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

Re: [alsa-devel] [PATCH] ALSA: hda - Do not assign streams in reverse order
by Takashi Iwai 08 Apr '14

08 Apr '14

At Mon, 7 Apr 2014 22:36:38 +0300, Anssi Hannula wrote: > > Currently stream numbers are assigned in reverse order. > > Unfortunately commit 7546abfb8e1f9933b5 ("ALSA: hda - Increment > default stream numbers for AMD HDMI controllers") assumed this was not > the case (specifically, it had the "old cards had single device only" > => "extra unused stream numbers do not matter" assumption), causing > non-working audio regressions for AMD Radeon HDMI users on v3.14. > > Change the stream numbers to be assigned in increasing order instead. > > Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=77002 > Reported-by: Christian Güdel <cg(a)dmesg.ch> > Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> > Tested-by: Christian Güdel <cg(a)dmesg.ch> > Cc: Alex Deucher <alexander.deucher(a)amd.com> > Cc: <stable(a)vger.kernel.org> # 3.14 > --- > > Takashi, please check I didn't miss anything (I'm not that familiar > with this low-level HDA code). :) In that case, let's revert the commit instead. There was a reason to behave in that way, although I don't remember well. That said, this change might lead to another regression. Thus, if a fix is needed for stable, the revert would be the best choice as a first aid. Of course, this change is good to have, so let's implement this for 3.15 and see whether it works well for all nasty controllers. Besides, this patch can't be applied any longer to the current code base since it's been rewritten a lot; most of codes have been moved to another file hda_controller.c. Could you rebase the patch for the latest tree? Meanwhile I'll prepare a revert that is applied only to 3.14.x. thanks, Takashi > > > sound/pci/hda/hda_intel.c | 9 +++++---- > 1 file changed, 5 insertions(+), 4 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index e354ab1ec20f..7a7b7b10dddf 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -1856,13 +1856,14 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream) > struct azx_dev *azx_dev = &chip->azx_dev[dev]; > dsp_lock(azx_dev); > if (!azx_dev->opened && !dsp_is_locked(azx_dev)) { > - res = azx_dev; > - if (res->assigned_key == key) { > - res->opened = 1; > - res->assigned_key = key; > + if (azx_dev->assigned_key == key) { > + azx_dev->opened = 1; > + azx_dev->assigned_key = key; > dsp_unlock(azx_dev); > return azx_dev; > } > + if (!res) > + res = azx_dev; > } > dsp_unlock(azx_dev); > } > -- > 1.8.4.5 >

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Re: [alsa-devel] Audio codec with multiple I2C devices
by Lars-Peter Clausen 08 Apr '14

08 Apr '14

On 04/07/2014 09:29 PM, Gilles wrote: > On Apr 7, 2014, at 12:03 , Lars-Peter Clausen <lars(a)metafoo.de> wrote: > >> On 04/07/2014 08:32 PM, Gilles wrote: >> [...] >>> Doesn't it make perfect sense to have the Volume be a control of the Codec driver even if it's provided by a separate chip on the same sound card? >> >> Not if they are standalone components that can also be used in different configurations on different sound cards. E.g. you code suggests that you are using a AK4113 and a DAC121C085. The CODEC driver is not the sound card driver, the sound card driver is the piece of code that glues all the different components that are found on the sound card together. >> >> i2c_new_dummy() is typically used for devices with multiple I2C addresses. Or where you have e.g. multiple devices with different addresses on the same die so they become a inseparable unit. >> >> - Lars > > Yes, but the functionality is very specific to the way the components are wired on the hardware. They are wired to work in conjunction with the ASIC Codec. > > The DAC121C085 is a very specific case as the DAC is actually wired to control the voltage of 4 quadrant multiplier which is hardwired to the Codec ASIC. This is a prototype. In the final product, the DAC chip may be built into the ASIC. For this one, I feel like the "Volume" should be a control of the same driver as the ASIC. > > This said I see your point about the AK4113 which is a generic multi channel receiver. In this specific application it is wired so only inputs 1,2 and 3 are hard wired (this actually isn't a sound card, but rather a finished product using Linux as the SoC OS as can be seen on the home page of http://birdland.com right now). > > There is already a working prototype, but as of now, the inputs are selected by the application talking directly to the I2C AKM. I'm just in the process of streamlining the code, and naturally, I envisioned the input selection be part of the sound card codec as I'd much rather have the sound card have a enum control with the 4 inputs shown by name show under amixer (this would enable my application to be more generic). > > Can you please point me to an example of a Sound board driver that uses multiple codecs? I suppose I can modify my sound board driver to use a codec for the audio and another for the AKM input selector). Anything that uses aux devs. E.g. samsung/speyside.c There is also this series https://lkml.org/lkml/2014/4/5/92 which adds DT support for aux devs. - Lars

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[alsa-devel] [PATCH 00/44 v3] Enhancement of support for Firewire devices
by Takashi Sakamoto 08 Apr '14

08 Apr '14

This series is an update for my previous series: [alsa-devel] [GIT PULL][PATCH 00/39 v2] Enhancement of support for Firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073868.html Updates since previous pull-request: firewire-lib: - remove packet sorting - remove adding transfer delay to out-packets - fix a bug of MIDI capturing for several ports - add detecting discontinuity of packets - add a support for deferred transaction - add some flags for device's quirk related to AMDTP packets - improve return value of some AV/C general commands fireworks: - add a workaround for out-of-specification dbc meaning - add a workaround for AudioFire9 firmware - add a workaround for firmware version 5.5 bebob: - add some flags for device's quirk related to AMDTP packets - add a workaround for discontinuity of packets at bus reset - add workarounds for M-Audio spefial Firewire series This time I intentionally add no restriction of version for Fireworks because I confirmed that my model doesn't always transmit packets with disorder. I expect users to retry starging streams several times till detecting no discontinuity of packets. [alsa-devel] [PATCH 09/39] firewire-lib: Add sort function for transmitted packet http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074157.html I note that a workaround for M-Audio special Firewire series at [44/44] relates to Linux Firewire subsystem. Please follow: Generating bus-reset by kernel module in the other subsystem http://sourceforge.net/p/linux1394/mailman/linux1394-devel/thread/5326B651.… ---------------------------------------------------------------- The following changes since commit 2df6742f613840a0b0a1590fb28f7af5b058a673: ALSA: hda - verify pin:cvt connection on preparing a stream for Intel HDMI codec (2014-03-20 07:36:17 +0100) are available in the git repository at: https://github.com/takaswie/sound.git request for you to fetch changes up to aa0564fdd013e5754b73ba89b50c722bbedbc4ac: firewire/bebob: Add a workaround for M-Audio special Firewire series (2014-03-21 19:35:04 +0900) ---------------------------------------------------------------- Takashi Sakamoto (44): firewire-lib: Add macros instead of fixed value for AMDTP firewire-lib: Add 'direction' member to 'amdtp_stream' structure firewire-lib: Split some codes into functions to reuse for both streams firewire-lib: Add support for AMDTP in-stream and PCM capture firewire-lib: Add support for MIDI capture/playback firewire-lib: Give syt value as parameter to handle_out_packet() firewire-lib: Add support for duplex streams synchronization in blocking mode firewire-lib: Add support for channel mapping firewire-lib: Restrict calling flush_context_completion() when context exists firewire-lib: Rename macros, variables and functions for CMP firewire-lib: Add 'direction' member to 'cmp_connection' structure firewire-lib: Add handling output connection by CMP firewire-lib: Add a new function to check others' connection firewire-lib: Add support for deferred transaction firewire-lib: Add some AV/C general commands fireworks: Add skelton for Fireworks based devices fireworks: Add transaction and some commands fireworks: Add connection and stream management fireworks/firewire-lib: Add a quirk for empty packet with TAG0 fireworks/firewire-lib: Add a quirk for the meaning of dbc fireworks/firewire-lib: Add a quirk for wrong dbs in tx packets fireworks/firewire-libs: Add a quirk for fixed interval of reported dbc fireworks: Add proc interface for debugging purpose fireworks: Add MIDI interface fireworks/firewire-lib: Add a quirk for data blocks for MIDI in out-stream fireworks: Add PCM interface fireworks: Add hwdep interface fireworks: Add command/response functionality into hwdep interface bebob: Add skelton for BeBoB based devices bebob: Add commands and connections/streams management bebob/firewire-lib: Add a quirk for discontinuity at bus reset bebob: Add proc interface for debugging purpose bebob: Add MIDI interface bebob: Add PCM interface bebob: Add hwdep interface bebob: Prepare for device specific operations bebob: Add support for Terratec PHASE, EWS series and Aureon bebob: Add support for Yamaha GO series bebob: Add support for Focusrite Saffire/SaffirePro series bebob: Add support for M-Audio usual Firewire series bebob: Add support for M-Audio special Firewire series bebob/firewire-lib: Add a quirk of wrong dbc in empty packet for M-Audio special Firewire series bebob: Send a cue to load firmware for M-Audio Firewire series firewire/bebob: Add a workaround for M-Audio special Firewire series drivers/firewire/core.h | 1 - include/linux/firewire.h | 3 + include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 22 +- sound/firewire/Kconfig | 59 ++ sound/firewire/Makefile | 2 + sound/firewire/amdtp.c | 668 ++++++++++++---- sound/firewire/amdtp.h | 124 ++- sound/firewire/bebob/Makefile | 4 + sound/firewire/bebob/bebob.c | 470 ++++++++++++ sound/firewire/bebob/bebob.h | 257 +++++++ sound/firewire/bebob/bebob_command.c | 311 ++++++++ sound/firewire/bebob/bebob_focusrite.c | 287 +++++++ sound/firewire/bebob/bebob_hwdep.c | 197 +++++ sound/firewire/bebob/bebob_maudio.c | 832 ++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 168 ++++ sound/firewire/bebob/bebob_pcm.c | 451 +++++++++++ sound/firewire/bebob/bebob_proc.c | 201 +++++ sound/firewire/bebob/bebob_stream.c | 929 +++++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 75 ++ sound/firewire/bebob/bebob_yamaha.c | 54 ++ sound/firewire/cmp.c | 203 +++-- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 2 +- sound/firewire/fcp.c | 192 ++++- sound/firewire/fcp.h | 21 + sound/firewire/fireworks/Makefile | 4 + sound/firewire/fireworks/fireworks.c | 331 ++++++++ sound/firewire/fireworks/fireworks.h | 227 ++++++ sound/firewire/fireworks/fireworks_command.c | 397 ++++++++++ sound/firewire/fireworks/fireworks_hwdep.c | 294 +++++++ sound/firewire/fireworks/fireworks_midi.c | 179 +++++ sound/firewire/fireworks/fireworks_pcm.c | 477 ++++++++++++ sound/firewire/fireworks/fireworks_proc.c | 233 ++++++ sound/firewire/fireworks/fireworks_stream.c | 347 +++++++++ sound/firewire/fireworks/fireworks_transaction.c | 327 ++++++++ sound/firewire/speakers.c | 48 +- 37 files changed, 8157 insertions(+), 258 deletions(-) create mode 100644 sound/firewire/bebob/Makefile create mode 100644 sound/firewire/bebob/bebob.c create mode 100644 sound/firewire/bebob/bebob.h create mode 100644 sound/firewire/bebob/bebob_command.c create mode 100644 sound/firewire/bebob/bebob_focusrite.c create mode 100644 sound/firewire/bebob/bebob_hwdep.c create mode 100644 sound/firewire/bebob/bebob_maudio.c create mode 100644 sound/firewire/bebob/bebob_midi.c create mode 100644 sound/firewire/bebob/bebob_pcm.c create mode 100644 sound/firewire/bebob/bebob_proc.c create mode 100644 sound/firewire/bebob/bebob_stream.c create mode 100644 sound/firewire/bebob/bebob_terratec.c create mode 100644 sound/firewire/bebob/bebob_yamaha.c create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_hwdep.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c create mode 100644 sound/firewire/fireworks/fireworks_transaction.c -- 1.8.3.2

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[alsa-devel] Audio codec with multiple I2C devices
by Gilles 07 Apr '14

07 Apr '14

Folks, I'm writing an ALSA SoC codec which uses a 3 distinct I2C devices (a DAC, a CAG for volume and a input selector) all 3 at different I2C addresses. My existing code relies on DT to declare the primary I2C address (in this case the DAC) and it works. But I now need to add support for the extended functionality. Can someone point me to the best way to declare the second I2C address in the DT and make use of it in the driver? I know I could always hard code it and add a new i2c_new_device() in the probe() into the private data but I'm trying to make it a bit more DT friendly. Not having much luck on finding examples of this. I'm attaching the existing code code below if anyone wants to reference to snipets of it. As you can see, the set/get volume are declared as _EXT since the volume is controlled by a secondary GC I2C device. Similarily, the input selector talks to a third I2C device. Any help would be appreciated. Thanks! /* // ALSA SoC codec driver for Birdland Audio SDM-50 Digital Amplifier // Copyright (C) 2012 Birdland Audio - http://birdland.com // Author: Gilles Gameiro <gilles(a)whospot.com> // */ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/gpio.h> #include <linux/of_gpio.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/initval.h> #include <sound/tlv.h> #define DRV_NAME "sdm2x-audio-codec" #define BIRDLAND_MODEL_250 0 #define SDM2X_REG0_VOLUME 0 #define SDM2X_REG1_MUXCLK 1 #define SDM2X_INPUT_EXTERNAL 0x80 #define SDM2X_PLAYRATE_48000 0x08 #define SDM2X_RATES ( SNDRV_PCM_RATE_32000 \ | SNDRV_PCM_RATE_44100 \ | SNDRV_PCM_RATE_48000 \ | SNDRV_PCM_RATE_64000 \ | SNDRV_PCM_RATE_88200 \ | SNDRV_PCM_RATE_96000 \ | SNDRV_PCM_RATE_176400 \ | SNDRV_PCM_RATE_192000 ) #define SDM2X_FORMATS ( SNDRV_PCM_FMTBIT_S16_LE \ | SNDRV_PCM_FMTBIT_S20_3LE \ | SNDRV_PCM_FMTBIT_S24_3LE \ | SNDRV_PCM_FMTBIT_S32_LE ) /* codec private data */ struct sdm2x_priv { struct snd_soc_codec *codec; enum snd_soc_control_type control_type; u16 model; unsigned int sysclk; u8 input_cache; // Caches the back panel input selected (Controlled by AK4113) u8 volume_cache; // Caches the current value of the volume (Controlled by DAC121C085) }; // FPGA I2C Slave IP is write only. Use register cache #define SDM2X_RCACHE_SIZE 4 static const u8 sdm2x_reg_cache[SDM2X_RCACHE_SIZE] = { 0x00, (SDM2X_INPUT_EXTERNAL | SDM2X_PLAYRATE_48000), 0x00, 0x00, }; static int sdm250_get_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); ucontrol->value.integer.value[0] = sdm2x->volume_cache; return 0; } static int sdm250_set_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); int volume = ucontrol->value.integer.value[0] & 0x7f; sdm2x->volume_cache = volume; // TODO: Write it to GA return 0; } static int sdm250_get_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); ucontrol->value.integer.value[0] = sdm2x->input_cache; return 0; } static int sdm250_set_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); int input = ucontrol->value.integer.value[0] & 0x7f; sdm2x->input_cache = volume; // TODO: Write it to AK4113 return 0; } /* General Volume -127dB to 0dB on 1dB steps */ static DECLARE_TLV_DB_SCALE(general_volume_tlv, -12700, 100, 0); /* Enum declarations */ static const char *sdm2x_srcin[] = {"Internal", "External"}; static const struct soc_enum sdm2x_srcin_enum = SOC_ENUM_SINGLE(SDM2X_REG1_MUXCLK, 7, 2, sdm2x_srcin); static const char *sdm2x_i2sin[] = {"Optical 1", "Optical 2", "Optical 3", "XLR" }; static const struct soc_enum sdm2x_i2sin_enum = SOC_ENUM_SINGLE(SDM2X_REG1_MUXCLK, 0, 3, sdm2x_i2sin); static const struct snd_kcontrol_new sdm2x_snd_controls[] = { /* General Volume: This is set outside of the FPGA so use getter/setter functions */ SOC_SINGLE_EXT_TLV("Amplifier Volume", SDM2X_REG0_VOLUME, 0, 127, 0, sdm250_get_volume, sdm250_set_volume, general_volume_tlv), /* FPGA Manged Controls (primary I2C of this driver) */ SOC_SINGLE("Digital Mute", SDM2X_REG0_VOLUME, 7, 1, 0), // TODO: Make this a DAI op? SOC_ENUM("SRC Input Source", sdm2x_srcin_enum), /* Back Pannel Controls (AK4113) */ SOC_ENUM_EXT("Back Panel Input", sdm2x_i2sin_enum, sdm250_get_input, sdm250_set_input) }; static int sdm2x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; //struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); int fsref; unsigned char divreg; // Set Data word length not needed in I2S) fsref = params_format(params); // Set Sampling Rate: I haven't figured out how to make the mcasp I2S master for TX and slave for RX so let the FPGA generate the clocks Sampling Rate switch (fsref) { case SNDRV_PCM_RATE_32000: divreg = 0x0C; break; case SNDRV_PCM_RATE_44100: case SNDRV_PCM_RATE_48000: divreg = 0x08; break; case SNDRV_PCM_RATE_64000: divreg = 0x06; break; case SNDRV_PCM_RATE_88200: case SNDRV_PCM_RATE_96000: divreg = 0x04; break; case SNDRV_PCM_RATE_176400: case SNDRV_PCM_RATE_192000: divreg = 0x02; break; default: printk(KERN_ERR "sdm2x-codec.%s(): Requested rate 0x%08x not supported\n", __func__, fsref); return -EINVAL; } if (fsref % 11025 == 0) divreg |= 0x10; snd_soc_update_bits(codec, SDM2X_REG1_MUXCLK, 0x1F, divreg); return 0; } // DAI Ops static int sdm2x_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { //struct snd_soc_codec *codec = codec_dai->codec; //struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); // SDM-50 only supports vanilla I2S format. switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; default: printk(KERN_ERR "sdm2x-codec.%s(): Requested Format 0x%08x not supported\n", __func__, (fmt & SND_SOC_DAIFMT_FORMAT_MASK)); return -EINVAL; } // SDM-50 Uses normal I2S Clock polarities switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: break; default: printk(KERN_ERR "sdm2x-codec.%s(): Requested Clock Polarity 0x%08x not supported\n", __func__, (fmt & SND_SOC_DAIFMT_INV_MASK)); return -EINVAL; } // And SDM-50 is the I2S Clock Master for both Bit and Frame switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case (SND_SOC_DAIFMT_CBM_CFM): break; default: printk(KERN_ERR "sdm2x-codec.%s(): Requested Master/Slave Clock 0x%08x not supported\n", __func__, (fmt & SND_SOC_DAIFMT_MASTER_MASK)); return -EINVAL; } return 0; } static int sdm2x_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); sdm2x->sysclk = freq; return 0; } static const struct snd_soc_dai_ops sdm2x_dai_ops = { .hw_params = sdm2x_hw_params, .set_fmt = sdm2x_set_dai_fmt, .set_sysclk = sdm2x_set_dai_sysclk, //.digital_mute = sdm2x_mute, }; static struct snd_soc_dai_driver sdm2x_dai = { .name = "sdm2x-hifi", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = SDM2X_RATES, .formats = SDM2X_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = SDM2X_RATES, .formats = SDM2X_FORMATS, }, .ops = &sdm2x_dai_ops, .symmetric_rates = 1, }; static int sdm2x_suspend(struct snd_soc_codec *codec) { return 0; } static int sdm2x_resume(struct snd_soc_codec *codec) { return 0; } /* PROBE: Setup private data, initialize codec and attach it to ASoC */ static int sdm2x_probe(struct snd_soc_codec *codec) { int ret; struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); sdm2x->codec = codec; // This belongs in the core but not sure if 3.8.13 supports it already.. ret = snd_soc_codec_set_cache_io(codec, 8, 8, sdm2x->control_type); if (ret != 0) { printk(KERN_ERR "sdm2x-codec.%s(): Failed to set cache I/O: %d\n", __func__, ret); return ret; } //codec->cache_only = 1; // Init... snd_soc_write(codec, SDM2X_REG0_VOLUME, 0); snd_soc_write(codec, SDM2X_REG1_MUXCLK, (SDM2X_INPUT_EXTERNAL | SDM2X_PLAYRATE_48000)); //snd_soc_update_bits(codec, SDM250_REG2_STATUS, SOFT_RESET, SOFT_RESET); // Auto Clears after reset complete (might not work with cached registers) // Add AMIXER controls (Volume, etc..) snd_soc_add_codec_controls(codec, sdm2x_snd_controls, ARRAY_SIZE(sdm2x_snd_controls)); return 0; } static int sdm2x_remove(struct snd_soc_codec *codec) { // struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); // TODO: Where is private data freed up? return 0; } static struct snd_soc_codec_driver soc_codec_dev_sdm2x = { .reg_cache_size = ARRAY_SIZE(sdm2x_reg_cache), .reg_word_size = sizeof(u8), .reg_cache_default = sdm2x_reg_cache, .probe = sdm2x_probe, .remove = sdm2x_remove, .suspend = sdm2x_suspend, .resume = sdm2x_resume, }; static const struct i2c_device_id sdm2x_i2c_id[] = { { "sdm2x", BIRDLAND_MODEL_250 }, { } }; MODULE_DEVICE_TABLE(i2c, sdm2x_i2c_id); /* If the i2c layer weren't so broken, we could pass this kind of data around */ static int sdm2x_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct sdm2x_priv *sdm2x; struct device_node *np = i2c->dev.of_node; int ret; sdm2x = devm_kzalloc(&i2c->dev, sizeof(struct sdm2x_priv), GFP_KERNEL); if (sdm2x == NULL) { printk(KERN_ERR "sdm2x-codec.%s(): Failed to set cache I/O\n", __func__); return -ENOMEM; } // This driver doesn't support the old style platform data if (!np) { printk(KERN_ERR "sdm2x-codec.%s(): missing DT data info\n", __func__); return -EINVAL; } sdm2x->control_type = SND_SOC_I2C; i2c_set_clientdata(i2c, sdm2x); sdm2x->model = id->driver_data; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_sdm2x, &sdm2x_dai, 1); return ret; } static int sdm2x_i2c_remove(struct i2c_client *client) { // TODO: Where is private data freed up? snd_soc_unregister_codec(&client->dev); return 0; } #if defined(CONFIG_OF) static const struct of_device_id sdm2x_of_match[] = { { .compatible = "birdland,sdm2x", }, {}, }; MODULE_DEVICE_TABLE(of, sdm2x_of_match); #endif /* machine i2c codec control layer */ static struct i2c_driver sdm2x_i2c_driver = { .driver = { .name = "sdm2x-codec", .owner = THIS_MODULE, .of_match_table = of_match_ptr(sdm2x_of_match), }, .probe = sdm2x_i2c_probe, .remove = sdm2x_i2c_remove, .id_table = sdm2x_i2c_id, }; module_i2c_driver(sdm2x_i2c_driver); MODULE_DESCRIPTION("ASoC SDM2x codec driver"); MODULE_AUTHOR("Gilles Gameiro"); MODULE_LICENSE("GPL");

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[alsa-devel] [PATCH v9] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 07 Apr '14

07 Apr '14

This patch adds initial support for the Behringer BCD2000 USB DJ controller. At the moment, only the MIDI part of the device is working, i.e. knobs, buttons and LEDs. I also plan to add support for the audio part, but I assume that this will require more effort than the rather simple MIDI interface. Progress can be tracked at https://github.com/anyc/snd-usb-bcd2000. Signed-off-by: Mario Kicherer <dev(a)kicherer.org> --- Changes since v8: - module_usb_driver() macro for module initialization - improved some comments Changes since v7: - replaced snd_card_create with snd_card_new Changes since v6: - applied more style improvements Changes since v5: - use kernel bitmap functions for devices_used Changes since v4: - devices_used as array to support arbitrary number of SNDRV_CARDS - removed unused array "enable" Changes since v3: - applied style and snd_printk changes as suggested by Daniel Mack Changes since v2: - applied more changes from Daniel Mack and Clemens Ladisch Changes since v1: - fixed the various code style issues, thanks to Daniel Mack and checkpatch.pl. sound/usb/Kconfig | 13 ++ sound/usb/Makefile | 2 +- sound/usb/bcd2000/Makefile | 3 + sound/usb/bcd2000/bcd2000.c | 461 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 478 insertions(+), 1 deletion(-) create mode 100644 sound/usb/bcd2000/Makefile create mode 100644 sound/usb/bcd2000/bcd2000.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index e05a86b..d393153 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -147,5 +147,18 @@ config SND_USB_HIFACE To compile this driver as a module, choose M here: the module will be called snd-usb-hiface. +config SND_BCD2000 + tristate "Behringer BCD2000 MIDI driver" + select SND_RAWMIDI + help + Say Y here to include MIDI support for the Behringer BCD2000 DJ + controller. + + Audio support is still work-in-progress at + https://github.com/anyc/snd-usb-bcd2000 + + To compile this driver as a module, choose M here: the module + will be called snd-bcd2000. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abe668f..2b92f0d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ diff --git a/sound/usb/bcd2000/Makefile b/sound/usb/bcd2000/Makefile new file mode 100644 index 0000000..f09ccc0 --- /dev/null +++ b/sound/usb/bcd2000/Makefile @@ -0,0 +1,3 @@ +snd-bcd2000-y := bcd2000.o + +obj-$(CONFIG_SND_BCD2000) += snd-bcd2000.o \ No newline at end of file diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c new file mode 100644 index 0000000..820d6ca --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,461 @@ +/* + * Behringer BCD2000 driver + * + * Copyright (C) 2014 Mario Kicherer (dev(a)kicherer.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 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 General Public License for more details. + */ + +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/bitmap.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> + +#define PREFIX "snd-bcd2000: " +#define BUFSIZE 64 + +static struct usb_device_id id_table[] = { + { USB_DEVICE(0x1397, 0x00bd) }, + { }, +}; + +static unsigned char device_cmd_prefix[] = {0x03, 0x00}; + +static unsigned char bcd2000_init_sequence[] = { + 0x07, 0x00, 0x00, 0x00, 0x78, 0x48, 0x1c, 0x81, + 0xc4, 0x00, 0x00, 0x00, 0x5e, 0x53, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff, 0x6c, 0xf3, 0x90, 0xff, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, + 0x18, 0xfa, 0x11, 0xff, 0x14, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0xf2, 0x34, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff +}; + +struct bcd2000 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf; + int card_index; + + int midi_out_active; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_receive_substream; + struct snd_rawmidi_substream *midi_out_substream; + + unsigned char midi_in_buf[BUFSIZE]; + unsigned char midi_out_buf[BUFSIZE]; + + struct urb *midi_out_urb; + struct urb *midi_in_urb; + + struct usb_anchor anchor; +}; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + +static DEFINE_MUTEX(devices_mutex); +DECLARE_BITMAP(devices_used, SNDRV_CARDS); +static struct usb_driver bcd2000_driver; + +#ifdef CONFIG_SND_DEBUG +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) +{ + print_hex_dump(KERN_DEBUG, prefix, + DUMP_PREFIX_NONE, 16, 1, + buf, len, false); +} +#else +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) {} +#endif + +static int bcd2000_midi_input_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_input_close(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +/* (de)register midi substream from client */ +static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + bcd2k->midi_receive_substream = up ? substream : NULL; +} + +static void bcd2000_midi_handle_input(struct bcd2000 *bcd2k, + const unsigned char *buf, unsigned int buf_len) +{ + unsigned int payload_length, tocopy; + struct snd_rawmidi_substream *midi_receive_substream; + + midi_receive_substream = ACCESS_ONCE(bcd2k->midi_receive_substream); + if (!midi_receive_substream) + return; + + bcd2000_dump_buffer(PREFIX "received from device: ", buf, buf_len); + + if (buf_len < 2) + return; + + payload_length = buf[0]; + + /* ignore packets without payload */ + if (payload_length == 0) + return; + + tocopy = min(payload_length, buf_len-1); + + bcd2000_dump_buffer(PREFIX "sending to userspace: ", + &buf[1], tocopy); + + snd_rawmidi_receive(midi_receive_substream, + &buf[1], tocopy); +} + +static void bcd2000_midi_send(struct bcd2000 *bcd2k) +{ + int len, ret; + struct snd_rawmidi_substream *midi_out_substream; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + + midi_out_substream = ACCESS_ONCE(bcd2k->midi_out_substream); + if (!midi_out_substream) + return; + + /* copy command prefix bytes */ + memcpy(bcd2k->midi_out_buf, device_cmd_prefix, + sizeof(device_cmd_prefix)); + + /* + * get MIDI packet and leave space for command prefix + * and payload length + */ + len = snd_rawmidi_transmit(midi_out_substream, + bcd2k->midi_out_buf + 3, BUFSIZE - 3); + + if (len < 0) + dev_err(&bcd2k->dev->dev, "%s: snd_rawmidi_transmit error %d\n", + __func__, len); + + if (len <= 0) + return; + + /* set payload length */ + bcd2k->midi_out_buf[2] = len; + bcd2k->midi_out_urb->transfer_buffer_length = BUFSIZE; + + bcd2000_dump_buffer(PREFIX "sending to device: ", + bcd2k->midi_out_buf, len+3); + + /* send packet to the BCD2000 */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s (%p): usb_submit_urb() failed, ret=%d, len=%d\n", + __func__, midi_out_substream, ret, len); + else + bcd2k->midi_out_active = 1; +} + +static int bcd2000_midi_output_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (bcd2k->midi_out_active) { + usb_kill_urb(bcd2k->midi_out_urb); + bcd2k->midi_out_active = 0; + } + + return 0; +} + +/* (de)register midi substream from client */ +static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (up) { + bcd2k->midi_out_substream = substream; + /* check if there is data userspace wants to send */ + if (!bcd2k->midi_out_active) + bcd2000_midi_send(bcd2k); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static void bcd2000_output_complete(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + + bcd2k->midi_out_active = 0; + + if (urb->status) + dev_warn(&urb->dev->dev, + PREFIX "output urb->status: %d\n", urb->status); + + if (urb->status == -ESHUTDOWN) + return; + + /* check if there is more data userspace wants to send */ + bcd2000_midi_send(bcd2k); +} + +static void bcd2000_input_complete(struct urb *urb) +{ + int ret; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status) + dev_warn(&urb->dev->dev, + PREFIX "input urb->status: %i\n", urb->status); + + if (!bcd2k || urb->status == -ESHUTDOWN) + return; + + if (urb->actual_length > 0) + bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer, + urb->actual_length); + + /* return URB to device */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() failed, ret=%d\n", + __func__, ret); +} + +static struct snd_rawmidi_ops bcd2000_midi_output = { + .open = bcd2000_midi_output_open, + .close = bcd2000_midi_output_close, + .trigger = bcd2000_midi_output_trigger, +}; + +static struct snd_rawmidi_ops bcd2000_midi_input = { + .open = bcd2000_midi_input_open, + .close = bcd2000_midi_input_close, + .trigger = bcd2000_midi_input_trigger, +}; + +static void bcd2000_init_device(struct bcd2000 *bcd2k) +{ + int ret; + + init_usb_anchor(&bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_out_urb, &bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_in_urb, &bcd2k->anchor); + + /* copy init sequence into buffer */ + memcpy(bcd2k->midi_out_buf, bcd2000_init_sequence, 52); + bcd2k->midi_out_urb->transfer_buffer_length = 52; + + /* submit sequence */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() out failed, ret=%d: ", + __func__, ret); + else + bcd2k->midi_out_active = 1; + + /* pass URB to device to enable button and controller events */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() in failed, ret=%d: ", + __func__, ret); + + /* ensure initialization is finished */ + usb_wait_anchor_empty_timeout(&bcd2k->anchor, 1000); +} + +static int bcd2000_init_midi(struct bcd2000 *bcd2k) +{ + int ret; + struct snd_rawmidi *rmidi; + + ret = snd_rawmidi_new(bcd2k->card, bcd2k->card->shortname, 0, + 1, /* output */ + 1, /* input */ + &rmidi); + + if (ret < 0) + return ret; + + strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + + rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = bcd2k; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &bcd2000_midi_output); + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &bcd2000_midi_input); + + bcd2k->rmidi = rmidi; + + bcd2k->midi_in_urb = usb_alloc_urb(0, GFP_KERNEL); + bcd2k->midi_out_urb = usb_alloc_urb(0, GFP_KERNEL); + + if (!bcd2k->midi_in_urb || !bcd2k->midi_out_urb) { + dev_err(&bcd2k->dev->dev, PREFIX "usb_alloc_urb failed\n"); + return -ENOMEM; + } + + usb_fill_int_urb(bcd2k->midi_in_urb, bcd2k->dev, + usb_rcvintpipe(bcd2k->dev, 0x81), + bcd2k->midi_in_buf, BUFSIZE, + bcd2000_input_complete, bcd2k, 1); + + usb_fill_int_urb(bcd2k->midi_out_urb, bcd2k->dev, + usb_sndintpipe(bcd2k->dev, 0x1), + bcd2k->midi_out_buf, BUFSIZE, + bcd2000_output_complete, bcd2k, 1); + + bcd2000_init_device(bcd2k); + + return 0; +} + +static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, + struct usb_interface *interface) +{ + /* usb_kill_urb not necessary, urb is aborted automatically */ + + usb_free_urb(bcd2k->midi_out_urb); + usb_free_urb(bcd2k->midi_in_urb); + + if (bcd2k->intf) { + usb_set_intfdata(bcd2k->intf, NULL); + bcd2k->intf = NULL; + } +} + +static int bcd2000_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + struct snd_card *card; + struct bcd2000 *bcd2k; + unsigned int card_index; + char usb_path[32]; + int err; + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (!test_bit(card_index, devices_used)) + break; + + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + + err = snd_card_new(&interface->dev, index[card_index], id[card_index], + THIS_MODULE, sizeof(*bcd2k), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + + bcd2k = card->private_data; + bcd2k->dev = interface_to_usbdev(interface); + bcd2k->card = card; + bcd2k->card_index = card_index; + bcd2k->intf = interface; + + snd_card_set_dev(card, &interface->dev); + + strncpy(card->driver, "snd-bcd2000", sizeof(card->driver)); + strncpy(card->shortname, "BCD2000", sizeof(card->shortname)); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + "Behringer BCD2000 at %s", + usb_path); + + err = bcd2000_init_midi(bcd2k); + if (err < 0) + goto probe_error; + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, bcd2k); + set_bit(card_index, devices_used); + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + dev_info(&bcd2k->dev->dev, PREFIX "error during probing"); + bcd2000_free_usb_related_resources(bcd2k, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void bcd2000_disconnect(struct usb_interface *interface) +{ + struct bcd2000 *bcd2k = usb_get_intfdata(interface); + + if (!bcd2k) + return; + + mutex_lock(&devices_mutex); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(bcd2k->card); + + bcd2000_free_usb_related_resources(bcd2k, interface); + + clear_bit(bcd2k->card_index, devices_used); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = "snd-bcd2000", + .probe = bcd2000_probe, + .disconnect = bcd2000_disconnect, + .id_table = id_table, +}; + +module_usb_driver(bcd2000_driver); + +MODULE_DEVICE_TABLE(usb, id_table); +MODULE_AUTHOR("Mario Kicherer, dev(a)kicherer.org"); +MODULE_DESCRIPTION("Behringer BCD2000 driver"); +MODULE_LICENSE("GPL"); -- 1.8.3.2

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[alsa-devel] Tenor TE8802, USB : clics and crackles during music play
by Eric F 07 Apr '14

07 Apr '14

Hello, Been searching for days on Internet for this issue, without luck :( I own a Teac UD-H01 : when connected via USB => crackles and clicks

4 7

[alsa-devel] [PATCH] ASoC: da732x: Print correct major id
by Axel Lin 07 Apr '14

07 Apr '14

DA732X_ID_MAJOR_MASK is 0xF0, so the major id is (reg & DA732X_ID_MAJOR_MASK) >> 4. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/da732x.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 7d168ec..48f3fef 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1571,7 +1571,8 @@ static int da732x_i2c_probe(struct i2c_client *i2c, } dev_info(&i2c->dev, "Revision: %d.%d\n", - (reg & DA732X_ID_MAJOR_MASK), (reg & DA732X_ID_MINOR_MASK)); + (reg & DA732X_ID_MAJOR_MASK) >> 4, + (reg & DA732X_ID_MINOR_MASK)); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da732x, da732x_dai, ARRAY_SIZE(da732x_dai)); -- 1.8.3.2

3 2

[alsa-devel] [PATCH alsa-lib] pcm: rate: Don't return negative frame count on success in rewind
by Andrew Eikum 07 Apr '14

07 Apr '14

snd_pcm_rewind is documented to return <0 on failure and >=0 on success. Signed-off-by: Andrew Eikum <aeikum(a)codeweavers.com> --- I checked all other rewind implementations, and this was the only one that obviously had this bug. diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 54a3e67..2eb4b1b 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -702,7 +702,7 @@ static snd_pcm_sframes_t snd_pcm_rate_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t f snd_atomic_write_begin(&rate->watom); n = snd_pcm_rate_move_applptr(pcm, -frames); snd_atomic_write_end(&rate->watom); - return n; + return n < 0 ? -n : n; } static snd_pcm_sframes_t snd_pcm_rate_forward(snd_pcm_t *pcm, snd_pcm_uframes_t frames) -- 1.9.1

3 2

[alsa-devel] [PATCH] Allow snd_soc_update_bits use 32 bits register
by bardliao＠realtek.com 07 Apr '14

07 Apr '14

From: Bard Liao <bardliao(a)realtek.com> Change reg's type from unsigned short to unsigned int. So that we can use 32 bits reg value in snd_soc_update_bits. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/soc.h | 2 +- sound/soc/soc-core.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0b83168..687dec0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -469,7 +469,7 @@ static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, #endif /* codec register bit access */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value); int snd_soc_update_bits_locked(struct snd_soc_codec *codec, unsigned short reg, unsigned int mask, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index caebd63..fde636c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2316,7 +2316,7 @@ EXPORT_SYMBOL_GPL(snd_soc_write); * * Returns 1 for change, 0 for no change, or negative error code. */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value) { bool change; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: cs42l52: Fix mask for REVID
by Axel Lin 07 Apr '14

07 Apr '14

BIT[0:2] of register 01h is REVID, so the mask for REVID should be 0x7. Also updates the code to use CS42L52_CHIP_REV_MASK. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/cs42l52.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index f0ca6be..460d355 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1259,7 +1259,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, } dev_info(&i2c_client->dev, "Cirrus Logic CS42L52, Revision: %02X\n", - reg & 0xFF); + reg & CS42L52_CHIP_REV_MASK); /* Set Platform Data */ if (cs42l52->pdata.mica_diff_cfg) diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 6fb8f00..ac445993 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -37,7 +37,7 @@ #define CS42L52_CHIP_REV_A0 0x00 #define CS42L52_CHIP_REV_A1 0x01 #define CS42L52_CHIP_REV_B0 0x02 -#define CS42L52_CHIP_REV_MASK 0x03 +#define CS42L52_CHIP_REV_MASK 0x07 #define CS42L52_PWRCTL1 0x02 #define CS42L52_PWRCTL1_PDN_ALL 0x9F -- 1.8.3.2

3 2