July 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH 1/2] ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
by Kai Vehmanen 07 Jul '20

07 Jul '20

When HDMI PCM devices are opened in a specific order, with at least one HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the connected monitor, on recent Intel platforms (ICL/JSL and newer). While this is not a typical sequence, at least Pulseaudio does this every time when it is started, to discover the available PCMs. The rootcause is an invalid assumption in hdmi_add_pin(), where the total number of converters is assumed to be known at the time the function is called. On older Intel platforms this held true, but after ICL/JSL, the order how pins and converters are in the subnode list as returned by snd_hda_get_sub_nodes(), was changed. As a result, information for some converters was not stored to per_pin->mux_nids. And this means some pins cannot be connected to all converters, and application instead gets -EBUSY instead at open. The assumption that converters are always before pins in the subnode list, is not really a valid one. Fix the problem in hdmi_parse_codec() by introducing separate loops for discovering converters and pins. BugLink: https://github.com/thesofproject/linux/issues/1978 BugLink: https://github.com/thesofproject/linux/issues/2216 BugLink: https://github.com/thesofproject/linux/issues/2217 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e2b21ef5d7d1..295cc5a989d5 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1804,33 +1804,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static int hdmi_parse_codec(struct hda_codec *codec) { - hda_nid_t nid; + hda_nid_t start_nid; + unsigned int caps; int i, nodes; - nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid); - if (!nid || nodes < 0) { + nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); + if (!start_nid || nodes < 0) { codec_warn(codec, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; + /* + * hdmi_add_pin() assumes total amount of converters to + * be known, so first discover all converters + */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; caps = get_wcaps(codec, nid); - type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL)) continue; - switch (type) { - case AC_WID_AUD_OUT: + if (get_wcaps_type(caps) == AC_WID_AUD_OUT) hdmi_add_cvt(codec, nid); - break; - case AC_WID_PIN: + } + + /* discover audio pins */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; + + caps = get_wcaps(codec, nid); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + if (get_wcaps_type(caps) == AC_WID_PIN) hdmi_add_pin(codec, nid); - break; - } } return 0; -- 2.27.0

2 3

[PATCH 1/1] ALSA: opl3: fix infoleak in opl3
by Xidong Wang 07 Jul '20

07 Jul '20

From: xidongwang <wangxidong_97(a)163.com> The stack object “info” in snd_opl3_ioctl() has a leaking problem. It has 2 padding bytes which are not initialized and leaked via “copy_to_user”. Signed-off-by: xidongwang <wangxidong_97(a)163.com> --- sound/drivers/opl3/opl3_synth.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index e69a4ef..08c10ac 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -91,6 +91,8 @@ int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file, { struct snd_dm_fm_info info; + memset(&info, 0, sizeof(info)); + info.fm_mode = opl3->fm_mode; info.rhythm = opl3->rhythm; if (copy_to_user(argp, &info, sizeof(struct snd_dm_fm_info))) -- 2.7.4

2 1

[RFC PATCH] ALSA: compress: add support to change codec profile in gapless playback
by Srinivas Kandagatla 06 Jul '20

06 Jul '20

For gapless playback its possible that each track can have different codec profile with same decoder, for example we have WMA album, we may have different tracks as WMA v9, WMA v10 and so on Existing code does not allow to change this profile while doing gapless playback. This patch adds new SNDRV_COMPRESS_SET_CODEC_PARAMS IOCTL to allow userspace to set this new parameters required for new codec profile. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- include/sound/compress_driver.h | 5 +++ include/sound/soc-component.h | 3 ++ include/sound/soc-dai.h | 5 +++ include/uapi/sound/compress_offload.h | 1 + sound/core/compress_offload.c | 54 ++++++++++++++++++++++++--- sound/soc/soc-compress.c | 30 +++++++++++++++ sound/soc/soc-dai.c | 14 +++++++ 7 files changed, 106 insertions(+), 6 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 70cbc5095e72..8d23351f7ad7 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -93,6 +93,9 @@ struct snd_compr_stream { * @set_params: Sets the compressed stream parameters, mandatory * This can be called in during stream creation only to set codec params * and the stream properties + * @set_codec_params: Sets the compressed stream codec parameters, mandatory + * This can be called in during gapless next track codec change only to set + * codec params * @get_params: retrieve the codec parameters, mandatory * @set_metadata: Set the metadata values for a stream * @get_metadata: retrieves the requested metadata values from stream @@ -112,6 +115,8 @@ struct snd_compr_ops { int (*free)(struct snd_compr_stream *stream); int (*set_params)(struct snd_compr_stream *stream, struct snd_compr_params *params); + int (*set_codec_params)(struct snd_compr_stream *stream, + struct snd_codec *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); int (*set_metadata)(struct snd_compr_stream *stream, diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 5663891148e3..1e69c54ed0b9 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -36,6 +36,9 @@ struct snd_compress_ops { int (*get_params)(struct snd_soc_component *component, struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_codec_params)(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct snd_codec *params); int (*set_metadata)(struct snd_soc_component *component, struct snd_compr_stream *stream, struct snd_compr_metadata *metadata); diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 212257e84fac..526794ee555b 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -192,6 +192,9 @@ int snd_soc_dai_compr_trigger(struct snd_soc_dai *dai, int snd_soc_dai_compr_set_params(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, struct snd_compr_params *params); +int snd_soc_dai_compr_set_codec_params(struct snd_soc_dai *dai, + struct snd_compr_stream *cstream, + struct snd_codec *codec); int snd_soc_dai_compr_get_params(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, struct snd_codec *params); @@ -292,6 +295,8 @@ struct snd_soc_cdai_ops { struct snd_soc_dai *); int (*set_params)(struct snd_compr_stream *, struct snd_compr_params *, struct snd_soc_dai *); + int (*set_codec_params)(struct snd_compr_stream *, + struct snd_codec *, struct snd_soc_dai *); int (*get_params)(struct snd_compr_stream *, struct snd_codec *, struct snd_soc_dai *); int (*set_metadata)(struct snd_compr_stream *, diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 7184265c0b0d..c46286113a4b 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -172,6 +172,7 @@ struct snd_compr_metadata { struct snd_compr_metadata) #define SNDRV_COMPRESS_GET_METADATA _IOWR('C', 0x15,\ struct snd_compr_metadata) +#define SNDRV_COMPRESS_SET_CODEC_PARAMS _IOW('C', 0x16, struct snd_codec) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 0e53f6f31916..1c4b2cf450a0 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -566,6 +566,18 @@ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, return 0; } +static int snd_compress_check_codec_params(struct snd_codec *codec) +{ + /* now codec parameters */ + if (codec->id == 0 || codec->id > SND_AUDIOCODEC_MAX) + return -EINVAL; + + if (codec->ch_in == 0 || codec->ch_out == 0) + return -EINVAL; + + return 0; +} + static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ @@ -574,14 +586,41 @@ static int snd_compress_check_input(struct snd_compr_params *params) params->buffer.fragments == 0) return -EINVAL; - /* now codec parameters */ - if (params->codec.id == 0 || params->codec.id > SND_AUDIOCODEC_MAX) - return -EINVAL; + return snd_compress_check_codec_params(&params->codec); - if (params->codec.ch_in == 0 || params->codec.ch_out == 0) - return -EINVAL; +} - return 0; +static int snd_compr_set_codec_params(struct snd_compr_stream *stream, + unsigned long arg) +{ + struct snd_codec *params; + int retval; + + if (!stream->ops->set_codec_params) + return -EPERM; + + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + + /* codec params can be only set when next track has been signalled */ + if (stream->next_track == false) + return -EPERM; + + params = memdup_user((void __user *)arg, sizeof(*params)); + if (IS_ERR(params)) + return PTR_ERR(params); + + retval = snd_compress_check_codec_params(params); + if (retval) + goto out; + + retval = stream->ops->set_codec_params(stream, params); + if (retval) + goto out; + +out: + kfree(params); + return retval; } static int @@ -964,6 +1003,9 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_SET_PARAMS): retval = snd_compr_set_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_CODEC_PARAMS): + retval = snd_compr_set_codec_params(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 4984b6a2c370..e549e0197aca 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -518,6 +518,34 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, return ret; } +static int soc_compr_set_codec_params(struct snd_compr_stream *cstream, + struct snd_codec *codec) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_component *component; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + int i, ret; + + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + + ret = snd_soc_dai_compr_set_codec_params(cpu_dai, cstream, codec); + if (ret < 0) + goto err; + + for_each_rtd_components(rtd, i, component) { + if (!component->driver->compress_ops || + !component->driver->compress_ops->set_codec_params) + continue; + + ret = component->driver->compress_ops->set_codec_params(component, cstream, + codec); + break; + } +err: + mutex_unlock(&rtd->card->pcm_mutex); + return ret; +} + static int soc_compr_get_params(struct snd_compr_stream *cstream, struct snd_codec *params) { @@ -728,6 +756,7 @@ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, .free = soc_compr_free, .set_params = soc_compr_set_params, + .set_codec_params = soc_compr_set_codec_params, .set_metadata = soc_compr_set_metadata, .get_metadata = soc_compr_get_metadata, .get_params = soc_compr_get_params, @@ -744,6 +773,7 @@ static struct snd_compr_ops soc_compr_dyn_ops = { .free = soc_compr_free_fe, .set_params = soc_compr_set_params_fe, .get_params = soc_compr_get_params, + .set_codec_params = soc_compr_set_codec_params, .set_metadata = soc_compr_set_metadata, .get_metadata = soc_compr_get_metadata, .trigger = soc_compr_trigger_fe, diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index b05e18b63a1c..06481d0278b8 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -594,6 +594,20 @@ int snd_soc_dai_compr_get_params(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(snd_soc_dai_compr_get_params); +int snd_soc_dai_compr_set_codec_params(struct snd_soc_dai *dai, + struct snd_compr_stream *cstream, + struct snd_codec *codec) +{ int ret = 0; + + if (dai->driver->cops && + dai->driver->cops->set_codec_params) + ret = dai->driver->cops->set_codec_params(cstream, codec, dai); + + return soc_dai_ret(dai, ret); + +} +EXPORT_SYMBOL_GPL(snd_soc_dai_compr_set_codec_params); + int snd_soc_dai_compr_ack(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, size_t bytes) -- 2.21.0

3 5

[PATCH] ASoC: codecs: wm8400: remove unused variables
by Pierre-Louis Bossart 06 Jul '20

06 Jul '20

Fix W=1 warning by removing unused variables Suggested-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/wm8400.c | 12 ------------ 1 file changed, 12 deletions(-) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 2551eb0f1868..35ffa7765c85 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -67,16 +67,12 @@ static void wm8400_component_reset(struct snd_soc_component *component) wm8400_reset_codec_reg_cache(wm8400->wm8400); } -static const DECLARE_TLV_DB_SCALE(rec_mix_tlv, -1500, 600, 0); - static const DECLARE_TLV_DB_SCALE(in_pga_tlv, -1650, 3000, 0); static const DECLARE_TLV_DB_SCALE(out_mix_tlv, -2100, 0, 0); static const DECLARE_TLV_DB_SCALE(out_pga_tlv, -7300, 600, 0); -static const DECLARE_TLV_DB_SCALE(out_omix_tlv, -600, 0, 0); - static const DECLARE_TLV_DB_SCALE(out_dac_tlv, -7163, 0, 0); static const DECLARE_TLV_DB_SCALE(in_adc_tlv, -7163, 1763, 0); @@ -439,14 +435,6 @@ static SOC_ENUM_SINGLE_DECL(wm8400_ainrmux_enum, static const struct snd_kcontrol_new wm8400_dapm_ainrmux_controls = SOC_DAPM_ENUM("Route", wm8400_ainrmux_enum); -/* RXVOICE */ -static const struct snd_kcontrol_new wm8400_dapm_rxvoice_controls[] = { -SOC_DAPM_SINGLE_TLV("LIN4/RXN", WM8400_INPUT_MIXER5, WM8400_LR4BVOL_SHIFT, - WM8400_LR4BVOL_MASK, 0, in_mix_tlv), -SOC_DAPM_SINGLE_TLV("RIN4/RXP", WM8400_INPUT_MIXER6, WM8400_RL4BVOL_SHIFT, - WM8400_RL4BVOL_MASK, 0, in_mix_tlv), -}; - /* LOMIX */ static const struct snd_kcontrol_new wm8400_dapm_lomix_controls[] = { SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8400_OUTPUT_MIXER1, -- 2.25.1

2 1

Re: [PATCH v2 1/7] Documentation: device-tree: sound: Update lpass-cpu driver binding
by Rohit Kumar 06 Jul '20

06 Jul '20

On 5/16/2020 11:07 PM, Ajit Pandey wrote: > On 5/14/2020 10:14 PM, Mark Brown wrote: >> On Thu, May 14, 2020 at 10:08:12PM +0530, Ajit Pandey wrote: >>> Done the required cleanups to update lpass-cpu binding with newer >>> yaml formats. >> Please do YAML conversions as the last thing in any patch series, there >> is quite a big backlog of reviews of YAML conversions so they can slow >> down the code development. It's good to do them, just try to make sure >> other patches don't depend on them. >> >> Please submit patches using subject lines reflecting the style for the >> subsystem, this makes it easier for people to identify relevant patches. >> Look at what existing commits in the area you're changing are doing and >> make sure your subject lines visually resemble what they're doing. >> There's no need to resubmit to fix this alone. > > Ok.. we'll do the yaml conversion as a separate patch chain so that it > won't block > > merging of this feature in ASoC core. @Mark, I was planning to post v3 patchset series for this and keep yaml change as last patch in the series as suggested. However, we have one update in the driver to support another compatible string. If we keep Documentation change at the last, the driver change will have checkpatch errors. Can you please suggest if should make first Documentation change in text file itself before driver change and finally have a patch to convert it to yaml at the end? Thanks, Rohit -- Qualcomm INDIA, on behalf of Qualcomm Innovation Center, Inc.is a member of the Code Aurora Forum, hosted by the Linux Foundation.

2 1

[PATCH v2] ASoC: cros_ec_codec: Log results when EC commands fail
by Yu-Hsuan Hsu 06 Jul '20

06 Jul '20

Log results of failed EC commands to identify a problem more easily. Replace cros_ec_cmd_xfer_status with cros_ec_cmd_xfer because the result has already been checked in this function. The wrapper is not needed. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/codecs/cros_ec_codec.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 8d45c628e988e..a4ab62f59efa6 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -90,10 +90,17 @@ static int send_ec_host_command(struct cros_ec_device *ec_dev, uint32_t cmd, if (outsize) memcpy(msg->data, out, outsize); - ret = cros_ec_cmd_xfer_status(ec_dev, msg); + ret = cros_ec_cmd_xfer(ec_dev, msg); if (ret < 0) goto error; + if (msg->result != EC_RES_SUCCESS) { + dev_err(ec_dev->dev, "Command %d failed: %d\n", cmd, + msg->result); + ret = -EPROTO; + goto error; + } + if (insize) memcpy(in, msg->data, insize); -- 2.27.0.212.ge8ba1cc988-goog

4 12

Move ownership of snd-firewire-ctl-services repository
by Takashi Sakamoto 05 Jul '20

05 Jul '20

Hi Jaroslav, I'm currently working for the idea of 'ALSA control service in userspace'[1] for audio and music units on IEEE 1394 bus under my personal repository[2], and the work mostly done for Digidesign Digi 002/003 family and Tascam FireWire series[3]. As well as libhinawa and alsa-gobject repositories[4], I'd like to move ownership of the repository to 'GObject Introspection' team of alsa-project account, then continue the work. I'd like you to arrange access rights and so on, again. Besides, as I tols you in previous message[5], the nested team is not good in reasons. I strongly request you to leave 'GObject Introspection' team and drop 'Hinawa' team, again. Would I expect your arrangement for it? [1] https://mailman.alsa-project.org/pipermail/alsa-devel/2020-June/169141.html [2] https://github.com/takaswie/snd-firewire-ctl-services/ [3] https://github.com/takaswie/snd-firewire-ctl-services/tree/topic/services-f… [4] https://mailman.alsa-project.org/pipermail/alsa-devel/2020-June/169141.html [5] https://mailman.alsa-project.org/pipermail/alsa-devel/2020-June/169191.html Thanks Takashi Sakamoto

1 1

Some questions about asoc codec mixer elements naming
by Станислав Александров 05 Jul '20

05 Jul '20

Hello. I am writing an asoc codec driver and stuck with mixer and output naming. Could someone clarify a few things? This codec has two playback PGAs and 4 outputs: PGA1 -> [Speaker,Headphone,Line Out] PGA2 -> [Mono Output] [PGA1] controls volume and mute for all three outputs. [Speaker,Headphone,Line Out] have only mute. [PGA2] controls volume and mute. [Mono Output] also has mute. Codec is bound to platform driver via simple-audio-card: simple-audio-card,widgets = "Headphone", "Headphones", "Speaker", "Speakers", "Line", "Line Out Jack"; simple-audio-card,routing = "Headphones", "HPOUT", "Speakers", "SPKOUT", "Line Out Jack", "LINEOUT"; Is it possible to name mixer controls so that no further configuration is required when the driver is loaded? i.e. pulseaudio automatically detects all these outputs and controls, correctly maps them and allows switching between them? Currently i have this configuration(simplified): SOC_DOUBLE_S8_TLV("Master Playback Volume", PGA1, ...) SOC_DOUBLE("Master Playback Switch", PGA1, ...) SOC_SINGLE("Speaker Playback Switch", ...) SOC_SINGLE("Headphone Playback Switch", ...) SOC_SINGLE("Line Out Playback Switch", ...) SOC_SINGLE_S8_TLV("Mono Output Playback Volume", PGA2...) SOC_SINGLE("Mono Output Playback Switch", PGA2....) With these settings pulseaudio only detects Speaker and Headphone (but not Line Out because it wants to find its Jack) But as I understand "Master Playback" assumes control for all sound streams, so this is probably not correct =\ If it is impossible to name those elements that way so everything is autodetected, what should i use then? Write a machine audio driver or use alsaucm?

1 0

Asus Xonar SE
by Jefferson Smith Pizarro Gutierrez 05 Jul '20

05 Jul '20

hi, i have issues with my Asus Xonar SE souncard, i need to get a solution, please helpme to get sound through the spdif of the soundcard. i dont have sound

1 1

Looking for guidance for porting home made audio driver to ASoC/FSL
by Christophe Leroy 04 Jul '20

04 Jul '20

Hello I'm looking for guidance for porting a home made audio driver to ASoC/FSL on a custom board. The board has an MPC8321 freescale/NXP cpu. The board has a TDM bus connected to one of the cpu TDM port. On that TDM bus, there are three IDT 821034 quad codecs (PCM G.711) and an E1 chip that carries IP over E1. Today, there is a home made audio driver providing raw read/write devices plus an home made WAN HDLC driver for the E1. The CPU Quicc Engine (QE) is configured to connect the TDM port to one UCC. QUICC Multi-Channel Controller (QMC) is used to distribute E1 timeslots to the HDLC driver and each of the 12 codec timeslots to the audio driver. (The QMC is able to manage two types of channels: transparent and HDLC). We would like to make all this more standard and port it to use ALSA ASoC for the audio part, and the Freescale Quicc Engine HDLC. As of today, the Freescale QE HDLC wan driver uses the UCC directly, it doesn't interface through the QMC. And the FSL ASoC drivers seem to neither handle QE TDM via QE QMC nor QE UCC. We are thinking about adding a QE QMC layer in driver/soc/fsl/ which would more or less present to the consumer the same kind of API as the FSL UCC layer, and enhance the WAN FSL HDLC driver to be able to use that new layer. For the ALSA ASoC part, could some parts of existing FSL ASoC drivers be re-used of would we have to start implementing an ASoC from scratch ? Any recommendation and/or guidance would be welcomed and appreciated. Thanks in advance Christophe

1 0