April 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: imx-audmux: Fix ssi port numbers in sysfs
by Fabio Estevam 05 Apr '12

05 Apr '12

Doing a 'cat /sys/kernel/debug/audmux/ssi7' causes the following oops to be printed by the kernel: Uhandled fault: external abort on non-linefetch (0x008) at 0xf53b003c Internal error: : 8 [#1] PREEMPT Modules linked in: CPU: 0 Not tainted (3.3.0-00033-gecc726e-dirty #307) PC is at audmux_read_file+0x68/0x2f4 LR is at clk_enable+0x3c/0x48 pc : [<c001b8c8>] lr : [<c00190a0>] psr: a0000013 sp : c3ad3f38 ip : c30a4000 fp : 00000003 r10: 00001000 r9 : be83fb00 r8 : c3ad3f80 r7 : c3ad3f80 r6 : 00000007 r5 : 00031010 r4 : c30a5000 r3 : f53b0000 r2 : 0000003c r1 : 380fa100 r0 : c068dda0 Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 0005317f Table: 83034000 DAC: 00000015 Process cat (pid: 1042, stack limit = 0xc3ad2270) Stack: (0xc3ad3f38 to 0xc3ad4000) 3f20: c3139180 00000000 3f40: c3bc6500 00001000 be83fb00 c3ad3f80 00001000 c3ad2000 00000000 c0095f3c 3f60: 00000003 c3bc6508 c3bc6500 be83fb00 00000000 00000000 00001000 c0096010 3f80: 00000000 00000000 b6fe2050 00000000 00001000 be83fb00 00000003 00000003 3fa0: c000eb88 c000e9e0 00001000 be83fb00 00000003 be83fb00 00001000 00000000 3fc0: 00001000 be83fb00 00000003 00000003 00000001 00000001 00000000 00000003 3fe0: 000bec8c be83fae0 0000f808 b6ea8d5c 60000010 00000003 7dff7ede 749bedf1 [<c001b8c8>] (audmux_read_file+0x68/0x2f4) from [<c0095f3c>] (vfs_read+0xb0/0x144) [<c0095f3c>] (vfs_read+0xb0/0x144) from [<c0096010>] (sys_read+0x40/0x70) [<c0096010>] (sys_read+0x40/0x70) from [<c000e9e0>] (ret_fast_syscall+0x0/0x2c) Code: e1a02186 e2822004 e3500000 e7935186 (e7937002) ---[ end trace 4d046e31309023de ]--- Fix the ssi port numbers in sysfs to fix this problem. Reported-by: Joan Carles <joancarles(a)fqingenieria.es> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 1765a19..d467a1b 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -152,7 +152,7 @@ static void __init audmux_debugfs_init(void) return; } - for (i = 1; i < 8; i++) { + for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) { snprintf(buf, sizeof(buf), "ssi%d", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) -- 1.7.1

2 2

[alsa-devel] [PATCH] ASoC: wm8994: Implement FLL bypass support
by Mark Brown 05 Apr '12

05 Apr '12

Later WM8994 class devices can bypass the FLL from BCLK. Do this automatically when the FLL input and output frequencies match up. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 19 +++++++++++++++++++ sound/soc/codecs/wm8994.h | 1 + 2 files changed, 20 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e94a84d..1dee30d 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2112,6 +2112,14 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, WM8994_FLL1_ENA, 0); + if (wm8994->fll_byp && src == WM8994_FLL_SRC_BCLK && + freq_in == freq_out) { + dev_dbg(codec->dev, "Bypassing FLL%d\n", id + 1); + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, + WM8958_FLL1_BYP, WM8958_FLL1_BYP); + goto out; + } + reg = (fll.outdiv << WM8994_FLL1_OUTDIV_SHIFT) | (fll.fll_fratio << WM8994_FLL1_FRATIO_SHIFT); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_2 + reg_offset, @@ -2126,6 +2134,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, fll.n << WM8994_FLL1_N_SHIFT); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, + WM8958_FLL1_BYP | WM8994_FLL1_REFCLK_DIV_MASK | WM8994_FLL1_REFCLK_SRC_MASK, (fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT) | @@ -2188,6 +2197,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, } } +out: wm8994->fll[id].in = freq_in; wm8994->fll[id].out = freq_out; wm8994->fll[id].src = src; @@ -3714,6 +3724,14 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8958: wm8994->hubs.dcs_readback_mode = 1; wm8994->hubs.hp_startup_mode = 1; + + switch (wm8994->revision) { + case 0: + break; + default: + wm8994->fll_byp = true; + break; + } break; case WM1811: @@ -3721,6 +3739,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.no_series_update = 1; wm8994->hubs.hp_startup_mode = 1; wm8994->hubs.no_cache_class_w = true; + wm8994->fll_byp = true; switch (wm8994->revision) { case 0: diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index c724112..91650bb 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -79,6 +79,7 @@ struct wm8994_priv { struct wm8994_fll_config fll[2], fll_suspend[2]; struct completion fll_locked[2]; bool fll_locked_irq; + bool fll_byp; int vmid_refcount; int active_refcount; -- 1.7.9.1

1 0

Re: [alsa-devel] [PATCH 2/2] ASoC: imx-audmux: Check for NULL pointer
by Mark Brown 05 Apr '12

05 Apr '12

On Thu, Apr 05, 2012 at 02:49:16PM +0200, Marc Kleine-Budde wrote: > On 04/05/2012 02:45 PM, Fabio Estevam wrote: > > if (audmux_clk) > > clk_prepare_enable(audmux_clk); > > + if (!audmux_base) > > + return -ENOSYS; > What about "clk_prepare_enable"? Do you need to undo it in case of an error? Or alternatively just reverse the order of the two tests so you don't need to worry about that.

1 0

[alsa-devel] [PATCH] ASoC: ak4642: fixup: mute needs +1 step
by Kuninori Morimoto 05 Apr '12

05 Apr '12

ak4642 out_tlv is +12.0dB to -115.0 dB, and it supports mute. But current settings didn't care +1 step for mute. This patch adds it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 278c0a0..45a087b 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -140,7 +140,7 @@ * min : 0xFE : -115.0 dB * mute: 0xFF */ -static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); +static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1); static const struct snd_kcontrol_new ak4642_snd_controls[] = { -- 1.7.5.4

2 1

[alsa-devel] UHCI dropped audio samples / DMA starvation fix
by Monty Montgomery 04 Apr '12

04 Apr '12

Hello Clemens and Takashi (and list), I've been living with USB audio dropping partial frames on UHCI controllers for quite a few years. I've finally found the cause (DMA starvation) and have a lightweight fix. problem summary: Playback (but not capture) on UHCI controllers randomly loses samples; they are skipped completely during playback. This happens anywhere from a few times a second to once an hour or so. These are not ALSA underruns; the UHCI host controller itself is reporting hard DMA starvation errors. ALSA ignores these error and continues as if nothing happened, and the UHCI linux driver does not log this error anywhere. UHCI overlaps sending on the USB bus with DMAing the data from memory. It does not have all the data it needs to transmit a complete packet before it begins doing so. If the data it needs is in a dirty cache line on a CPU, the flush triggers after DMA is already started; the UHCI host controller has to wait for the flush before accessing it, and if the flush happens too late, UHCI aborts the packet. This is happening regularly on all the UHCI controllers I've tried thusfar. My guess is that this issue never attracted much attention because a) every one knows isoch transfers are 'unreliable' and b) other transfer types automatically retry so they eventually succeed without any obvious sign that something went wrong (and they're usually smaller than one cacheline and so unaffected anyway). Fix: The fix for usbaudio is dead easy. Call: clflush_cache_range(urb->transfer_buffer, urb->transfer_buffer_length); as the last step of prepare_playback_urb() and prepare_nodata_playback_urb(). This completely corrects the DMA starvation behavior on my machines here. Note that the clflush_cache_range() has nothing to do with maintaining coherency. The call is only to force the flush ahead of time to avoid the fatal latency spikes from on-demand flushes when DMA is already running. The drawback to this fix is that it's in one high level driver (ALSA USB audio) instead of inside UHCI where it really belongs. Unfortunately, the URB abstraction gives us no reliable way to do it in the HCD, because other drivers don't necessarily pass a valid CPU-side buffer address when using URB_NO_TRANSFER_DMA_MAP. Should I whip up some patches for usb audio for 2.6.x and 3.3.x and leave it at that, or should we discuss a bit more 'why isn't the kernel doing this for us?' Monty

2 2

[alsa-devel] [PATCH] ASoC: codecs: Remove rtd->codec usage from CODEC drivers
by Mark Brown 04 Apr '12

04 Apr '12

In order to support CODEC<->CODEC links remove the assumption that there is only a single CODEC on a DAI link by removing the use of the CODEC pointer in the rtd from the CODEC drivers. They are already being passed their DAI whenever they are passed an rtd and can get the CODEC from there. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ac97.c | 4 +--- sound/soc/codecs/ad1836.c | 4 +--- sound/soc/codecs/ad193x.c | 4 +--- sound/soc/codecs/adau1701.c | 3 +-- sound/soc/codecs/ak4104.c | 3 +-- sound/soc/codecs/ak4535.c | 3 +-- sound/soc/codecs/ak4641.c | 3 +-- sound/soc/codecs/alc5623.c | 3 +-- sound/soc/codecs/alc5632.c | 3 +-- sound/soc/codecs/cs4270.c | 3 +-- sound/soc/codecs/cs4271.c | 3 +-- sound/soc/codecs/cs42l51.c | 3 +-- sound/soc/codecs/cs42l73.c | 3 +-- sound/soc/codecs/da7210.c | 3 +-- sound/soc/codecs/jz4740.c | 3 +-- sound/soc/codecs/lm49453.c | 3 +-- sound/soc/codecs/rt5631.c | 3 +-- sound/soc/codecs/sgtl5000.c | 3 +-- sound/soc/codecs/ssm2602.c | 9 +++------ sound/soc/codecs/sta32x.c | 3 +-- sound/soc/codecs/tlv320aic23.c | 9 +++------ sound/soc/codecs/tlv320aic26.c | 3 +-- sound/soc/codecs/tlv320aic3x.c | 3 +-- sound/soc/codecs/tlv320dac33.c | 35 +++++++++++++++-------------------- sound/soc/codecs/twl4030.c | 18 ++++++------------ sound/soc/codecs/twl6040.c | 9 +++------ sound/soc/codecs/uda134x.c | 6 ++---- sound/soc/codecs/uda1380.c | 6 ++---- sound/soc/codecs/wl1273.c | 6 ++---- sound/soc/codecs/wm8400.c | 3 +-- sound/soc/codecs/wm8510.c | 3 +-- sound/soc/codecs/wm8523.c | 3 +-- sound/soc/codecs/wm8728.c | 3 +-- sound/soc/codecs/wm8737.c | 3 +-- sound/soc/codecs/wm8741.c | 3 +-- sound/soc/codecs/wm8750.c | 3 +-- sound/soc/codecs/wm8753.c | 6 ++---- sound/soc/codecs/wm8900.c | 3 +-- sound/soc/codecs/wm8903.c | 3 +-- sound/soc/codecs/wm8940.c | 3 +-- sound/soc/codecs/wm8960.c | 3 +-- sound/soc/codecs/wm8962.c | 3 +-- sound/soc/codecs/wm8971.c | 3 +-- sound/soc/codecs/wm8978.c | 3 +-- sound/soc/codecs/wm8988.c | 3 +-- sound/soc/codecs/wm8990.c | 3 +-- sound/soc/codecs/wm9705.c | 4 +--- sound/soc/codecs/wm9712.c | 8 ++------ 48 files changed, 78 insertions(+), 152 deletions(-) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 1bbad4c..a99a1b3 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -26,9 +26,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 12e3b41..c67b50d 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -162,9 +162,7 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { int word_len = 0; - - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* bit size */ switch (params_format(params)) { diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index a4a6bef..13e62be 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -245,9 +245,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { int word_len = 0, master_rate = 0; - - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); /* bit size */ diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 78e9ce4..3d50fc8 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -258,8 +258,7 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, static int adau1701_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; snd_pcm_format_t format; unsigned int val; diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index ceb96ec..31d4483 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -88,8 +88,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int val = 0; /* set the IEC958 bits: consumer mode, no copyright bit */ diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 838ae8b..618fdc3 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -262,8 +262,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); u8 mode2 = snd_soc_read(codec, AK4535_MODE2) & ~(0x3 << 5); int rate = params_rate(params), fs = 256; diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 7f42d4a..543a12f 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -296,8 +296,7 @@ static int ak4641_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); int rate = params_rate(params), fs = 256; u8 mode2; diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index d47b62d..3061d35 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -705,8 +705,7 @@ static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai, static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); int coeff, rate; u16 iface; diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index e2111e0..93a5909 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -861,8 +861,7 @@ static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai, static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int coeff, rate; u16 iface; diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index df8fe38..047917f 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -307,8 +307,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); int ret; unsigned int i; diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index bf71412..9eb01d7 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -318,8 +318,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); int i, ret; unsigned int ratio, val; diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 85d6069..091d019 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -356,8 +356,7 @@ static int cs42l51_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); int ret; unsigned int i; diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index d39b3e7..2cc50b3 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1089,8 +1089,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); int id = dai->id; int mclk_coeff; diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 8e45aa9..5dfdf6e 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -716,8 +716,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u32 dai_cfg1; u32 fs, bypass; diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 4624e75..85d9cab 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -164,8 +164,7 @@ static int jz4740_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { uint32_t val; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; switch (params_rate(params)) { case 8000: diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 744063d..89bbe03 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1140,8 +1140,7 @@ static int lm49453_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); u16 clk_div = 0; diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 20c324c..95147e4 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1361,8 +1361,7 @@ static int get_coeff(int mclk, int rate, int timesofbclk) static int rt5631_hifi_pcm_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); int timesofbclk = 32, coeff; unsigned int iface = 0; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index ce1d5ce..8af6a524 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -665,8 +665,7 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); int channels = params_channels(params); int i2s_ctl = 0; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index de2b2054..4c94fd2 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -254,8 +254,7 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, SSM2602_IFACE) & 0xfff3; int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params)); @@ -291,8 +290,7 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, static int ssm2602_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); struct snd_pcm_runtime *master_runtime; @@ -328,8 +326,7 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, static void ssm2602_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); if (ssm2602->master_substream == substream) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 7db6fa5..8d717f4 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -609,8 +609,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); unsigned int rate; int i, mcs = -1, ir = -1; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index b0cafd3..8c758b21 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -323,8 +323,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface_reg; int ret; struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); @@ -369,8 +368,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* set active */ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001); @@ -381,8 +379,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); /* deactivate */ diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 802064b..85944e9 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -126,8 +126,7 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); int fsref, divisor, wlen, pval, jval, dval, qval; u16 reg; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 8d20f6e..bde2d1a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -802,8 +802,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 4587ddd..0dd41077 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -62,8 +62,10 @@ #define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \ (((samples)*5000) / (((burstrate)*5000) / ((burstrate) - (playrate)))) -static void dac33_calculate_times(struct snd_pcm_substream *substream); -static int dac33_prepare_chip(struct snd_pcm_substream *substream); +static void dac33_calculate_times(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec); +static int dac33_prepare_chip(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec); enum dac33_state { DAC33_IDLE = 0, @@ -427,8 +429,8 @@ static int dac33_playback_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (likely(dac33->substream)) { - dac33_calculate_times(dac33->substream); - dac33_prepare_chip(dac33->substream); + dac33_calculate_times(dac33->substream, w->codec); + dac33_prepare_chip(dac33->substream, w->codec); } break; case SND_SOC_DAPM_POST_PMD: @@ -799,8 +801,7 @@ static void dac33_oscwait(struct snd_soc_codec *codec) static int dac33_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); /* Stream started, save the substream pointer */ @@ -812,8 +813,7 @@ static int dac33_startup(struct snd_pcm_substream *substream, static void dac33_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); dac33->substream = NULL; @@ -825,8 +825,7 @@ static int dac33_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); /* Check parameters for validity */ @@ -868,10 +867,9 @@ static int dac33_hw_params(struct snd_pcm_substream *substream, * writes happens in different order, than dac33 might end up in unknown state. * Use the known, working sequence of register writes to initialize the dac33. */ -static int dac33_prepare_chip(struct snd_pcm_substream *substream) +static int dac33_prepare_chip(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned int oscset, ratioset, pwr_ctrl, reg_tmp; u8 aictrl_a, aictrl_b, fifoctrl_a; @@ -1067,10 +1065,9 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) return 0; } -static void dac33_calculate_times(struct snd_pcm_substream *substream) +static void dac33_calculate_times(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned int period_size = substream->runtime->period_size; unsigned int rate = substream->runtime->rate; @@ -1128,8 +1125,7 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); int ret = 0; @@ -1161,8 +1157,7 @@ static snd_pcm_sframes_t dac33_dai_delay( struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned long long t0, t1, t_now; unsigned int time_delta, uthr; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 170cf9a..391fcfc 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1685,8 +1685,7 @@ static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction, static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); if (twl4030->master_substream) { @@ -1715,8 +1714,7 @@ static int twl4030_startup(struct snd_pcm_substream *substream, static void twl4030_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); if (twl4030->master_substream == substream) @@ -1740,8 +1738,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 mode, old_mode, format, old_format; @@ -1974,8 +1971,7 @@ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, static int twl4030_voice_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 mode; @@ -2007,8 +2003,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* Enable voice digital filters */ twl4030_voice_enable(codec, substream->stream, 0); @@ -2017,8 +2012,7 @@ static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 old_mode, mode; diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 2d8c6b8..ccbc88a 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1340,8 +1340,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, static int twl6040_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); snd_pcm_hw_constraint_list(substream->runtime, 0, @@ -1355,8 +1354,7 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int rate; @@ -1392,8 +1390,7 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, static int twl6040_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040 *twl6040 = codec->control_data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int ret; diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 797b0dd..6c3d43b 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -159,8 +159,7 @@ static int uda134x_mute(struct snd_soc_dai *dai, int mute) static int uda134x_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); struct snd_pcm_runtime *master_runtime; @@ -191,8 +190,7 @@ static int uda134x_startup(struct snd_pcm_substream *substream, static void uda134x_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); if (uda134x->master_substream == substream) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 4f1b23d..2502214 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -502,8 +502,7 @@ static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai, static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda1380_priv *uda1380 = snd_soc_codec_get_drvdata(codec); int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER); @@ -528,8 +527,7 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* set WSPLL power and divider if running from this clock */ diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 3d868dc..7b24d6d 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -293,8 +293,7 @@ static const struct snd_kcontrol_new wl1273_controls[] = { static int wl1273_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); switch (wl1273->mode) { @@ -329,8 +328,7 @@ static int wl1273_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(rtd->codec); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(dai->codec); struct wl1273_core *core = wl1273->core; unsigned int rate, width, r; diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 898979d..fcc0cac 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1145,8 +1145,7 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1); audio1 &= ~WM8400_AIF_WL_MASK; diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 9166126..56a0495 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -392,8 +392,7 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface = snd_soc_read(codec, WM8510_IFACE) & 0x19f; u16 adn = snd_soc_read(codec, WM8510_ADD) & 0x1f1; diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 7fea2c3..1c3ffb2 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -145,8 +145,7 @@ static int wm8523_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); int i; u16 aifctrl1 = snd_soc_read(codec, WM8523_AIF_CTRL1); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index fc3d59e..1467f97 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -88,8 +88,7 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 dac = snd_soc_read(codec, WM8728_DACCTL); dac &= ~0x18; diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 4fe9d19..d052012 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -329,8 +329,7 @@ static int wm8737_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); int i; u16 clocking = 0; diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 3941f50..6e849cb 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -203,8 +203,7 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC; int i; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index e4c50ce..89151ca 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -547,8 +547,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8750_priv *wm8750 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8750_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8750_SRATE) & 0x1c0; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index e27e7b6..a26482c 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -931,8 +931,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 voice = snd_soc_read(codec, WM8753_PCM) & 0x01f3; u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x017f; @@ -1161,8 +1160,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x01c0; u16 hifi = snd_soc_read(codec, WM8753_HIFI) & 0x01f3; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index f18c554..077c962 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -610,8 +610,7 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 reg; reg = snd_soc_read(codec, WM8900_REG_AUDIO1) & ~0x60; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 302c76a..8365b43 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1453,8 +1453,7 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); int fs = params_rate(params); int bclk; diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index d2883af..481a3d9 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -371,8 +371,7 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface = snd_soc_read(codec, WM8940_IFACE) & 0xFD9F; u16 addcntrl = snd_soc_read(codec, WM8940_ADDCNTRL) & 0xFFF1; u16 companding = snd_soc_read(codec, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 840d720..8bc659d 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -505,8 +505,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; int i; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 7dba006..28a0fe2 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2534,8 +2534,7 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int i; int aif0 = 0; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 28fe59e..eef783f 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -478,8 +478,7 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8971_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8971_SRATE) & 0x1c0; diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 72d5fdc..a5be3ad 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -723,8 +723,7 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec); /* Word length mask = 0x60 */ u16 iface_ctl = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x60; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 6cdf6a2b..1d4c5cf 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -668,8 +668,7 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8988_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8988_SRATE) & 0x180; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 9d24235..db63c97 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1112,8 +1112,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 audio1 = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_1); audio1 &= ~WM8990_AIF_WL_MASK; diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index cacc6a8..7c09593 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -236,9 +236,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg; u16 vra; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index b342ae5..2603863 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -467,9 +467,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg; u16 vra; @@ -487,9 +485,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, static int ac97_aux_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 vra, xsle; vra = ac97_read(codec, AC97_EXTENDED_STATUS); -- 1.7.9.1

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[alsa-devel] [PATCH] ASoC: tlv320aic23: Remove driver-specific version number
by Mark Brown 04 Apr '12

04 Apr '12

It's never been updated since the driver was merged. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320aic23.c | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 16d55f9..b0cafd3 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -34,8 +34,6 @@ #include "tlv320aic23.h" -#define AIC23_VERSION "0.1" - /* * AIC23 register cache */ @@ -548,8 +546,6 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); int ret; - printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); - ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); -- 1.7.9.1

1 0

Re: [alsa-devel] trying to get an eleven rack to work
by David Alexander 04 Apr '12

04 Apr '12

Grant, I had an entry using the USB_DEVICE which is very similar to the one you suggested. I changed it over to the USB_DEVICE_VENDOR_SPEC you had listed. I am getting: [ 60.952067] usb 2-4: new high speed USB device number 2 using ehci_hcd [ 61.468413] ALSA mixer.c:1938 usbaudio: unit 32: unexpected type 0x09 [ 61.468433] snd-usb-audio: probe of 2-4:1.1 failed with error -5 [ 61.469257] ALSA mixer.c:1938 usbaudio: unit 32: unexpected type 0x09 [ 61.469266] snd-usb-audio: probe of 2-4:1.3 failed with error -5 [ 61.469280] ALSA mixer.c:1938 usbaudio: unit 32: unexpected type 0x09 [ 61.469285] snd-usb-audio: probe of 2-4:1.4 failed with error -5 [ 61.469305] usbcore: registered new interface driver snd-usb-audio and I have entries in /proc/asound as before: ls -l total 0 -r--r--r-- 1 root root 0 2012-04-02 13:47 id -r--r--r-- 1 root root 0 2012-04-02 13:47 midi0 -r--r--r-- 1 root root 0 2012-04-02 13:47 usbbus -r--r--r-- 1 root root 0 2012-04-02 13:47 usbid Those entries had disappeared after making the changed to mixer.c. It didn't seem to be the magic bullet though, as I need some sort of pcm entries to be created? What is the difference between the usb_device and usb_device_vendor_spec? Any other ideas? thanks, David On Fri, Mar 30, 2012 at 5:02 PM, Grant Diffey <gdiffey(a)gmail.com> wrote: > > On Sat, Mar 31, 2012 at 4:27 AM, David Alexander <lidxv0n(a)gmail.com> wrote: >> >> Grant, >> >> You had mentioned something about the end point for the mixer. I have >> found this information from the trace I had performed: >> >> From a dispatch urb_function_get_descriptor_from_device message under >> the fid_USBPORT_Endpoint_Descriptor: >> >> fid_bLength 7 >> fid_bDescriptortype 5 >> fid_bEndpointAddress 0 >> fid_bmAttributes 0 >> fid_wMaxPacketSize 64 >> fid_bInterval 0 >> >> Is this what you were asking about, and does it help us? >> >> thanks, >> David >> >> >> > > What does the quirk look like that you have now? because what I wrote was > wrong. > > it seems I can't count to 5 :) > > so based on what Clemens said and me re-reading the interface descriptor > > Interface 0 is dfu upload > Interface 1 is a mixer > > Interface 2 is the midi > Interface 3 is audio output > Interface 4 is audio Input and provides clock data for output > > 0 and 2 were previously recognised so do not need to be quirked. > > so I think this is what the quirk should look like: > > > USB_DEVICE_VENDOR_SPEC(0x0dba, 0xb011), > > .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { > .vendor_name = "Digidesign", > .product_name = "Eleven Rack", > .ifnum = QUIRK_ANY_INTERFACE, > .type = QUIRK_COMPOSITE, > .data = & (const struct snd_usb_audio_quirk[]) { > > { > .ifnum = 1, > .type = QUIRK_AUDIO_STANDARD_MIXER > }, > { > .ifnum = 3, > .type = QUIRK_AUDIO_STANDARD_INTERFACE > > }, > { > .ifnum = 4, > .type = QUIRK_AUDIO_STANDARD_INTERFACE > }, > { > .ifnum = -1 > } > } > } > } > > is that similar to what you have? > > Grant. >

3 5

[alsa-devel] [PATCH 1/2] ASoC: fsl: remove redundant Kconfig option SND_SOC_FSL_SSI
by Shawn Guo 04 Apr '12

04 Apr '12

While commit 606d620 (ASoC: imx: merge sound/soc/imx into sound/soc/fsl) adds SND_SOC_FSL_SSI outside "menuconfig SND_POWERPC_SOC" to make it visible for both PowerPC and ARM/IMX, it forgot removing the one inside "menuconfig SND_POWERPC_SOC". Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/fsl/Kconfig | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index aa14fc9..d23d612 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -16,10 +16,6 @@ if SND_POWERPC_SOC config SND_MPC52xx_DMA tristate -config SND_SOC_FSL_SSI - tristate - depends on FSL_SOC - config SND_SOC_POWERPC_DMA tristate depends on FSL_SOC -- 1.7.5.4

2 2

[alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix warning due to the lack of REGULATOR_CHANGE_VOLTAGE
by Fabio Estevam 04 Apr '12

04 Apr '12

Fix the following warning during kernel boot: 0-000a: 850 <--> 1600 mV at 1200 mV normal 0-000a: Voltage range but no REGULATOR_CHANGE_VOLTAGE Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: Keep the voltage constant sound/soc/codecs/sgtl5000.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8e92fb8..431c9c1 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -84,8 +84,8 @@ static struct regulator_consumer_supply ldo_consumer[] = { static struct regulator_init_data ldo_init_data = { .constraints = { - .min_uV = 850000, - .max_uV = 1600000, + .min_uV = 1200000, + .max_uV = 1200000, .valid_modes_mask = REGULATOR_MODE_NORMAL, .valid_ops_mask = REGULATOR_CHANGE_STATUS, }, -- 1.7.1

2 1