September 2011 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH] ASoC: Remove unneeded mutex_init in wl1273_probe()
by Mark Brown 23 Sep '11

23 Sep '11

On Fri, Sep 23, 2011 at 01:10:57PM +0800, Axel Lin wrote: > Since f0fba2ad "ASoC: multi-component - ASoC Multi-Component Support", > snd_soc_register_codec() now does all the codec list and mutex init. > Thus don't need to call mutex_init(&codec->mutex) in wl1273_probe() any more. Applied, thanks.

1 0

[alsa-devel] [PATCH, resend] ASoC: omap-mcbsp: Do not attempt to change DAI sysclk if stream is active
by Jarkko Nikula 23 Sep '11

23 Sep '11

Attempt to change McBSP CLKS source while another stream is active is not safe after commit d135865 ("OMAP: McBSP: implement functional clock switching via clock framework") in 2.6.37. CLKS parent clock switching using clock framework have to idle the McBSP before switching and then activate it again. This short break can cause a DMA transaction error to already running stream which halts and recovers only by closing and restarting the stream. This goes more fatal after commit e2fa61d ("OMAP3: l3: Introduce l3-interconnect error handling driver") in 2.6.39 where l3 driver detects a severe timeout error and does BUG_ON(). Fix this by not changing any configuration in omap_mcbsp_dai_set_dai_sysclk if the McBSP is already active. This test should have been here just from the beginning anyway. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: stable(a)kernel.org Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- For 2.6.37 and onward. Mark: I didn't add your acked-by even you acked this earlier: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-June/040717.html Just in case if this go through your signed-off during kernel.org hazzle. --- sound/soc/omap/omap-mcbsp.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index ebcc2d4..478d607 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -516,6 +516,12 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; + if (mcbsp_data->active) + if (freq == mcbsp_data->in_freq) + return 0; + else + return -EBUSY; + /* The McBSP signal muxing functions are only available on McBSP1 */ if (clk_id == OMAP_MCBSP_CLKR_SRC_CLKR || clk_id == OMAP_MCBSP_CLKR_SRC_CLKX || -- 1.7.5.4

2 1

Re: [alsa-devel] [PATCH] ASoC: Add driver for rt5631
by Mark Brown 23 Sep '11

23 Sep '11

On Thu, Sep 22, 2011 at 09:48:55AM +0800, johnnyhsu wrote: > Besides, could you show me the hint or address where I can post questions > about driver? I wonder how to work between modem and codec and listen to > others' experience. The alsa-devel list is the place for discussions like that.

1 0

[alsa-devel] [PATCHv2] ASoC: omap-mcbsp: Do not attempt to change DAI sysclk if stream is active
by Jarkko Nikula 23 Sep '11

23 Sep '11

Attempt to change McBSP CLKS source while another stream is active is not safe after commit d135865 ("OMAP: McBSP: implement functional clock switching via clock framework") in 2.6.37. CLKS parent clock switching using clock framework have to idle the McBSP before switching and then activate it again. This short break can cause a DMA transaction error to already running stream which halts and recovers only by closing and restarting the stream. This goes more fatal after commit e2fa61d ("OMAP3: l3: Introduce l3-interconnect error handling driver") in 2.6.39 where l3 driver detects a severe timeout error and does BUG_ON(). Fix this by not changing any configuration in omap_mcbsp_dai_set_dai_sysclk if the McBSP is already active. This test should have been here just from the beginning anyway. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: stable(a)kernel.org --- v2: - Returns -EBUSY from omap_mcbsp_dai_set_dai_sysclk if the input clock rate doesn't match with another stream's configuration and 0 otherwise. For 2.6.37 and onward. This issue is valid to those machine drivers that use the McBSP as a DAI link master and are configuring the CLKS source (like sound/soc/omap/omap3pandora.c in mainline). --- sound/soc/omap/omap-mcbsp.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 07b7723..4b82290 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -516,6 +516,12 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; + if (mcbsp_data->active) + if (freq == mcbsp_data->in_freq) + return 0; + else + return -EBUSY; + /* The McBSP signal muxing functions are only available on McBSP1 */ if (clk_id == OMAP_MCBSP_CLKR_SRC_CLKR || clk_id == OMAP_MCBSP_CLKR_SRC_CLKX || -- 1.7.4.4

4 5

[alsa-devel] snd_pcm_hw_rule_noresample
by Raymond Yau 23 Sep '11

23 Sep '11

60fdc82f96bff2e3a31e9fcf8a9503f5a37c1f85 ALSA: pcm: add snd_pcm_hw_rule_noresample() Add a helper function to allow drivers to disable hardware resampling when the application has specified the SNDRV_PCM_HW_PARAMS_NORESAMPLE flag. Refer to commit and the three corresponding patch with ymfpci, emu10k1, via82xx, it seem to be a regression according to http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___h_w___params.… int snd_pcm_hw_params_set_rate_resample ( snd_pcm_t * pcm, snd_pcm_hw_params_t * params, unsigned int val ) Restrict a configuration space to contain only real hardware rates. ymfpci and emu10k1 have SNDRV_PCM_RATE_CONTINUOUS and support 8000Hz to 48000Hz after this patch and if application using snd_pcm_hw_params_set_rate_resample(,pcm , params ,0) why the supported rate is only 48000Hz ? but snd_pcm_hw_params_get_rate_min() is still 8000Hz This patch won't affect the accuracy of the playback position of the ymfpci attach a test program which print the the playback position and the result ./alsa_test hw:0 Hardware PCM card 0 'Yamaha DS-1 (YMF724F)' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16383 period_size : 5461 period_time : 123832 tstamp_mode : ENABLE period_step : 1 avail_min : 5461 period_event : 0 start_threshold : 1 stop_threshold : 16383 silence_threshold: 0 silence_size : 0 boundary : 2147352576 appl_ptr : 0 hw_ptr : 0 pcm write 5461 pcm write 5461 pcm write 5461 0 0 1 0 2 0 3 0 4 0 5 0 6 0 7 0 8 0 9 0 10 0 11 235 12 235 13 235 14 235 15 235 16 470 17 470 18 470 19 470 20 470 21 705 22 705 23 705 24 705 25 705 26 940 27 940 28 940 29 940 30 940 31 940 32 1175 33 1175 34 1175 35 1175 36 1175 37 1411 38 1411 39 1411 40 1411 41 1411 42 1646 43 1646 44 1646 45 1646 46 1646 47 1646 48 1881 49 1881 50 1881 51 1881 52 1881 53 2116 54 2116 55 2116 56 2116 57 2116 58 2351 59 2351 60 2351 61 2351 62 2351 63 2587 64 2587 65 2587 66 2587 67 2587 68 2587 69 2822 70 2822 71 2822 72 2822 73 2822 74 3057 75 3057 76 3057 77 3057 78 3057 79 3292 80 3292 81 3292 82 3292 83 3292 84 3527 85 3527 86 3527 87 3527 88 3527 89 3527 90 3763 91 3763 92 3763 93 3763 94 3763 95 3998 96 3998 97 3998 98 3998 99 3998 100 4233 101 4233 102 4233 103 4233 104 4233 105 4233 106 4468 107 4468 108 4468 109 4468 110 4468 111 4703 112 4703 113 4703 114 4703 115 4703 116 4939 117 4939 118 4939 119 4939 120 4939 121 5174 122 5174 123 5174 124 5174 125 5174 126 5174 127 5409 128 5409 129 5409 130 5409 131 5409 132 5644 133 5644 134 5644 135 5644 136 5644 137 5879 138 5879 139 5879 140 5879 141 5879 142 6115 143 6115 144 6115 145 6115 146 6115 147 6115 148 6350 149 6350 150 6350 151 6350 152 6350 153 6585 154 6585 155 6585 156 6585 157 6585 158 6820 159 6820 160 6820 161 6820 162 6820 163 6820 164 7055 165 7055 166 7055 167 7055 168 7055 169 7291 170 7291 171 7291 172 7291 173 7291 174 7526 175 7526 176 7526 177 7526 178 7526 179 7761 180 7761 181 7761 182 7761 183 7761 184 7761 185 7996 186 7996 187 7996 188 7996 189 7996 190 8231 191 8231 192 8231 193 8231 194 8231 195 8467 196 8467 197 8467 198 8467 199 8467 200 8702 201 8702 202 8702 203 8702 204 8702 205 8702 206 8937 207 8937 208 8937 209 8937 210 8937 211 9172 212 9172 213 9172 214 9172 215 9172 216 9407 217 9407 218 9407 219 9407 220 9407 221 9407 222 9643 223 9643 224 9643 225 9643 226 9643 227 9878 228 9878 229 9878 230 9878 231 9878 232 10113 233 10113 234 10113 235 10113 236 10113 237 10348 238 10348 239 10348 240 10348 241 10348 242 10348 243 10583 244 10583 245 10583 246 10583 247 10583 248 10819 249 10819 250 10819 251 10819 252 10819 253 11054 254 11054 255 11054 256 11054 257 11054 258 11289 259 11289 260 11289 261 11289 262 11289 263 11289 264 11524 265 11524 266 11524 267 11524 268 11524 269 11759 270 11759 271 11759 272 11759 273 11759 274 11995 275 11995 276 11995 277 11995 278 11995 279 11995 280 12230 281 12230 282 12230 283 12230 284 12230 285 12465 286 12465 287 12465 288 12465 289 12465 290 12700 291 12700 292 12700 293 12700 294 12700 295 12935 296 12935 297 12935 298 12935 299 12935 300 12935 301 13171 302 13171 303 13171 304 13171 305 13171 306 13406 307 13406 308 13406 309 13406 310 13406 311 13641 312 13641 313 13641 314 13641 315 13641 316 13876 317 13876 318 13876 319 13876 320 13876 321 13876 322 14111 323 14111 324 14111 325 14111 326 14111 327 14347 328 14347 329 14347 330 14347 331 14347 332 14582 333 14582 334 14582 335 14582 336 14582 337 14582 338 14817 339 14817 340 14817 341 14817 342 14817 343 15052 344 15052 345 15052 346 15052 347 15052 348 15287 349 15287 350 15287 351 15287 352 15287 353 15523 354 15523 355 15523 356 15523 357 15523 358 15523 359 15758 360 15758 361 15758 362 15758 363 15758 364 15993 365 15993 366 15993 367 15993 368 15993 369 16228 370 16228 371 16228 372 16228 373 16228 374 -32

2 4

Re: [alsa-devel] [patch 2/2] ALSA: hdspm - cleanup __user tags in ioctl()
by Takashi Iwai 23 Sep '11

23 Sep '11

At Fri, 23 Sep 2011 09:25:05 +0300, Dan Carpenter wrote: > > This makes the code cleaner and silences a Sparse complaint: > sound/pci/rme9652/hdspm.c:6341:23: warning: incorrect type in assignment (incompatible argument 4 (different address spaces)) > sound/pci/rme9652/hdspm.c:6341:23: expected int ( *ioctl )( ... ) > sound/pci/rme9652/hdspm.c:6341:23: got int ( static [toplevel] *<noident> )( ... ) > sound/pci/rme9652/hdspm.c:6102:44: warning: dereference of noderef expression > sound/pci/rme9652/hdspm.c:6225:50: warning: dereference of noderef expression > sound/pci/rme9652/hdspm.c:6264:50: warning: dereference of noderef expression > sound/pci/rme9652/hdspm.c:6283:50: warning: dereference of noderef expression > sound/pci/rme9652/hdspm.c:6289:59: warning: dereference of noderef expression > > Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Applied now. Thanks. Takashi > diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c > index bf438d1..6e2f7ef 100644 > --- a/sound/pci/rme9652/hdspm.c > +++ b/sound/pci/rme9652/hdspm.c > @@ -6097,7 +6097,7 @@ static inline int copy_u32_le(void __user *dest, void __iomem *src) > } > > static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, > - unsigned int cmd, unsigned long __user arg) > + unsigned int cmd, unsigned long arg) > { > void __user *argp = (void __user *)arg; > struct hdspm *hdspm = hw->private_data; > @@ -6222,7 +6222,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, > info.line_out = hdspm_line_out(hdspm); > info.passthru = 0; > spin_unlock_irq(&hdspm->lock); > - if (copy_to_user((void __user *) arg, &info, sizeof(info))) > + if (copy_to_user(argp, &info, sizeof(info))) > return -EFAULT; > break; > > @@ -6261,7 +6261,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, > break; > } > > - if (copy_to_user((void __user *) arg, &status, sizeof(status))) > + if (copy_to_user(argp, &status, sizeof(status))) > return -EFAULT; > > > @@ -6280,13 +6280,13 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, > if (hdspm->tco) > hdspm_version.addons |= HDSPM_ADDON_TCO; > > - if (copy_to_user((void __user *) arg, &hdspm_version, > + if (copy_to_user(argp, &hdspm_version, > sizeof(hdspm_version))) > return -EFAULT; > break; > > case SNDRV_HDSPM_IOCTL_GET_MIXER: > - if (copy_from_user(&mixer, (void __user *)arg, sizeof(mixer))) > + if (copy_from_user(&mixer, argp, sizeof(mixer))) > return -EFAULT; > if (copy_to_user((void __user *)mixer.mixer, hdspm->mixer, > sizeof(struct hdspm_mixer))) >

1 0

Re: [alsa-devel] [patch 1/2] ALSA: hdspm - potential info leak in snd_hdspm_hwdep_ioctl()
by Takashi Iwai 23 Sep '11

23 Sep '11

At Fri, 23 Sep 2011 09:24:21 +0300, Dan Carpenter wrote: > > Smatch has a new check for Rosenberg type information leaks where > structs are copied to the user with uninitialized stack data in them. > > The status struct has a hole in it, and on some paths not all the > members were initialized. > > struct hdspm_status { > unsigned char card_type; /* 0 1 */ > /* XXX 3 bytes hole, try to pack */ > enum hdspm_syncsource autosync_source; /* 4 4 */ > long long unsigned int card_clock; /* 8 8 */ > > The hdspm_version struct had holes in it as well. > > struct hdspm_version { > unsigned char card_type; /* 0 1 */ > char cardname[20]; /* 1 20 */ > /* XXX 3 bytes hole, try to pack */ > unsigned int serial; /* 24 4 */ > short unsigned int firmware_rev; /* 28 2 */ > /* XXX 2 bytes hole, try to pack */ > int addons; /* 32 4 */ > > Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Applied now. Thanks. Takashi > diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c > index 214110d..bf438d1 100644 > --- a/sound/pci/rme9652/hdspm.c > +++ b/sound/pci/rme9652/hdspm.c > @@ -6227,6 +6227,8 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, > break; > > case SNDRV_HDSPM_IOCTL_GET_STATUS: > + memset(&status, 0, sizeof(status)); > + > status.card_type = hdspm->io_type; > > status.autosync_source = hdspm_autosync_ref(hdspm); > @@ -6266,6 +6268,8 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, > break; > > case SNDRV_HDSPM_IOCTL_GET_VERSION: > + memset(&hdspm_version, 0, sizeof(hdspm_version)); > + > hdspm_version.card_type = hdspm->io_type; > strncpy(hdspm_version.cardname, hdspm->card_name, > sizeof(hdspm_version.cardname)); >

1 0

[alsa-devel] ASoC trees while master is down
by Mark Brown 23 Sep '11

23 Sep '11

Due to the continued outage of kernel.org I'm currently publishing my for-whatever branches on: git://opensource.wolfsonmicro.com/linux-2.6-asoc If you need current code please look there.

2 2

[alsa-devel] [PATCH] ASoC: Add missed BCLK rate to WM5100 driver
by Mark Brown 22 Sep '11

22 Sep '11

Reported-by: Axel Lin <axel.lin(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 443d76d..cb940a8 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1511,6 +1511,7 @@ static int wm5100_bclk_rates_dat[WM5100_NUM_BCLK_RATES] = { 96000, 128000, 192000, + 256000, 384000, 512000, 768000, -- 1.7.6.3

1 0

[alsa-devel] [PATCH v2 00/10] ASoC: twl6040: Cleanups and fixes
by Peter Ujfalusi 22 Sep '11

22 Sep '11

Hello, Changes since v1 [1]: - Left out patches already applied by Mark - Description on patch 01 changed to describe more precisely the need to lower the chip default gains after power on. - Extended the commit message for patch 02 to explain the change. - Added Acked-by from Liam to the patches. Short summary for v2 series: Code cleanup, and fixes for the twl6040 codec driver. Main changes: - codec initialization changes - Correction for the Earphone path - AUX L/R output support - Headset/Handsfree control name changes - Headset DC offset workaround The series has been created on top of: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-3.2 branch + OMAP McPDM changes [2] [1] first series: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-September/043847.… [2] OMAP McPDM changes: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-September/044172.… Regards, Peter --- Peter Ujfalusi (10): ASoC: twl6040: Lower the power on gain values at startup ASoC: twl6040: Fix comments for register names ASoC: twl6040: Remove strings "NULL" from DAPM route ASoC: twl6040: Introduce SW only shadow register ASoC: twl6040: Earphone path correction ASoC: twl6040: Use consistent names for Handsfree path ASoC: twl6040: Use consistent names for Headset path ASoC: twl6040: Support for AUX L/R output ASoC: twl6040/sdp4430: Change legacy DAI name ASoC/MFD: twl6040: Combine bit definitions for Headset control registers include/linux/mfd/twl6040.h | 13 +-- sound/soc/codecs/twl6040.c | 223 ++++++++++++++++++++++++++----------------- sound/soc/omap/sdp4430.c | 2 +- 3 files changed, 140 insertions(+), 98 deletions(-) -- 1.7.6.1

2 11