuccessfully. I guess I just don't understand the "pin out." This particular laptop has two speakers and a woofer. I ran "alsa-info.sh" and here is the section on the pins: > /sys/class/sound/hwC0D0/init_pin_configs: > 0x03 0x185600f0 > 0x05 0x585600f0 > 0x07 0x585600f0 > 0x09 0x585600f0 > 0x0b 0x585600f0 > 0x0d 0x585600f0 and > /sys/class/sound/hwC1D0/init_pin_configs: > 0x0a 0x03a11020 > 0x0b 0x0321101f > 0x0c 0x40f000f0 > 0x0d 0x92170110 > 0x0e 0x40f000f0 > 0x0f 0x40f000f0 > 0x10 0x40f000f0 > 0x11 0xd5a30130 > 0x1f 0x40f000f0 If you need the results of the whole script, I will run it again and provide the link. I just don't understand the mapping for the pins to be able to "override" them in hda-jack-retask. Here is more, I hope helpful info. Output from "speaker-test"--whether run as root or an unprivileged user: > speaker-test 1.0.27.2 > > Playback device is default > Stream parameters are 48000Hz, S16_LE, 1 channels > Using 16 octaves of pink noise > Channels count (1) not available for playbacks: Invalid argument > Setting of hwparams failed: Invalid argument From "aplay -L > null > Discard all samples (playback) or generate zero samples (capture) > hdmi:CARD=HDMI,DEV=0 > HDA ATI HDMI, HDMI 0 > HDMI Audio Output > sysdefault:CARD=Generic > HD-Audio Generic, 92HD91BXX Analog > Default Audio Device > front:CARD=Generic,DEV=0 > HD-Audio Generic, 92HD91BXX Analog > Front speakers > surround40:CARD=Generic,DEV=0 > HD-Audio Generic, 92HD91BXX Analog > 4.0 Surround output to Front and Rear speakers > surround41:CARD=Generic,DEV=0 > HD-Audio Generic, 92HD91BXX Analog > 4.1 Surround output to Front, Rear and Subwoofer speakers > surround50:CARD=Generic,DEV=0 > HD-Audio Generic, 92HD91BXX Analog > 5.0 Surround output to Front, Center and Rear speakers > surround51:CARD=Generic,DEV=0 > HD-Audio Generic, 92HD91BXX Analog > 5.1 Surround output to Front, Center, Rear and Subwoofer speakers > surround71:CARD=Generic,DEV=0 > HD-Audio Generic, 92HD91BXX Analog > 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers I will be grateful for any help. Also, once again, if this list is not the appropriate one, please let me know and I'll go to "users." Thanks, Dan

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[alsa-devel] [PATCH] ALSA: hda - Enable subwoofer on Dell Vostro 5460/5470
by David Henningsson 20 Dec '13

20 Dec '13

In this case, there are two DACs, and DAC 0x03 is mono. In order to make headphones and front speaker use DAC 0x02, and subwoofer use DAC 0x03, we artificially cut the connection from nodes 0x14 and 0x15 to node 0x03, so they can only use DAC 0x02. In addition, the 5460 and 5470 differs in the sense that 5470 also needs a headset mic patch, whereas 5460 has individual detection for headphone and headset mic. BugLink: https://bugs.launchpad.net/bugs/1211920 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- I'm adding Kailang to cc on this one; hopefully he'll scream if this patch is doing anything dangerous, such as if there's a risk to blow up the subwoofer. sound/pci/hda/patch_realtek.c | 43 +++++++++++++++++++++++++++++++++++------ 1 file changed, 37 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dfe4080..16c0caa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3770,10 +3770,14 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec, static void alc290_fixup_mono_speakers(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - if (action == HDA_FIXUP_ACT_PRE_PROBE) - /* Remove DAC node 0x03, as it seems to be - giving mono output */ - snd_hda_override_wcaps(codec, 0x03, 0); + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + /* DAC node 0x03 is giving mono output. We therefore want to + make sure 0x14 (front speaker) and 0x15 (headphones) use the + stereo DAC, while leaving 0x17 (bass speaker) for node 0x03. */ + hda_nid_t conn1[2] = { 0x0c }; + snd_hda_override_conn_list(codec, 0x14, 1, conn1); + snd_hda_override_conn_list(codec, 0x15, 1, conn1); + } } #if IS_ENABLED(CONFIG_THINKPAD_ACPI) @@ -3913,6 +3917,9 @@ enum { ALC282_FIXUP_ASUS_TX300, ALC283_FIXUP_INT_MIC, ALC290_FIXUP_MONO_SPEAKERS, + ALC290_FIXUP_MONO_SPEAKERS_HSJACK, + ALC290_FIXUP_SUBWOOFER, + ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC255_FIXUP_HEADSET_MODE, @@ -4235,9 +4242,31 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, + [ALC290_FIXUP_SUBWOOFER_HSJACK] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x17, 0x90170112 }, /* subwoofer */ + { } + }, + .chained = true, + .chain_id = ALC290_FIXUP_MONO_SPEAKERS_HSJACK, + }, + [ALC290_FIXUP_SUBWOOFER] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x17, 0x90170112 }, /* subwoofer */ + { } + }, + .chained = true, + .chain_id = ALC290_FIXUP_MONO_SPEAKERS, + }, [ALC290_FIXUP_MONO_SPEAKERS] = { .type = HDA_FIXUP_FUNC, .v.func = alc290_fixup_mono_speakers, + }, + [ALC290_FIXUP_MONO_SPEAKERS_HSJACK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc290_fixup_mono_speakers, .chained = true, .chain_id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, }, @@ -4282,6 +4311,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05da, "Dell Vostro 5460", ALC290_FIXUP_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e0, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -4303,10 +4333,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), + SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), + SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0629, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS), + SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK), SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0640, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - Don't be too specific for conflicting boost ctl names
by Takashi Iwai 20 Dec '13

20 Dec '13

When a boost control influences on multiple input paths, we shouldn't pick up the name string specific to one input but rather choose a more generic name. A problem seen often is that a single mic boost controls both internal and external mics although the driver picks up the very first name randomly like "Internal Mic Boost". This should have been "Mic Boost", instead. This patch tries to correct that behavior: when a boost control is available, check whether it conflicts with other inputs. If it does, use a common string ("Mic", "Line") as long as possible, or take a generic name "Input". Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 44 ++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 40 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8cebdcfdcfdc..e3f934703aa2 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3651,6 +3651,8 @@ static int parse_mic_boost(struct hda_codec *codec) struct hda_gen_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; struct hda_input_mux *imux = &spec->input_mux; + static const char *input_type_labels[3] = { "Mic", "Line", "Input" }; + int input_type_idxs[3] = {}; int i; if (!spec->num_adc_nids) @@ -3659,7 +3661,9 @@ static int parse_mic_boost(struct hda_codec *codec) for (i = 0; i < imux->num_items; i++) { struct nid_path *path; unsigned int val; - int idx; + int idx, type, j; + bool conflict; + const char *pfx; char boost_label[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; idx = imux->items[i].index; @@ -3678,11 +3682,43 @@ static int parse_mic_boost(struct hda_codec *codec) if (!val) continue; + /* check whether conflicting with other input paths */ + type = cfg->inputs[idx].type; + conflict = false; + for (j = i + 1; j < imux->num_items; j++) { + int idx_next = imux->items[i].index; + int type_next; + struct nid_path *path_next; + if (idx_next >= imux->num_items) + continue; + type_next = cfg->inputs[idx_next].type; + if (type_next > AUTO_PIN_LINE_IN) + continue; + path_next = get_input_path(codec, 0, j); + if (!path_next) + continue; + /* conflicting value? */ + if (look_for_boost_amp(codec, path_next) == val) { + conflict = true; + if (type != type_next) { + type = 2; /* generic input */ + break; + } + } + } + /* create a boost control */ + if (!conflict) { + pfx = spec->input_labels[idx]; + idx = spec->input_label_idxs[idx]; + } else { + pfx = input_type_labels[type]; + idx = input_type_idxs[type]++; + } + snprintf(boost_label, sizeof(boost_label), - "%s Boost Volume", spec->input_labels[idx]); - if (!add_control(spec, HDA_CTL_WIDGET_VOL, boost_label, - spec->input_label_idxs[idx], val)) + "%s Boost Volume", pfx); + if (!add_control(spec, HDA_CTL_WIDGET_VOL, boost_label, idx, val)) return -ENOMEM; path->ctls[NID_PATH_BOOST_CTL] = val; -- 1.8.5

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[alsa-devel] [PATCH] ASoC: Add resource managed devm_snd_soc_register_codec()
by Xiubo Li 20 Dec '13

20 Dec '13

Adds a resource managed version of snd_soc_register_codec() which makes it possible to simplify the remove function as well as the error path for codec drivers. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- include/sound/soc.h | 3 +++ sound/soc/soc-devres.c | 38 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 41 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5a049d9..60d6c67 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -378,6 +378,9 @@ int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_codec(struct device *dev); +int devm_snd_soc_register_codec(struct device *dev, + const struct snd_soc_codec_driver *codec_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai); int snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *cmpnt_drv, struct snd_soc_dai_driver *dai_drv, int num_dai); diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 7ac745d..3c8552f 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -125,3 +125,41 @@ int devm_snd_dmaengine_pcm_register(struct device *dev, EXPORT_SYMBOL_GPL(devm_snd_dmaengine_pcm_register); #endif + +static void devm_codec_release(struct device *dev, void *res) +{ + snd_soc_unregister_codec(*(struct device **)res); +} + +/** + * devm_snd_soc_register_codec - resource managed codec registration + * @codec_drv: codec to register + * @dai_drv: DAI to register + * @num_dai: Number of DAIs + * + * Register a codec with automatic unregistration when the device is + * unregistered. + */ +int devm_snd_soc_register_codec(struct device *dev, + const struct snd_soc_codec_driver *codec_drv, + struct snd_soc_dai_driver *dai_drv, + int num_dai) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_codec_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_soc_register_codec(dev, codec_drv, dai_drv, num_dai); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_soc_register_codec); -- 1.8.4

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Re: [alsa-devel] How to interrupt snd_pcm_writei in blocking mode?
by Giridhara 20 Dec '13

20 Dec '13

Takashi Iwai <tiwai <at> suse.de> writes: > > At Thu, 30 Jun 2005 20:42:28 +0200, > Hynek Hanke wrote: > > > > > > Hynek Hanke wrote: > > > > > I'm running it in two separate threads. One does all the preparations and calls > > > > > snd_pcm_writei() in blocking mode. The other one then decides it wants to stop > > > > > playing, so it calls snd_pcm_drop(). This causes the device to switch from > > > > > state RUNNING to state SETUP, but the snd_pcm_writei() hangs forever and never > > > > > returns. Why is this so? > > > Well, but pcm_lib.c::snd_pcm_lib_write1() should check the state SETUP > > > and break the loop immediately. Ditto for snd_pcm_lib_read1(). > > > The quick hack is below. Please give a shot. > > > > I tried to apply the patch to the 2.6.11 kernel but after > > compilation/instalation, there is no difference. It still hangs. Logfiles say > > it was in the state RUNNING just before snd_pcm_drop() was called and got into > > state SETUP immediatelly after. The logfile from the other thread shows > > snd_pcm_writei() never returns (not even after the sound would surely be > > completed). > > Hmm, then still something is missing. I'll check on it later. > > > So, what can I do? Do I really have to use the non-blocking mode or is this > > supposed to work (if I take care about all the other thread related issues > > myself)? My colleague is reporting it's working for him under some different > > configuration. > > The non-blocking mode and poll() should work (hopefully :) > If not, let me know. > > > Can I enable some debugging messages or something to help you to see what's > > wrong? > > Check the code flow in snd_pcm_lib_write1(). You can see the places > my patch tried to fix. Basically, the thread should be woken up and > get out of the loop. But according to your report, it didn't work as > expected by some reason... > > Takashi > > ------------------------------------------------------- > SF.Net email is sponsored by: Discover Easy Linux Migration Strategies > from IBM. Find simple to follow Roadmaps, straightforward articles, > informative Webcasts and more! Get everything you need to get up to > speed, fast. http://ads.osdn.com/?ad_id=7477&alloc_id=16492&op=click > Hello Takashi, We have the same problem. snd_pcm_writei blocks when we switch between playback and record continuously for several times. Is this issue is fixed in any kernel versions. I am running kernel version 2.6.31 Alsa version 1.0.18 is it a issue of buffers we allocate using snd_pcm_hw_params_set_period_size_near whats the way to calculate this value and how should be the ideal value. I appreciate any help/suggestions in this regard. Thanks and Regards, Giri

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[alsa-devel] [PATCH v3 0/9] ASoC: fsl-ssi: offline/online configuration and cleanups
by Markus Pargmann 20 Dec '13

20 Dec '13

Hi, This patch series fixes some fsl-ssi code that does not act exactly as it is described in the reference manuals. The reference manuals before imx51 do mention that some register bits, including TDMAE/RDMAE, should not be changed while the SSI unit is enabled (SSIEN). At the same time the SDMA unit has a undefined request handling if there are DMA requests before the SDMA engine and its channel is configured. The reference manual states that SSI DMA requests are sent as soon as the FIFO and the DMAE bits are enabled. It does not mention a dependency to SCR's TE/RE bit. To avoid sending DMA requests before a channel is configured, we have to seperate the fsl-ssi behavior into offline and online configuration. SoCs before imx51 have to use offline configuration while later SoC versions can use online reconfiguration. This series adds support for online configuration and cleans up the configuration register code especially in fsl_ssi_trigger. At the end we have a seperation between enable/disable logic and the actual configuration register values used to enable/disable TX/RX. v3 adds another fixup for the interrupt stats. The interrupt handler currently writes back a subset of the interrupt flags to clear the interrupts in the status register. This is not implemented on every processor. For example imx27 does not allow writebacks, imx53 does not allow writeback of RFRC and TFRC. The interrupt stats were disabled for imx by default. 'Fix interrupt mapping and release' enables it for mx53 again and changes the writeback flags. I tested this series on mx53. The series is based on Mark Brown's topic/fsl branch. Regards, Markus Changes in v3: - Rebase on latest topic/fsl - Added patch to cleanup probe function error handling - Fix interrupt stats for imx53 - Replace imx50-ssi by imx51-ssi Markus Pargmann (9): ASoC: fsl-ssi: Fix probe error handling ASoC: fsl-ssi: Move sysfs stats to debugfs ASoC: fsl-ssi: Add imx51-ssi and of_device_id matching ASoC: fsl-ssi: Fix interrupt mapping and release ASoC: fsl-ssi: Add offline_config flag ASoC: fsl-ssi: Add configuration helper functions ASoC: fsl-ssi: Move RX/TX configuration to seperate functions ASoC: fsl-ssi: Drop ac97 specific trigger function ARM: DTS: imx5* imx6*, use imx51-ssi arch/arm/boot/dts/imx51.dtsi | 3 +- arch/arm/boot/dts/imx53.dtsi | 10 +- arch/arm/boot/dts/imx6qdl.dtsi | 12 +- arch/arm/boot/dts/imx6sl.dtsi | 12 +- sound/soc/fsl/fsl_ssi.c | 539 +++++++++++++++++++++++++++-------------- 5 files changed, 387 insertions(+), 189 deletions(-) -- 1.8.5.1

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