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

[alsa-devel] [PATCH] ALSA: au88x0: avoid theoretical uninitialized access
by Arnd Bergmann 24 Mar '17

24 Mar '17

The latest gcc-7.0.1 snapshot points out that we if nr_ch is zero, we never initialize some variables: sound/pci/au88x0/au88x0_core.c: In function 'vortex_adb_allocroute': sound/pci/au88x0/au88x0_core.c:2304:68: error: 'mix[0]' may be used uninitialized in this function [-Werror=maybe-uninitialized] sound/pci/au88x0/au88x0_core.c:2305:58: error: 'src[0]' may be used uninitialized in this function [-Werror=maybe-uninitialized] I assume this can never happen in practice, but adding a check here doesn't hurt either and avoids the warning. The code has been unchanged since the start of git history. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/au88x0/au88x0_core.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index e1af24f87566..c308a4f70550 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2279,6 +2279,9 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, } else { int src[2], mix[2]; + if (nr_ch < 1) + return -EINVAL; + /* Get SRC and MIXER hardware resources. */ for (i = 0; i < nr_ch; i++) { if ((mix[i] = -- 2.9.0

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[alsa-devel] [PATCH] ASoC: core: remove pointless auxiliary from snd_soc_component
by Kuninori Morimoto 24 Mar '17

24 Mar '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 1a653aa44725 ("ASoC: core: replace aux_comp_list to ...") tried to replace aux_comp_list to component_dev_list, but it failed because of binding timing. Thus, Sylwester fixuped it by commit d2e3a1358c37 ("ASoC: Fix binding and probing of auxiliary..."). One of main purpose of commit 1a653aa44725 ("ASoC: core: replace...") was remove replaceable list (= list_aux) from snd_soc_component by using new "auxiliary" flags (but it failed). Because of this background, current code has reborned card_aux_list (= same as original list_aux), and almost pointless "auxiliary" flags. Let's remove pointless "auxiliary" flags by this patch This means, it is same as revert both commit 1a653aa44725 ("ASoC: core: replace aux_comp_list to ...") and commit d2e3a1358c37 ("ASoC: Fix binding and probing of auxiliary..."). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- include/sound/soc.h | 1 - sound/soc/soc-core.c | 15 +++++---------- 2 files changed, 5 insertions(+), 11 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index dd78507..316fdce 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -815,7 +815,6 @@ struct snd_soc_component { unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ unsigned int registered_as_component:1; - unsigned int auxiliary:1; /* for auxiliary component of the card */ unsigned int suspended:1; /* is in suspend PM state */ struct list_head list; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e7d876a..5933851 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1777,7 +1777,6 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) } component->init = aux_dev->init; - component->auxiliary = 1; list_add(&component->card_aux_list, &card->aux_comp_list); return 0; @@ -1789,14 +1788,13 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) static int soc_probe_aux_devices(struct snd_soc_card *card) { - struct snd_soc_component *comp, *tmp; + struct snd_soc_component *comp; int order; int ret; for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - list_for_each_entry_safe(comp, tmp, &card->aux_comp_list, - card_aux_list) { + list_for_each_entry(comp, &card->aux_comp_list, card_aux_list) { if (comp->driver->probe_order == order) { ret = soc_probe_component(card, comp); if (ret < 0) { @@ -1805,7 +1803,6 @@ static int soc_probe_aux_devices(struct snd_soc_card *card) comp->name, ret); return ret; } - list_del(&comp->card_aux_list); } } } @@ -1821,14 +1818,12 @@ static void soc_remove_aux_devices(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { list_for_each_entry_safe(comp, _comp, - &card->component_dev_list, card_list) { - - if (!comp->auxiliary) - continue; + &card->aux_comp_list, card_aux_list) { if (comp->driver->remove_order == order) { soc_remove_component(comp); - comp->auxiliary = 0; + /* remove it from the card's aux_comp_list */ + list_del(&comp->card_aux_list); } } } -- 1.9.1 _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] [PATCH] ALSA: hda - Adding a group of pin definition to fix headset problem
by Hui Wang 23 Mar '17

23 Mar '17

A new Dell laptop needs to apply ALC269_FIXUP_DELL1_MIC_NO_PRESENCE to fix the headset problem, and the pin definiton of this machine is not in the pin quirk table yet, now adding it to the table. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1561076..efa657f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6191,6 +6191,8 @@ enum { ALC295_STANDARD_PINS, {0x17, 0x21014040}, {0x18, 0x21a19050}), + SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC295_STANDARD_PINS), SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC298_STANDARD_PINS, {0x17, 0x90170110}), -- 1.9.1

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[alsa-devel] [PATCH v3] ASoC: rt286: fix headphone click/crack noise on Dell XPS 9343 I2S mode
by Kai-Heng Feng 23 Mar '17

23 Mar '17

HDA mode fixed the issue by these two commits: '9476d369d7b3 ALSA: hda - Mute headphone pin on suspend on XPS13 9333' '3e1b0c4a9d56 ALSA: hda - Fix click noise at start on Dell XPS13' Apply the same workarounds to rt286 can solve the issue. When jack is plugged, it rapidly generates I2C interrupts, which triggers rt286_irq() and rt286_jack_detect(), which produces the click noise. alc_fixup_dell_xps13() in patch_realtek.c sets up a pin that can stop the frantic interrupts, hence avoids the click noise. When rt286 is under powersaving state, play a sound with headphone or plug a headphone in will produce a loud crack sound. Set AMP_OUT_MUTE before power events can make the noise less noticeable. Unmute the AMP when the power is up. v3: Implicit conversion instead of tenary operator. v2: Use 'HP Power' instead of individual power events. Link: https://bugzilla.kernel.org/show_bug.cgi?id=112611 Link: https://bugzilla.redhat.com/show_bug.cgi?id=1313434 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/soc/codecs/rt286.c | 48 +++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 45 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 9c365a7f758d..97b52697f974 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -36,6 +36,9 @@ #define RT286_VENDOR_ID 0x10ec0286 #define RT288_VENDOR_ID 0x10ec0288 +#define AMP_OUT_MUTE 0xb080 +#define AMP_OUT_UNMUTE 0xb000 + struct rt286_priv { struct reg_default *index_cache; int index_cache_size; @@ -47,6 +50,7 @@ struct rt286_priv { struct delayed_work jack_detect_work; int sys_clk; int clk_id; + bool is_dell_dino; }; static const struct reg_default rt286_index_def[] = { @@ -472,6 +476,32 @@ static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, return 0; } +/* Power event function to workaround headphone crack noise */ +static int rt286_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (!rt286->is_dell_dino) + return 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_POST_PMD: + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, RT286_SET_AMP_GAIN_HPO, AMP_OUT_MUTE); + break; + case SND_SOC_DAPM_PRE_PMU: + snd_soc_write(codec, RT286_SET_AMP_GAIN_HPO, AMP_OUT_UNMUTE); + break; + default: + return 0; + } + + return 0; +} + static int rt286_ldo2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -578,7 +608,9 @@ static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt286_hpo_mux), SND_SOC_DAPM_SUPPLY("HP Power", RT286_SET_PIN_HPO, - RT286_SET_PIN_SFT, 0, NULL, 0), + RT286_SET_PIN_SFT, 0, rt286_hp_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Output Mixer */ SND_SOC_DAPM_MIXER("Front", RT286_SET_POWER(RT286_DAC_OUT1), 0, 1, @@ -1175,8 +1207,10 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; + rt286->is_dell_dino = dmi_check_system(dmi_dell_dino); + if (dmi_check_system(force_combo_jack_table) || - dmi_check_system(dmi_dell_dino)) + rt286->is_dell_dino) rt286->pdata.cbj_en = true; regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); @@ -1192,6 +1226,11 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_CBJ_CTRL1, 0xf000, 0xb000); } else { + /* Fix headphone click noise */ + if (rt286->is_dell_dino) + regmap_write(rt286->regmap, + RT286_MIC1_DET_CTRL, 0x0020); + regmap_update_bits(rt286->regmap, RT286_CBJ_CTRL1, 0xf000, 0x5000); } @@ -1215,7 +1254,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); - if (dmi_check_system(dmi_dell_dino)) { + if (rt286->is_dell_dino) { regmap_update_bits(rt286->regmap, RT286_SET_GPIO_MASK, 0x40, 0x40); regmap_update_bits(rt286->regmap, @@ -1224,6 +1263,9 @@ static int rt286_i2c_probe(struct i2c_client *i2c, RT286_SET_GPIO_DATA, 0x40, 0x40); regmap_update_bits(rt286->regmap, RT286_GPIO_CTRL, 0xc, 0x8); + /* Workaound headphone crack noise when probing */ + regmap_write(rt286->regmap, RT286_SET_AMP_GAIN_HPO, + AMP_OUT_MUTE); } if (rt286->i2c->irq) { -- 2.12.0

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[alsa-devel] [PATCH 0/4] ASoC: sun8i-codec: Clean-up
by Mylène Josserand 21 Mar '17

21 Mar '17

Hi all, Thanks to Chen-Yu's review [1], here are some patches to clean-up my sun8i A33 audio driver. Patches tested on asoc/for-next branch with additionnal patches to apply before: https://patchwork.kernel.org/patch/9447631/ https://patchwork.kernel.org/patch/9423999/ https://patchwork.kernel.org/patch/9521121/ https://www.spinics.net/lists/arm-kernel/msg560091.html Patch 1: Removes the HP widget which is available in the analog part. Patch 2: Update the driver to use SOC_DAPM_DOUBLE and rename the controls accordingly. Patch 3: Convert SND_SOC_DAPM_MIXER to SOC_MIXER_ARRAY. Patch 3: Convert SND_SOC_DAPM_DAC to SND_SOC_DAPM_AIF_IN definition. The last Chen Yu's comments are still in progress (ie use of set_pll and set_sysclk and reworking the audio route). After this series, amixer commands to test an audio file: amixer set 'Headphone' 75% amixer set 'Headphone' on amixer set 'DAC' on amixer set 'AIF1 Slot 0 Digital DAC' on [1] https://lkml.org/lkml/2017/2/7/103 Thank you in advance, Best regards, Mylène Josserand (4): ASoC: sun8i-codec: Remove analog "HP" widget ASoC: sun8i-codec: Update mixer to use SOC_DAPM_DOUBLE ASoC: sun8i-codec: Convert to SOC_MIXER_ARRAY ASoC: sun8i-codec: Convert to use SND_SOC_DAPM_AIF_IN arch/arm/boot/dts/sun8i-a33.dtsi | 4 +-- sound/soc/sunxi/sun8i-codec.c | 65 +++++++++++++++++----------------------- 2 files changed, 30 insertions(+), 39 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH v2 0/3] ASoC: sun8i-codec: Clean-up
by Mylène Josserand 21 Mar '17

21 Mar '17

Hello everyone, Thanks to Chen-Yu's review [1], a first version of this patchset to clean-up my sun8i A33 audio driver has been sent and merged. One patch [2] needs a rework so this cover-letter is for its V2. Changes with v1: - Remove patches 01, 02 and 03 as they are merged in asoc/for-next. - Added a patch to fix an error on an widget name - Split the patch 04 into two patches to separate the device tree modifications with the codec driver's one. Patches tested on asoc/for-next branch, last commit: 698b3eb9af. Patch 1: Fix a dapm routing error because of an unwanted space. Patch 2: Convert SND_SOC_DAPM_DAC to SND_SOC_DAPM_AIF_IN definition. The AIF interfaces are renamed to be more precise. Patch 3: Update the audio-routing's device tree to use the AIF interfaces renamed in patch 01. After this series, amixer commands to test an audio file: amixer set 'Headphone' 75% amixer set 'Headphone' on amixer set 'DAC' on amixer set 'AIF1 Slot 0 Digital DAC' on [1] https://lkml.org/lkml/2017/2/7/103 [2] https://patchwork.kernel.org/patch/9566325/ Thank you in advance, Best regards, Mylène Josserand (3): ASoC: sun8i-codec: Fix space on audio-routing widget ASoC: sun8i-codec: Convert to use SND_SOC_DAPM_AIF_IN ARM: dts: sun8i: Update audio-routing with renamed widgets arch/arm/boot/dts/sun8i-a33.dtsi | 4 ++-- sound/soc/sunxi/sun8i-codec.c | 22 ++++++++++++---------- 2 files changed, 14 insertions(+), 12 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH v3 0/5] ASoC: sun8i-a33 clean-up for 4.11
by Mylène Josserand 21 Mar '17

21 Mar '17

Hi everyone, As discussed in this thread [1], here are a V3 of the cleanup series for the sun8i-a33 driver (introduced in v4.11-rc1) as fixes for v4.11. Based on: Linus repository - master branch Last commit: 8841b5f0cd ("Merge tag 'nfs-for-4.11-2' of git://git.linux-nfs.org/projects/anna/linux-nfs") Changes since v1 and v2: - Patch 01: No changes from v1, it applies directly - Patch 02: No changes from v1, it applies directly once patch 01 has been applied. - Patch 03: No changes from v2, it applies directly - Patch 04: Rebased on master branch - Patch 05: No changes from v2, it applies directly Patch 01 removes some unnecessary widgets. Patch 02 updates the driver to use SOC_DAPM_DOUBLE. Patch 03 removes a space on a widget name that implies an error on audio routing as the widget is unknown. Patch 04 improves some widgets (names and types). As widget names changed in patch 04, the patch 05 is needed to use the same names on audio routing. Otherwise, the device tree for sun8i-a33 will be broken and the audio codec will not work. Only patches 01 and 02 were initially wanted as fixes for v4.11 but, with Chen-Yu, we thought that patch 03 and 04 could be also applied as fixes. As patch 04 creates a break with the device tree, I added patch 05 in this series. Let me know what you think. [1]: https://lkml.org/lkml/2017/3/14/4 Thank you in advance, Best regards, Mylène Josserand (5): ASoC: sun8i-codec: Remove analog "HP" widget ASoC: sun8i-codec: Update mixer to use SOC_DAPM_DOUBLE ASoC: sun8i-codec: Fix space on audio-routing widget ASoC: sun8i-codec: Convert to use SND_SOC_DAPM_AIF_IN ARM: dts: sun8i: Update audio-routing with renamed widgets arch/arm/boot/dts/sun8i-a33.dtsi | 4 +-- sound/soc/sunxi/sun8i-codec.c | 67 ++++++++++++++++++---------------------- 2 files changed, 32 insertions(+), 39 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH v2] ASoC: Intel: Update bxt_da7219_max98357a to add a new
by Harsha Priya 21 Mar '17

21 Mar '17

This patch adds a platform clock widget to turn off the clock only when both headset capture and headset playback are not in use. This removes turning off the clock in hw_free so that the clock is on when either capture or playback of headset is in progress. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 97 +++++++++++++++------------ 1 file changed, 53 insertions(+), 44 deletions(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 2cda06c..3a8c4d9 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -55,6 +55,54 @@ enum { BXT_DPCM_AUDIO_HDMI3_PB, }; +static inline struct snd_soc_dai *bxt_get_codec_dai(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + + list_for_each_entry(rtd, &card->rtd_list, list) { + + if (!strncmp(rtd->codec_dai->name, BXT_DIALOG_CODEC_DAI, + strlen(BXT_DIALOG_CODEC_DAI))) + return rtd->codec_dai; + } + + return NULL; +} + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + int ret = 0; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + + codec_dai = bxt_get_codec_dai(card); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, + DA7219_SYSCLK_MCLK, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop PLL: %d\n", ret); + } else if(SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, + DA7219_CLKSRC_MCLK, 19200000, SND_SOC_CLOCK_IN); + if (ret) + dev_err(card->dev, "can't set codec sysclk configuration\n"); + + ret = snd_soc_dai_set_pll(codec_dai, 0, + DA7219_SYSCLK_PLL_SRM, 0, DA7219_PLL_FREQ_OUT_98304); + if (ret) + dev_err(card->dev, "failed to start PLL: %d\n", ret); + } + + return ret; +} + static const struct snd_kcontrol_new broxton_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -69,6 +117,8 @@ enum { SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), SND_SOC_DAPM_SPK("HDMI3", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_POST_PMD|SND_SOC_DAPM_PRE_PMU), }; static const struct snd_soc_dapm_route broxton_map[] = { @@ -109,6 +159,9 @@ enum { /* DMIC */ {"dmic01_hifi", NULL, "DMIC01 Rx"}, {"DMIC01 Rx", NULL, "DMIC AIF"}, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, }; static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd, @@ -243,49 +296,6 @@ static int bxt_fe_startup(struct snd_pcm_substream *substream) .startup = bxt_fe_startup, }; -static int broxton_da7219_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - DA7219_CLKSRC_MCLK, 19200000, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(codec_dai->dev, "can't set codec sysclk configuration\n"); - - ret = snd_soc_dai_set_pll(codec_dai, 0, - DA7219_SYSCLK_PLL_SRM, 0, DA7219_PLL_FREQ_OUT_98304); - if (ret < 0) { - dev_err(codec_dai->dev, "failed to start PLL: %d\n", ret); - return -EIO; - } - - return ret; -} - -static int broxton_da7219_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_pll(codec_dai, 0, - DA7219_SYSCLK_MCLK, 0, 0); - if (ret < 0) { - dev_err(codec_dai->dev, "failed to stop PLL: %d\n", ret); - return -EIO; - } - - return ret; -} - -static const struct snd_soc_ops broxton_da7219_ops = { - .hw_params = broxton_da7219_hw_params, - .hw_free = broxton_da7219_hw_free, -}; - static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -467,7 +477,6 @@ static int broxton_refcap_startup(struct snd_pcm_substream *substream) SND_SOC_DAIFMT_CBS_CFS, .ignore_pmdown_time = 1, .be_hw_params_fixup = broxton_ssp_fixup, - .ops = &broxton_da7219_ops, .dpcm_playback = 1, .dpcm_capture = 1, }, -- 1.9.1

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[alsa-devel] [PATCH 1/4] dt-bindings: Document the dmas and dma-names properties for bcm2835-analog-audio
by Michael Zoran 21 Mar '17

21 Mar '17

Add documentation for the DMA properties required when using the PWM controller of the bcm2835 for analog audio output. Signed-off-by: Michael Zoran <mzoran(a)crowfest.net> --- Documentation/devicetree/bindings/pwm/pwm-bcm2835.txt | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/pwm/pwm-bcm2835.txt b/Documentation/devicetree/bindings/pwm/pwm-bcm2835.txt index cf573e85b11d..08fc06fc083a 100644 --- a/Documentation/devicetree/bindings/pwm/pwm-bcm2835.txt +++ b/Documentation/devicetree/bindings/pwm/pwm-bcm2835.txt @@ -1,14 +1,21 @@ BCM2835 PWM controller (Raspberry Pi controller) Required properties: -- compatible: should be "brcm,bcm2835-pwm" +- compatible: should be "brcm,bcm2835-pwm" or "brcm,bcm2835-analog-audio" - reg: physical base address and length of the controller's registers - clocks: This clock defines the base clock frequency of the PWM hardware system, the period and the duty_cycle of the PWM signal is a multiple of the base period. + +Required properties(PWM Output): - #pwm-cells: Should be 2. See pwm.txt in this directory for a description of the cells format. +Required properties(Audio Output): +- dmas: Should contain one entry pointing to the DMA channel used to + transfer audio data +- dma-names: Should contain "tx" + Examples: pwm@2020c000 { -- 2.11.0

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[alsa-devel] hw/sw params API in alsa-lib miscompiled, result: unusable alsa-utils
by Deweloper 21 Mar '17

21 Mar '17

Hi, I built alsa-lib and alsa-utils (v.1.1.3) from sources, using GCC 6. When I tried launching arecord, it failed due to lack of memory. During analysis I found out that: - the message is printed because realloc(buf, period_size) returns NULL - realloc() returns NULL because period_size is 0 - period_size is 0 because snd_pcm_hw_params_get_period_size() puts 0 to *frames (instead of 32768) and returns 32768 (instead of 0) - snd_pcm_hw_params_get_period_size() misbehaves in this way because although a NEW API header is used, the OLD API implementation is actually called So it seems that I've run into exactly the same problem which was described by John Spencer in August 2013: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-August/065031.html > my problem is that the old version is getting called instead of the > new one Did anything change in this matter since August, 2013? --

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