September 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: line6: make snd_pcm_ops const
by Bhumika Goyal 02 Oct '17

02 Oct '17

Make these const as they are only passed to a const argument of the function snd_pcm_set_ops in the file referencing them. Also, add const to the declaration in the headers. Structures found using Coccinelle and changes done by hand. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/usb/line6/capture.c | 2 +- sound/usb/line6/capture.h | 2 +- sound/usb/line6/playback.c | 2 +- sound/usb/line6/playback.h | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/line6/capture.c b/sound/usb/line6/capture.c index 7c81256..947d616 100644 --- a/sound/usb/line6/capture.c +++ b/sound/usb/line6/capture.c @@ -248,7 +248,7 @@ static int snd_line6_capture_close(struct snd_pcm_substream *substream) } /* capture operators */ -struct snd_pcm_ops snd_line6_capture_ops = { +const struct snd_pcm_ops snd_line6_capture_ops = { .open = snd_line6_capture_open, .close = snd_line6_capture_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/usb/line6/capture.h b/sound/usb/line6/capture.h index 890b21b..b67ccc3 100644 --- a/sound/usb/line6/capture.h +++ b/sound/usb/line6/capture.h @@ -17,7 +17,7 @@ #include "driver.h" #include "pcm.h" -extern struct snd_pcm_ops snd_line6_capture_ops; +extern const struct snd_pcm_ops snd_line6_capture_ops; extern void line6_capture_copy(struct snd_line6_pcm *line6pcm, char *fbuf, int fsize); diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c index 812d181..819e9b2 100644 --- a/sound/usb/line6/playback.c +++ b/sound/usb/line6/playback.c @@ -393,7 +393,7 @@ static int snd_line6_playback_close(struct snd_pcm_substream *substream) } /* playback operators */ -struct snd_pcm_ops snd_line6_playback_ops = { +const struct snd_pcm_ops snd_line6_playback_ops = { .open = snd_line6_playback_open, .close = snd_line6_playback_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/usb/line6/playback.h b/sound/usb/line6/playback.h index 51fce29..d8d3b8a 100644 --- a/sound/usb/line6/playback.h +++ b/sound/usb/line6/playback.h @@ -27,7 +27,7 @@ */ #define USE_CLEAR_BUFFER_WORKAROUND 1 -extern struct snd_pcm_ops snd_line6_playback_ops; +extern const struct snd_pcm_ops snd_line6_playback_ops; extern int line6_create_audio_out_urbs(struct snd_line6_pcm *line6pcm); extern int line6_submit_audio_out_all_urbs(struct snd_line6_pcm *line6pcm); -- 1.9.1

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[alsa-devel] [PATCH] Revert "ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members"
by Takashi Sakamoto 02 Oct '17

02 Oct '17

This reverts commit 275353bb684e to fix a regression which can abort 'alsactl' program in alsa-utils due to assertion in alsa-lib. alsactl: control.c:2513: snd_ctl_elem_value_get_integer: Assertion `idx < sizeof(obj->value.integer.value) / sizeof(obj->value.integer.value[0])' failed. alsactl: control.c:2976: snd_ctl_elem_value_get_integer: Assertion `idx < ARRAY_SIZE(obj->value.integer.value)' failed. This commit is a band-aid. In a point of usage of ALSA control interface, the drivers still bring an issue that they prevent userspace applications to have a consistent way to parse each levels of the dimension information via ALSA control interface. Let me investigate this issue. Current implementation of the drivers have three control element sets with dimension information: * 'Monitor Mixer Volume' (type: integer) * 'VMixer Volume' (type: integer) * 'VU-meters' (type: boolean) Although the number of elements named as 'Monitor Mixer Volume' differs depending on drivers in this group, it can be calculated by macros defined by each driver (= (BX_NUM - BX_ANALOG_IN) * BX_ANALOG_IN). Each of the elements has one member for value and has dimension information with 2 levels (= BX_ANALOG_IN * (BX_NUM - BX_ANALOG_IN)). For these elements, userspace applications are expected to handle the dimension information so that all of the elements construct a matrix where the number of rows and columns are represented by the dimension information. The same way is applied to elements named as 'VMixer Volume'. The number of these elements can also be calculated by macros defined by each drivers (= PX_ANALOG_IN * BX_ANALOG_IN). Each of the element has one member for value and has dimension information with 2 levels (= BX_ANALOG_IN * PX_ANALOG_IN). All of the elements construct a matrix with the dimension information. An element named as 'VU-meters' gets a different way in a point of dimension information. The element includes 96 members for value. The element has dimension information with 3 levels (= 3 or 2 * 16 * 2). For this element, userspace applications are expected to handle the dimension information so that all of the members for value construct a matrix where the number of rows and columns are represented by the dimension information. This is different from the way for the former. As a summary, the drivers were not designed to produce a consistent way to parse the dimension information. This makes it hard for general userspace applications such as amixer to parse the information by a consistent way, and actually no userspace applications except for 'echomixer' utilize the dimension information. Additionally, no drivers excluding this group use the information. The reverted commit was written based on the latter way. A commit 860c1994a70a ('ALSA: control: add dimension validator for userspace elements') is written based on the latter way, too. The patch should be reconsider too in the same time to re-define a consistent way to parse the dimension information. Reported-by: Mark Hills <mark(a)xwax.org> Reported-by: S. Christian Collins <s.chriscollins(a)gmail.com> Fixes: 275353bb684e ('ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members') Cc: <stable(a)vger.kernel.org> # v4.8+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/pci/echoaudio/echoaudio.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 7326695bca33..d68f99e076a8 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1272,11 +1272,11 @@ static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol, chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = ECHOGAIN_MAXOUT; uinfo->dimen.d[0] = num_busses_out(chip); uinfo->dimen.d[1] = num_busses_in(chip); - uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1]; return 0; } @@ -1344,11 +1344,11 @@ static int snd_echo_vmixer_info(struct snd_kcontrol *kcontrol, chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = ECHOGAIN_MAXOUT; uinfo->dimen.d[0] = num_busses_out(chip); uinfo->dimen.d[1] = num_pipes_out(chip); - uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1]; return 0; } @@ -1728,6 +1728,7 @@ static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 96; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = 0; #ifdef ECHOCARD_HAS_VMIXER @@ -1737,7 +1738,6 @@ static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol, #endif uinfo->dimen.d[1] = 16; /* 16 channels */ uinfo->dimen.d[2] = 2; /* 0=level, 1=peak */ - uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1] * uinfo->dimen.d[2]; return 0; } -- 2.11.0

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[alsa-devel] [PATCH v1 0/2] Fix first detection issue with headset
by Damien Riegel 02 Oct '17

02 Oct '17

This series fix an issue with headset being misidentified on the very first time a device is plugged in by calling the function that configures the bias when the MBHC is setup. I've split this series into two patches because pm8916_wcd_setup_mbhc needs to be moved a few lines below to avoid forward declaring it. It is overkill but that was not part of the code snippet that Srinivas acked, and I wanted to be extra sure not to misattribute an Acked-by. Damien Riegel (2): ASoC: codecs: msm8916-wcd-analog: move pm8916_wcd_setup_mbhc ASoC: codecs: msm8916-wcd-analog: configure micbias in mbhc setup sound/soc/codecs/msm8916-wcd-analog.c | 94 +++++++++++++++++++---------------- 1 file changed, 50 insertions(+), 44 deletions(-) -- 2.14.1

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[alsa-devel] [PATCH v2] ALSA: hda - program ICT bits to support HBR audio
by Pierre-Louis Bossart 02 Oct '17

02 Oct '17

From: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> On recent Intel platforms (Haswell, Broadwell, Skylake, ApolloLake, KabyLake, ...), the IEC Coding Type (ICT) bitfield in the Digital Converter Control #3 needs to be set explicitly for HDMI/DisplayPort High Bit Rate (HBR) audio playback to work. This was not required in earlier platforms when HBR was first introduced. The ICT bits are defined in Section 7.3.3.9 of the HDaudio 1.0a specification. Since the ICT bitfield was not specified for HDAudio 1.0 devices (before 2009), we only program it on machines more recent than Haswell. We tested that this fix is not needed on Baytrail-I (MinnowBoard Turbot) and believe by extension it also does not apply to Braswell. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=98797 Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- v2: move digi3 verb in Set section, only set ICT for haswell and newer include/sound/hda_verbs.h | 1 + sound/pci/hda/patch_hdmi.c | 21 +++++++++++++++++++++ 2 files changed, 22 insertions(+) diff --git a/include/sound/hda_verbs.h b/include/sound/hda_verbs.h index d0509db6d0ec..d2004d7feb2b 100644 --- a/include/sound/hda_verbs.h +++ b/include/sound/hda_verbs.h @@ -54,6 +54,7 @@ enum { #define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c #define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d #define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e /* unused */ +#define AC_VERB_SET_DIGI_CONVERT_3 0x73e #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f /* f10-f1a: GPIO */ #define AC_VERB_GET_GPIO_DATA 0x0f15 diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 53f9311370de..8566de298ea3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -906,6 +906,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, hda_nid_t pin_nid, u32 stream_tag, int format) { struct hdmi_spec *spec = codec->spec; + unsigned int param; int err; err = spec->ops.pin_hbr_setup(codec, pin_nid, is_hbr_format(format)); @@ -915,6 +916,26 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, return err; } + if (is_haswell_plus(codec)) { + + /* + * on recent platforms IEC Coding Type is required for HBR + * support, read current Digital Converter settings and set + * ICT bitfield if needed. + */ + param = snd_hda_codec_read(codec, cvt_nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); + + param = (param >> 16) & ~(AC_DIG3_ICT); + + /* on recent platforms ICT mode is required for HBR support */ + if (is_hbr_format(format)) + param |= 0x1; + + snd_hda_codec_write(codec, cvt_nid, 0, + AC_VERB_SET_DIGI_CONVERT_3, param); + } + snd_hda_codec_setup_stream(codec, cvt_nid, stream_tag, 0, format); return 0; } -- 2.11.0

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[alsa-devel] [PATCH 0/2] ASoC: soc-core: add missing component functions
by Kuninori Morimoto 02 Oct '17

02 Oct '17

Hi Mark These are resend of one of "replace to component" patch set. These add missing functions which exist in "platforms" but not exist in "component". Kuninori Morimoto (2): ASoC: soc-core: add component lookup functions ASoC: soc-core: add snd_soc_add_component() include/sound/soc.h | 7 ++++++ sound/soc/soc-core.c | 60 ++++++++++++++++++++++++++++++++++++++++++---------- 2 files changed, 56 insertions(+), 11 deletions(-) -- 1.9.1

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[alsa-devel] 4.14-rc2: audio stopped working?
by Pavel Machek 29 Sep '17

29 Sep '17

Hi! I've had sound problems in chromium for long long time. It produces choppy auduo just after boot and then it works, probably after chromium restart. May have something to do with pulseaudio. Not sure. Anyway... sound in flightgear usually worked, and now it does not. mplayer works, chromium works (sometimes choppy, as usual). Any ideas? Pavel -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

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[alsa-devel] [RFC] ASoC: codecs: msm8916-wcd-analog: use btn0 released detection
by Damien Riegel 28 Sep '17

28 Sep '17

msm8916-wcd-analog uses button0 to differentiate between headphone and headset. Under some circumstances, button pressed and released interrupts are not fired as the driver expects it. For instance, with some connectors, there are spurious button-pressed interrupts when unplugging a headphone, without the corresponding button-released interrupt. But the codec always alternates between button pressed and released interrupts, it cannot fire two interrupts of the same kind in a row. That means that when the headphone is plugged back, only a button-released interrupt will be fired instead of pressed then released. This causes the driver to report headphone as headset. By changing the logic and relying on button 0 release interrupt, the driver could be made more robust for connectors that differ from the one used on the Dragonboard's audio mezzanine. Signed-off-by: Damien Riegel <damien.riegel(a)savoirfairelinux.com> --- sound/soc/codecs/msm8916-wcd-analog.c | 18 ++++++++---------- 1 file changed, 8 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 549c269acc7d..f562f2d86907 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -285,7 +285,7 @@ struct pm8916_wcd_analog_priv { u16 codec_version; bool mbhc_btn_enabled; /* special event to detect accessory type */ - bool mbhc_btn0_pressed; + int mbhc_btn0_released; bool detect_accessory_type; struct clk *mclk; struct snd_soc_codec *codec; @@ -483,7 +483,7 @@ static void pm8916_wcd_setup_mbhc(struct pm8916_wcd_analog_priv *wcd) snd_soc_update_bits(codec, CDC_D_INT_EN_CLR, int_en_mask, 0); snd_soc_update_bits(codec, CDC_D_INT_EN_SET, int_en_mask, int_en_mask); - wcd->mbhc_btn0_pressed = false; + wcd->mbhc_btn0_released = false; wcd->detect_accessory_type = true; } @@ -950,7 +950,7 @@ static irqreturn_t mbhc_btn_release_irq_handler(int irq, void *arg) /* check if its BTN0 thats released */ if ((val != -1) && !(val & CDC_A_MBHC_RESULT_1_BTN_RESULT_MASK)) - priv->mbhc_btn0_pressed = false; + priv->mbhc_btn0_released = true; } else { snd_soc_jack_report(priv->jack, 0, btn_mask); @@ -983,9 +983,7 @@ static irqreturn_t mbhc_btn_press_irq_handler(int irq, void *arg) break; case 0x0: /* handle BTN_0 specially for type detection */ - if (priv->detect_accessory_type) - priv->mbhc_btn0_pressed = true; - else + if (!priv->detect_accessory_type) snd_soc_jack_report(priv->jack, SND_JACK_BTN_0, btn_mask); break; @@ -1029,19 +1027,19 @@ static irqreturn_t pm8916_mbhc_switch_irq_handler(int irq, void *arg) * both press and release event received then its * a headset. */ - if (priv->mbhc_btn0_pressed) + if (priv->mbhc_btn0_released) snd_soc_jack_report(priv->jack, - SND_JACK_HEADPHONE, hs_jack_mask); + SND_JACK_HEADSET, hs_jack_mask); else snd_soc_jack_report(priv->jack, - SND_JACK_HEADSET, hs_jack_mask); + SND_JACK_HEADPHONE, hs_jack_mask); priv->detect_accessory_type = false; } else { /* removal */ snd_soc_jack_report(priv->jack, 0, hs_jack_mask); priv->detect_accessory_type = true; - priv->mbhc_btn0_pressed = false; + priv->mbhc_btn0_released = false; } return IRQ_HANDLED; -- 2.14.1

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix jack name format substitution
by Chintan Patel 28 Sep '17

28 Sep '17

Jack name is not getting formatted correctly hence resulting in invalid name for HDMI/DP input devices. This was recently exposed due changes brought by MST: commit 3a13347f05fd ("ASoC: Intel: kbl: Add jack port initialize in kbl machine drivers") Signed-off-by: Chintan Patel <chintan.m.patel(a)intel.com> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 4689991bcc86..c127b7b0ee3f 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -611,6 +611,8 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { codec = pcm->codec_dai->codec; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &ctx->kabylake_hdmi[i], NULL, 0); -- 1.9.1

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[alsa-devel] [PATCH v3] ASoC: rt5670: add set_bclk_ratio in dai ops
by Bard Liao 28 Sep '17

28 Sep '17

We need to set a specific bit for 50 bclk rate. So add set_bclk_ratio function to set the bit. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- v2: * add define for tdm data format select bit. v3: * change if (ratio == 50) to if ((ratio % 50) == 0) since the bit needs to be set if bclk ratio is divisible by 50. --- sound/soc/codecs/rt5670.c | 19 +++++++++++++++++++ sound/soc/codecs/rt5670.h | 4 ++++ 2 files changed, 23 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 9545764..78517d8 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2582,6 +2582,24 @@ static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, return 0; } +static int rt5670_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (dai->id != RT5670_AIF1) + return 0; + + if ((ratio % 50) == 0) + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, + RT5670_TDM_DATA_MODE_SEL, RT5670_TDM_DATA_MODE_50FS); + else + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, + RT5670_TDM_DATA_MODE_SEL, RT5670_TDM_DATA_MODE_NOR); + + return 0; +} + static int rt5670_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -2712,6 +2730,7 @@ static const struct snd_soc_dai_ops rt5670_aif_dai_ops = { .set_fmt = rt5670_set_dai_fmt, .set_tdm_slot = rt5670_set_tdm_slot, .set_pll = rt5670_set_dai_pll, + .set_bclk_ratio = rt5670_set_bclk_ratio, }; static struct snd_soc_dai_driver rt5670_dai[] = { diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 5ba485c..265df80 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1816,6 +1816,10 @@ #define RT5670_ZCD_HP_DIS (0x0 << 15) #define RT5670_ZCD_HP_EN (0x1 << 15) +/* General Control 3 (0xfc) */ +#define RT5670_TDM_DATA_MODE_SEL (0x1 << 11) +#define RT5670_TDM_DATA_MODE_NOR (0x0 << 11) +#define RT5670_TDM_DATA_MODE_50FS (0x1 << 11) /* Codec Private Register definition */ /* 3D Speaker Control (0x63) */ -- 2.7.4

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[alsa-devel] [PATCH v2] ASoC: dapm: Avoid creating kcontrol for params
by yesanishhere＠gmail.com 28 Sep '17

28 Sep '17

From: anish kumar <yesanishhere(a)gmail.com> Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- Changes in this version: - Review comments from charles to move common function of error handling in seperate function. Changes in v1: - Included the review comments from charles regarding releasing memory in error path. sound/soc/soc-dapm.c | 145 +++++++++++++++++++++++++++++++-------------------- 1 file changed, 89 insertions(+), 56 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d55cac6..5dae9d8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3778,18 +3778,27 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, return 0; } -int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - const struct snd_soc_pcm_stream *params, - unsigned int num_params, - struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) +void +snd_soc_dapm_free_kcontrol(struct snd_soc_card *card, + unsigned long *private_value, + int num_params, + const char **w_param_text) +{ + int count; + + devm_kfree(card->dev, (void *)*private_value); + for (count = 0 ; count < num_params; count++) + devm_kfree(card->dev, (void *)w_param_text[count]); + devm_kfree(card->dev, w_param_text); +} + +static struct snd_kcontrol_new * +snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card, + char *link_name, + const struct snd_soc_pcm_stream *params, + int num_params, const char **w_param_text, + unsigned long *private_value) { - struct snd_soc_dapm_widget template; - struct snd_soc_dapm_widget *w; - char *link_name; - int ret, count; - unsigned long private_value; - const char **w_param_text; struct soc_enum w_param_enum[] = { SOC_ENUM_SINGLE(0, 0, 0, NULL), }; @@ -3798,19 +3807,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, snd_soc_dapm_dai_link_get, snd_soc_dapm_dai_link_put), }; + struct snd_kcontrol_new *kcontrol_news; const struct snd_soc_pcm_stream *config = params; - - w_param_text = devm_kcalloc(card->dev, num_params, - sizeof(char *), GFP_KERNEL); - if (!w_param_text) - return -ENOMEM; - - link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", - source->name, sink->name); - if (!link_name) { - ret = -ENOMEM; - goto outfree_w_param; - } + int count; for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { @@ -3827,47 +3826,87 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, strlen(config->stream_name) + 1, GFP_KERNEL); } - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } + if (!w_param_text[count]) + goto outfree_w_param; config++; } + w_param_enum[0].items = num_params; w_param_enum[0].texts = w_param_text; - memset(&template, 0, sizeof(template)); - template.reg = SND_SOC_NOPM; - template.id = snd_soc_dapm_dai_link; - template.name = link_name; - template.event = snd_soc_dai_link_event; - template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD; - template.num_kcontrols = 1; - /* duplicate w_param_enum on heap so that memory persists */ - private_value = + *private_value = (unsigned long) devm_kmemdup(card->dev, (void *)(kcontrol_dai_link[0].private_value), sizeof(struct soc_enum), GFP_KERNEL); - if (!private_value) { + if (!*private_value) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_link_name; + goto outfree_w_param; } - kcontrol_dai_link[0].private_value = private_value; + kcontrol_dai_link[0].private_value = *private_value; /* duplicate kcontrol_dai_link on heap so that memory persists */ - template.kcontrol_news = - devm_kmemdup(card->dev, &kcontrol_dai_link[0], - sizeof(struct snd_kcontrol_new), - GFP_KERNEL); - if (!template.kcontrol_news) { + kcontrol_news = + devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + if (!kcontrol_news) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_private_value; + goto outfree_w_param; } + return kcontrol_news; +outfree_w_param: + snd_soc_dapm_free_kcontrol(card, private_value, num_params, w_param_text); + return NULL; +} + +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + unsigned int num_params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget template; + struct snd_soc_dapm_widget *w; + const char **w_param_text; + unsigned long private_value; + char *link_name; + int ret; + + link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", + source->name, sink->name); + if (!link_name) + return -ENOMEM; + + memset(&template, 0, sizeof(template)); + template.reg = SND_SOC_NOPM; + template.id = snd_soc_dapm_dai_link; + template.name = link_name; + template.event = snd_soc_dai_link_event; + template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD; + template.kcontrol_news = NULL; + + /* allocate memory for control, only in case of multiple configs */ + if (num_params > 1) { + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) { + ret = -ENOMEM; + goto param_fail; + } + + template.num_kcontrols = 1; + template.kcontrol_news = + snd_soc_dapm_alloc_kcontrol(card, + link_name, params, num_params, + w_param_text, &private_value); + if (!template.kcontrol_news) { + ret = -ENOMEM; + goto param_fail; + } + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); @@ -3899,15 +3938,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kfree(card->dev, w); outfree_kcontrol_news: devm_kfree(card->dev, (void *)template.kcontrol_news); -outfree_private_value: - devm_kfree(card->dev, (void *)private_value); -outfree_link_name: + snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text); +param_fail: devm_kfree(card->dev, link_name); - for (count = 0 ; count < num_params; count++) - devm_kfree(card->dev, (void *)w_param_text[count]); -outfree_w_param: - devm_kfree(card->dev, w_param_text); - return ret; } -- 2.5.4 (Apple Git-61)

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