May 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/5] ASoC: rsnd: tidyup patches
by Kuninori Morimoto 12 May '15

12 May '15

Hi Mark These are not so very important, but tidyup patches for Renesas R-Car driver. Kuninori Morimoto (5): ASoC: rsnd: add rsnd_dai_to_priv() macro ASoC: rsnd: make sure it uses lock when it calls rsnd_dai_call ASoC: rsnd: tidyup SSI parent related function/macro names ASoC: rsnd: make sure SSI parent/child uses same number of sound channel. ASoC: rsnd: care snd_kcontrol's index sound/soc/sh/rcar/core.c | 34 +++++++++++++++++++++++++++------- sound/soc/sh/rcar/ssi.c | 52 +++++++++++++++++++++++++++++++++++++++++++++------- 2 files changed, 72 insertions(+), 14 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH v2] ASoC: wm8741: Add differential mono mode support
by Sergej Sawazki 11 May '15

11 May '15

The WM8741 DAC supports several differential output modes (stereo, stereo reversed, mono left, mono right). Add platform data and DT bindings to configure it. Signed-off-by: Sergej Sawazki <ce3a(a)gmx.de> --- Documentation/devicetree/bindings/sound/wm8741.txt | 11 +++ sound/soc/codecs/wm8741.c | 108 ++++++++++++++++++--- sound/soc/codecs/wm8741.h | 10 ++ 3 files changed, 117 insertions(+), 12 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/wm8741.txt b/Documentation/devicetree/bindings/sound/wm8741.txt index 74bda58..a133154 100644 --- a/Documentation/devicetree/bindings/sound/wm8741.txt +++ b/Documentation/devicetree/bindings/sound/wm8741.txt @@ -10,9 +10,20 @@ Required properties: - reg : the I2C address of the device for I2C, the chip select number for SPI. +Optional properties: + + - diff-mode: Differential output mode configuration. Default value for field + DIFF in register R8 (MODE_CONTROL_2). If absent, the default is 0, shall be: + 0 = stereo + 1 = mono left + 2 = stereo reversed + 3 = mono right + Example: codec: wm8741@1a { compatible = "wlf,wm8741"; reg = <0x1a>; + + diff-mode = <3>; }; diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 9e71c76..cbf90ab 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -41,6 +41,7 @@ static const char *wm8741_supply_names[WM8741_NUM_SUPPLIES] = { /* codec private data */ struct wm8741_priv { + struct wm8741_platform_data pdata; struct regmap *regmap; struct regulator_bulk_data supplies[WM8741_NUM_SUPPLIES]; unsigned int sysclk; @@ -398,7 +399,7 @@ static struct snd_soc_dai_driver wm8741_dai = { .name = "wm8741", .playback = { .stream_name = "Playback", - .channels_min = 2, /* Mono modes not yet supported */ + .channels_min = 2, .channels_max = 2, .rates = WM8741_RATES, .formats = WM8741_FORMATS, @@ -416,6 +417,60 @@ static int wm8741_resume(struct snd_soc_codec *codec) #define wm8741_resume NULL #endif +static int wm8741_configure(struct snd_soc_codec *codec) +{ + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + /* Configure differential mode */ + switch (wm8741->pdata.diff_mode) { + case WM8741_DIFF_MODE_STEREO: + case WM8741_DIFF_MODE_STEREO_REVERSED: + case WM8741_DIFF_MODE_MONO_LEFT: + case WM8741_DIFF_MODE_MONO_RIGHT: + snd_soc_update_bits(codec, WM8741_MODE_CONTROL_2, + WM8741_DIFF_MASK, + wm8741->pdata.diff_mode << WM8741_DIFF_SHIFT); + break; + default: + return -EINVAL; + } + + /* Change some default settings - latch VU */ + snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, + WM8741_UPDATELL, WM8741_UPDATELL); + snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, + WM8741_UPDATELM, WM8741_UPDATELM); + snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + WM8741_UPDATERL, WM8741_UPDATERL); + snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION, + WM8741_UPDATERM, WM8741_UPDATERM); + + return 0; +} + +static int wm8741_add_controls(struct snd_soc_codec *codec) +{ + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + switch (wm8741->pdata.diff_mode) { + case WM8741_DIFF_MODE_STEREO: + case WM8741_DIFF_MODE_STEREO_REVERSED: + snd_soc_add_codec_controls(codec, wm8741_snd_controls, + ARRAY_SIZE(wm8741_snd_controls)); + break; + case WM8741_DIFF_MODE_MONO_LEFT: + case WM8741_DIFF_MODE_MONO_RIGHT: + /* The machine driver is responsible for mixer controls + * if the codec is configured in differential mono mode. + */ + break; + default: + return -EINVAL; + } + + return 0; +} + static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); @@ -434,15 +489,17 @@ static int wm8741_probe(struct snd_soc_codec *codec) goto err_enable; } - /* Change some default settings - latch VU */ - snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, - WM8741_UPDATELL, WM8741_UPDATELL); - snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, - WM8741_UPDATELM, WM8741_UPDATELM); - snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, - WM8741_UPDATERL, WM8741_UPDATERL); - snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION, - WM8741_UPDATERM, WM8741_UPDATERM); + ret = wm8741_configure(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to change default settings\n"); + goto err_enable; + } + + ret = wm8741_add_controls(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to add controls\n"); + goto err_enable; + } dev_dbg(codec->dev, "Successful registration\n"); return ret; @@ -467,8 +524,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .remove = wm8741_remove, .resume = wm8741_resume, - .controls = wm8741_snd_controls, - .num_controls = ARRAY_SIZE(wm8741_snd_controls), .dapm_widgets = wm8741_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8741_dapm_widgets), .dapm_routes = wm8741_dapm_routes, @@ -493,6 +548,23 @@ static const struct regmap_config wm8741_regmap = { .readable_reg = wm8741_readable, }; +static int wm8741_set_pdata(struct device *dev, struct wm8741_priv *wm8741) +{ + const struct wm8741_platform_data *pdata = dev_get_platdata(dev); + u32 diff_mode; + + if (dev->of_node) { + if (of_property_read_u32(dev->of_node, "diff-mode", &diff_mode) + >= 0) + wm8741->pdata.diff_mode = diff_mode; + } else { + if (pdata != NULL) + memcpy(&wm8741->pdata, pdata, sizeof(wm8741->pdata)); + } + + return 0; +} + #if IS_ENABLED(CONFIG_I2C) static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -522,6 +594,12 @@ static int wm8741_i2c_probe(struct i2c_client *i2c, return ret; } + wm8741_set_pdata(&i2c->dev, wm8741); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to set pdata: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, wm8741); ret = snd_soc_register_codec(&i2c->dev, @@ -582,6 +660,12 @@ static int wm8741_spi_probe(struct spi_device *spi) return ret; } + wm8741_set_pdata(&spi->dev, wm8741); + if (ret != 0) { + dev_err(&spi->dev, "Failed to set pdata: %d\n", ret); + return ret; + } + spi_set_drvdata(spi, wm8741); ret = snd_soc_register_codec(&spi->dev, diff --git a/sound/soc/codecs/wm8741.h b/sound/soc/codecs/wm8741.h index 56c1b1d..c8835f6 100644 --- a/sound/soc/codecs/wm8741.h +++ b/sound/soc/codecs/wm8741.h @@ -194,6 +194,12 @@ #define WM8741_DITHER_SHIFT 0 /* DITHER - [1:0] */ #define WM8741_DITHER_WIDTH 2 /* DITHER - [1:0] */ +/* DIFF field values */ +#define WM8741_DIFF_MODE_STEREO 0 /* stereo normal */ +#define WM8741_DIFF_MODE_STEREO_REVERSED 2 /* stereo reversed */ +#define WM8741_DIFF_MODE_MONO_LEFT 1 /* mono left */ +#define WM8741_DIFF_MODE_MONO_RIGHT 3 /* mono right */ + /* * R32 (0x20) - ADDITONAL_CONTROL_1 */ @@ -208,4 +214,8 @@ #define WM8741_SYSCLK 0 +struct wm8741_platform_data { + u32 diff_mode; /* Differential Output Mode */ +}; + #endif -- 1.9.1

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[alsa-devel] Atmel_usba_udc: Kernel panic in usba_ep_enable
by Angelo Compagnucci 11 May '15

11 May '15

Hello List, I'm having troubles with usb audio on an SAMA5D3 on 3.19.8 with a strange kernel panic. The function f_audio_set_alt (f_uac1.c) is passing a NULL out_ep and so the usba_ep_enable crashes with a null pointer dereference as you can see from the debugging output: [ 24.210000] ep1: ep_enable: desc= (null) Unfortunately, there is nothing strange with my configuration, I'm trying to use the soundcard (which plays files flawlessly) as a usb soundcard, so the only thing I did was to compile g_audio as a module, enable UAC1 and did a modprode. Can you share any advice? Following kernel panic output: [ 23.500000] g_audio gadget: adding config #1 'Linux USB Audio Gadget'/bf0232c8 [ 23.510000] g_audio gadget: adding 'g_audio'/cf9c6400 to config 'Linux USB Audio Gadget'/bf0232c8 [ 23.600000] g_audio gadget: Hardware params: access 3, format 2, channels 2, rate 48000 [ 23.630000] g_audio gadget: cfg 1/bf0232c8 speeds: high full [ 23.630000] g_audio gadget: interface 0 = g_audio/cf9c6400 [ 23.640000] g_audio gadget: interface 1 = g_audio/cf9c6400 [ 23.640000] g_audio gadget: Linux USB Audio Gadget, version: Feb 2, 2012 [ 23.650000] g_audio gadget: g_audio ready [ 24.200000] g_audio gadget: high-speed config #1: Linux USB Audio Gadget [ 24.200000] g_audio gadget: intf 0, alt 0 [ 24.200000] g_audio gadget: intf 1, alt 0 [ 24.210000] g_audio gadget: intf 1, alt 1 [ 24.210000] ep1: ep_enable: desc= (null) [ 24.210000] Unable to handle kernel NULL pointer dereference at virtual address 00000002 [ 24.210000] pgd = c0004000 [ 24.210000] [00000002] *pgd=00000000 [ 24.210000] Internal error: Oops: 17 [#1] ARM [ 24.210000] Modules linked in: usb_f_uac1 g_audio libcomposite [ 24.210000] CPU: 0 PID: 0 Comm: swapper Not tainted 3.19.8+ #69 [ 24.210000] Hardware name: Atmel SAMA5 (Device Tree) [ 24.210000] task: c052c7b8 ti: c0522000 task.ti: c0522000 [ 24.210000] PC is at usba_ep_enable+0x28/0x200 [ 24.210000] LR is at usba_ep_enable+0x24/0x200 [ 24.210000] pc : [<c02815bc>] lr : [<c02815b8>] psr: 200f0193 [ 24.210000] sp : c0523e18 ip : 00000007 fp : 0000bb80 [ 24.210000] r10: 00000100 r9 : cfb17c00 r8 : 00000001 [ 24.210000] r7 : 000000c8 r6 : cf99aa10 r5 : 00000000 r4 : cf99d868 [ 24.210000] r3 : 00000006 r2 : 00000007 r1 : 00000000 r0 : 00000001 [ 24.210000] Flags: nzCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment kernel [ 24.210000] Control: 10c53c7d Table: 2fb28059 DAC: 00000015 [ 24.210000] Process swapper (pid: 0, stack limit = 0xc0522230) [ 24.210000] Stack: (0xc0523e18 to 0xc0524000) [ 24.210000] 3e00: 00000001 cf99d868 [ 24.210000] 3e20: cf9c6400 bf028694 00000001 c0530008 cfa13ac0 cfb17c00 cfb17a00 00000001 [ 24.210000] 3e40: cf9c6400 00000001 00000001 cf99aa18 00000000 bf015104 c0523e74 c0034d9c [ 24.210000] 3e60: cf9471c0 600f0193 cfb05f6c c0530704 c00398b8 c00398c4 c05306f8 c0039400 [ 24.210000] 3e80: 00000000 200f0193 00000007 00000006 00000000 cf99d810 0000000b cf99d848 [ 24.210000] 3ea0: cf99aa10 00000000 00000000 00000000 00000800 c0282f0c 0000001c ffffffff [ 24.210000] 3ec0: 7fffffff c0547d80 00000000 00000125 00000001 c054e23e 00010b01 00000001 [ 24.210000] 3ee0: 00000000 cf998240 cf802540 00000000 00000000 0000002d cf901dc0 c054e23e [ 24.210000] 3f00: 00000000 c0040af0 cf802200 c054eb00 cf901dc0 cf802540 00000000 00000000 [ 24.210000] 3f20: cf802200 c051ac50 cfffc480 c0040be0 cf901dc0 c0042e00 0000002d 0000002d [ 24.210000] 3f40: 00000000 c0040314 c0534ca0 c0040538 c000eec4 600f0013 ffffffff c0523f9c [ 24.210000] 3f60: c054e340 c0011b00 00000000 c0530918 00000000 c0017880 c0522000 c052a0c4 [ 24.210000] 3f80: c054e23c c054e23c c054e340 c051ac50 cfffc480 00000000 01000000 c0523fb0 [ 24.210000] 3fa0: c000eec0 c000eec4 600f0013 ffffffff c0522000 c0039b58 ffffffff c04f5c30 [ 24.210000] 3fc0: ffffffff ffffffff c04f5678 00000000 00000000 c051ac50 c054e514 c052a070 [ 24.210000] 3fe0: c051ac4c c052d860 20004059 410fc051 00000000 20008070 00000000 00000000 [ 24.210000] [<c02815bc>] (usba_ep_enable) from [<bf028694>] (f_audio_set_alt+0xc4/0x19c [usb_f_uac1]) [ 24.210000] [<bf028694>] (f_audio_set_alt [usb_f_uac1]) from [<bf015104>] (composite_setup+0x4e0/0x1640 [libcomposite]) [ 24.210000] [<bf015104>] (composite_setup [libcomposite]) from [<c0282f0c>] (usba_udc_irq+0x75c/0xc74) [ 24.210000] [<c0282f0c>] (usba_udc_irq) from [<c0040af0>] (handle_irq_event_percpu+0x78/0x140) [ 24.210000] [<c0040af0>] (handle_irq_event_percpu) from [<c0040be0>] (handle_irq_event+0x28/0x38) [ 24.210000] [<c0040be0>] (handle_irq_event) from [<c0042e00>] (handle_fasteoi_irq+0x9c/0x164) [ 24.210000] [<c0042e00>] (handle_fasteoi_irq) from [<c0040314>] (generic_handle_irq+0x2c/0x3c) [ 24.210000] [<c0040314>] (generic_handle_irq) from [<c0040538>] (__handle_domain_irq+0x54/0xa8) [ 24.210000] [<c0040538>] (__handle_domain_irq) from [<c0011b00>] (__irq_svc+0x40/0x54) [ 24.210000] [<c0011b00>] (__irq_svc) from [<c000eec4>] (arch_cpu_idle+0x38/0x3c) [ 24.210000] [<c000eec4>] (arch_cpu_idle) from [<c0039b58>] (cpu_startup_entry+0xa8/0xf4) [ 24.210000] [<c0039b58>] (cpu_startup_entry) from [<c04f5c30>] (start_kernel+0x374/0x380) [ 24.210000] Code: e5941004 e5946024 eb04d705 e5d40033 (e5d51002) [ 24.210000] ---[ end trace d5ac053970e7d2d1 ]--- [ 24.210000] Kernel panic - not syncing: Fatal exception in interrupt [ 24.210000] ---[ end Kernel panic - not syncing: Fatal exception in interrupt -- Profile: http://it.linkedin.com/in/compagnucciangelo

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[alsa-devel] [PATCH 01/11] ASoC: 88pm860x: Replace direct snd_soc_codec dapm field access
by Lars-Peter Clausen 11 May '15

11 May '15

The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm.bias_level with snd_soc_codec_get_bias_level(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- All patches in this series have a dependency on topic/dapm --- sound/soc/codecs/88pm860x-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index c0b2686..ee31fa7 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1140,7 +1140,7 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ data = AUDIO_PLL | AUDIO_SECTION_ON; pm860x_reg_write(pm860x->i2c, REG_MISC2, data); -- 1.8.0

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[alsa-devel] [PATCH 0/3] ASoC: DPCM can care FE/BE merged format
by Kuninori Morimoto 11 May '15

11 May '15

Hi Mark These patches add new .dpcm_merged_foramt to merge FE/BE formats. It will be no sound if DPCM were below style, and user selects S24_LE. FE: S16_LE/S24_LE BE: S16_LE DPCM will selects FE/BE merged formats (= S16_LE here) if FE has .dpcm_merged_foramt. Kuninori Morimoto (3): ASoC: soc.h: tidyup struct snd_soc_dai_link definition order ASoC: soc-pcm: DPCM cares BE format ASoC: rsnd: rsrc-card uses FE/BE merged format when DPCM include/sound/soc.h | 39 +++++++++++++++++++++------------------ sound/soc/sh/rcar/rsrc-card.c | 1 + sound/soc/soc-pcm.c | 47 ++++++++++++++++++++++++++++++++++++++++++----- 3 files changed, 64 insertions(+), 23 deletions(-) Best regards --- Kuninori Morimoto

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Re: [alsa-devel] [PATCH 2/2] sound/usb: Update ALSA driver to use media controller API
by Shuah Khan 11 May '15

11 May '15

On 05/08/2015 10:51 PM, Eric Wong wrote: > Shuah Khan <shuahkh(a)osg.samsung.com> wrote: >> @@ -541,13 +591,19 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, >> [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk, >> [QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk, >> }; >> + int ret; >> >> + if (quirk->media_device) { >> + /* don't want to fail when media_device_init() doesn't work */ >> + ret = media_device_init(iface); >> + } >> if (quirk->type < QUIRK_TYPE_COUNT) { >> - return quirk_funcs[quirk->type](chip, iface, driver, quirk); >> + ret = quirk_funcs[quirk->type](chip, iface, driver, quirk); >> } else { >> usb_audio_err(chip, "invalid quirk type %d\n", quirk->type); >> return -ENXIO; >> } >> + return ret; >> } > > What is the point of saving 'ret' of media_device_init if it'll > only be clobbered or ignored for ENXIO? > Agreed. There is no point in saving it. thanks, -- Shuah -- Shuah Khan Sr. Linux Kernel Developer Open Source Innovation Group Samsung Research America (Silicon Valley) shuahkh(a)osg.samsung.com | (970) 217-8978

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Re: [alsa-devel] [PATCH 1/2] ASoC: rt5645: fix jack type detect error
by Mark Brown 11 May '15

11 May '15

On Mon, May 11, 2015 at 02:33:36AM +0000, John Lin wrote: > > -----Original Message----- > > From: Mark Brown [mailto:broonie@kernel.org] > > Sent: Saturday, May 09, 2015 1:14 AM > > To: John Lin > > Cc: lgirdwood(a)gmail.com; alsa-devel(a)alsa-project.org; Bard Liao; Flove; > > Oder Chiou; koro.chen(a)mediatek.com; yang.a.fang(a)intel.com; > > zhengxing(a)rock-chips.com > > Subject: Re: [PATCH 1/2] ASoC: rt5645: fix jack type detect error > > On Fri, May 08, 2015 at 01:53:33PM +0800, John Lin wrote: > > > rt5645_jack_detect doesn't report the correct jack type consistently. > > > This patch can make it more stable. > > Can you be more specific about what the problem is and how this fixes it > > pleased? It's a bit hard to review based on your changelog. > It mistakes OMTP type headset to CTIA type if "LDO2" power is disable in > jack detection function. So, we keep "LDO2" power enabled until jack out. > We also change the register sequence to make it more stable and remove > unnecessary power such as "micbias1" and "micbias2" for jack detection. This stuff needs to be in the changelog. It also sounds like there's at least two probably three patches here - one to keep the LDO power enabled for lonnger and one or more changes to the register sequence.

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[alsa-devel] stereo uac device control over mixer
by Arijit De 11 May '15

11 May '15

Hi, I am working on a USB audio MIC device(capture device), implemented UAC1.0 spec inside the firmware for that, my device is a stereo uac capture? device. I am trying to enable left and write channel separate control(volume, mute etc.) over mixer interface. my understanding about how to enable channel specific control is in UAC Feature Unit Descriptor if I add bmaControls(0) and bmaControls(1) with correct bitmap for Mute & Volume support amixer is able to identify my device as a two MIC device, amixer output for my usb audio capture device ********************************** Simple mixer control 'Mic',0 Capabilities: cvolumecaps joined 780 cvolume-joined cswitch cswitch-joined Capture channels: Mono Limits: Capture 0 - 15 Mono: Capture 9 [60%] [27.00dB] [on] Simple mixer control 'Mic',1 Capabilities: cvolumecaps joined 780 cvolume-joined cswitch cswitch-joined Capture channels: Mono Limits: Capture 0 - 15 Mono: Capture 9 [60%] [27.00dB] [on] ********************************** is the correct way of achieving it OR ideally there should be a single Mic with "Front Left - Front Right" support ? Thanks Arijit

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[alsa-devel] [PATCH 1/2] ASoC: fsl: fsl_dma: Use true/false for boolean init
by Fabio Estevam 11 May '15

11 May '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Bool initializations should use true and false. Bool tests don't need comparisons. Based on contributions from Joe Perches, Rusty Russell and Bruce W Allan. The semantic patch that makes this change is available in scripts/coccinelle/misc/boolinit.cocci. More information about semantic patching is available at http://coccinelle.lip6.fr/ Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 93d7e56..ccadefc 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -445,7 +445,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) return ret; } - dma->assigned = 1; + dma->assigned = true; snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware); @@ -814,7 +814,7 @@ static int fsl_dma_close(struct snd_pcm_substream *substream) substream->runtime->private_data = NULL; } - dma->assigned = 0; + dma->assigned = false; return 0; } -- 1.9.1

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Re: [alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 11 May '15

11 May '15

> I started MS Windows and without externals speakers plugged in I could test > 5.1 configuration and it seemed that sound was coming from different directions. > > What I managed to determine: > I've got 3 stacks: > 1 ST: Headphone, Line-out (FL and FR), SPDIF > 2 ST: Microphone, Line-out (CLFE) > 3 ST: Line-in, Line-out (RL and RR) > > I've got 5 responsive output pins > PIN 0x14: Internal FL FR > PIN 0x15: External FL FR > PIN 0x17: Internal LFE > PIN 0x18: External CLFE > PIN 0x1a: External RL RR > > How can we make it work with these information? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/plain/Document… seem chnmap type _FIXED , _VAR and _PAIRED does not fit when LFE and FC can be swapped for internal speaker and external speaker https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… How do the driver implement channel map when codec has swap center/lfe switch ? Do chnmap contain both {FL FR RL RR FC LFE} and {FL FR RL RR LFE FC} ? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… atihdmi_paired_swap_fc_lfe

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