December 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/1] ASoC: mop500: Deletion of unnecessary checks before the function call "of_node_put"
by SF Markus Elfring 03 Dec '14

03 Dec '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 2 Dec 2014 17:15:11 +0100 The of_node_put() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/ux500/mop500.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 7f7093b..4e0c0e5 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -63,10 +63,8 @@ static void mop500_of_node_put(void) int i; for (i = 0; i < 2; i++) { - if (mop500_dai_links[i].cpu_of_node) - of_node_put(mop500_dai_links[i].cpu_of_node); - if (mop500_dai_links[i].codec_of_node) - of_node_put(mop500_dai_links[i].codec_of_node); + of_node_put(mop500_dai_links[i].cpu_of_node); + of_node_put(mop500_dai_links[i].codec_of_node); } } -- 2.1.3

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[alsa-devel] [PATCH 1/1] ASoC: fsi: Deletion of unnecessary checks before the function call "clk_disable"
by SF Markus Elfring 03 Dec '14

03 Dec '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 2 Dec 2014 14:34:30 +0100 The clk_disable() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/sh/fsi.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 88e5df4..8869971 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -842,12 +842,9 @@ static int fsi_clk_disable(struct device *dev, return -EINVAL; if (1 == clock->count--) { - if (clock->xck) - clk_disable(clock->xck); - if (clock->ick) - clk_disable(clock->ick); - if (clock->div) - clk_disable(clock->div); + clk_disable(clock->xck); + clk_disable(clock->ick); + clk_disable(clock->div); } return 0; -- 2.1.3

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[alsa-devel] Analog sound (ALC262) lost on linux (any distro and kernel) on a dual boot system
by Ploumistos Alexandros 03 Dec '14

03 Dec '14

Hello, A while back I had installed Ubuntu on a friend's laptop -a Sony Vaio VGN-FW21M- alongside windows Vista. Everything worked fine for a few years, until at some point last year linux went mute, while in windows both the analog and the digital outputs continued working. She couldn't remember doing something "out of the ordinary", but she was sure it was not after an update. Several months later, she upgraded to windows 7 but that didn't change anything. When I got my hands on the laptop, I checked for muted controls and other usual suspects, deprecated quirks (there were none) and all inputs and outputs of the system. HDMI audio works fine and so does the internal mic. Speakers, headphones and the external mic input do not work, even though pulse audio volume meter acts as if there were actually sounds being played. I followed all of Ubuntu's troubleshooting procedures, I removed and reinstalled alsa and pulse, I purged all configuration files, I created a new user and I even installed the latest alsa-packages from the "ALSA daily build snapshots" repository and nothing worked. I also tried booting into windows, setting all volume levels to the max and disabled power saving on all sound and multimedia devices. Next, I tried retasking the jacks one by one to no avail. I remembered that I had performed the installation with an Ubuntu 11.10 live CD, so I booted the computer off of that, but this time I got no sound. I tried with all the live CDs of a multitude of distros I had lying around, with kernels ranging from 2.6.35 to 3.16.6 and none of them worked, so I figured the issue must have something to do with windows. Is it possible that windows somehow started locking the audio chip on shutdown? If that is the case, can it be deduced from the logs and more importantly, can it be fixed? I am attaching the output of alsa-info and lspci as well as kernel logs. Thank you for your time.

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[alsa-devel] [PATCH 5/5] ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap
by Andrew Jackson 03 Dec '14

03 Dec '14

If the FIFOs aren't flushed, the left/right channels may be swapped: this may occur if the FIFOs are not empty when the streams start. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 083779d..932abfd 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -258,6 +258,7 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, /* Iterate over set of channels - independently controlled. */ do { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + i2s_write_reg(dev->i2s_base, TXFFR, 1); i2s_write_reg(dev->i2s_base, TCR(ch_reg), xfer_resolution); i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); @@ -265,6 +266,7 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); } else { + i2s_write_reg(dev->i2s_base, RXFFR, 1); i2s_write_reg(dev->i2s_base, RCR(ch_reg), xfer_resolution); i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); -- 1.7.1

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[alsa-devel] [PATCH] sound: pci: cmipci.c: Cleaning up function that are not used anywhere
by Rickard Strandqvist 03 Dec '14

03 Dec '14

Remove function snd_cmipci_pll_rmn() that is not used anywhere. This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/cmipci.c | 41 ----------------------------------------- 1 file changed, 41 deletions(-) diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 85ed403..e33edae 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -615,47 +615,6 @@ static unsigned int snd_cmipci_rate_freq(unsigned int rate) #ifdef USE_VAR48KRATE /* - * Determine PLL values for frequency setup, maybe the CMI8338 (CMI8738???) - * does it this way .. maybe not. Never get any information from C-Media about - * that <werner(a)suse.de>. - */ -static int snd_cmipci_pll_rmn(unsigned int rate, unsigned int adcmult, int *r, int *m, int *n) -{ - unsigned int delta, tolerance; - int xm, xn, xr; - - for (*r = 0; rate < CM_MAXIMUM_RATE/adcmult; *r += (1<<5)) - rate <<= 1; - *n = -1; - if (*r > 0xff) - goto out; - tolerance = rate*CM_TOLERANCE_RATE; - - for (xn = (1+2); xn < (0x1f+2); xn++) { - for (xm = (1+2); xm < (0xff+2); xm++) { - xr = ((CM_REFFREQ_XIN/adcmult) * xm) / xn; - - if (xr < rate) - delta = rate - xr; - else - delta = xr - rate; - - /* - * If we found one, remember this, - * and try to find a closer one - */ - if (delta < tolerance) { - tolerance = delta; - *m = xm - 2; - *n = xn - 2; - } - } - } -out: - return (*n > -1); -} - -/* * Program pll register bits, I assume that the 8 registers 0xf8 up to 0xff * are mapped onto the 8 ADC/DAC sampling frequency which can be chosen * at the register CM_REG_FUNCTRL1 (0x04). -- 1.7.10.4

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[alsa-devel] [RFC] alsa-lib: Add snd_pcm_start_at
by Tim Cussins 03 Dec '14

03 Dec '14

snd_pcm_start_at() - alsa-lib The headlines: - Provide userspace snd_pcm_start_at, leverage new ioctl - Provide some decent doxygen guff diff --git a/include/pcm.h b/include/pcm.h index 0655e7f..bbe6e79 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -325,9 +325,10 @@ typedef enum _snd_pcm_tstamp { typedef enum _snd_pcm_tstamp_type { SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /** gettimeofday equivalent */ - SND_PCM_TSTAMP_TYPE_MONOTONIC, /** posix_clock_monotonic equivalent */ + SND_PCM_TSTAMP_TYPE_MONOTONIC, /** posix_clock_monotonic equivalent */ SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /** monotonic_raw (no NTP) */ - SND_PCM_TSTAMP_TYPE_LAST = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, + SND_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK, /** audio hw clock */ + SND_PCM_TSTAMP_TYPE_LAST = SND_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK, } snd_pcm_tstamp_type_t; /** Unsigned frames quantity */ @@ -478,6 +479,7 @@ int snd_pcm_prepare(snd_pcm_t *pcm); int snd_pcm_reset(snd_pcm_t *pcm); int snd_pcm_status(snd_pcm_t *pcm, snd_pcm_status_t *status); int snd_pcm_start(snd_pcm_t *pcm); +int snd_pcm_start_at(snd_pcm_t *pcm, const snd_htimestamp_t* start_time); int snd_pcm_drop(snd_pcm_t *pcm); int snd_pcm_drain(snd_pcm_t *pcm); int snd_pcm_pause(snd_pcm_t *pcm, int enable); diff --git a/include/sound/asound.h b/include/sound/asound.h index 941d32f..bcc49dc 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -467,8 +467,9 @@ struct snd_xfern { enum { SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ - SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ - SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, + SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ + SNDRV_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK, /* audio hw clock */ + SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK, }; /* channel positions */ @@ -549,6 +550,8 @@ enum { #define SNDRV_PCM_IOCTL_READN_FRAMES _IOR('A', 0x53, struct snd_xfern) #define SNDRV_PCM_IOCTL_LINK _IOW('A', 0x60, int) #define SNDRV_PCM_IOCTL_UNLINK _IO('A', 0x61) +#define SNDRV_PCM_IOCTL_START_AT _IOW('A', 0x62, snd_htimestamp_t) + /***************************************************************************** * * diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index baa47c7..61158cb 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1085,6 +1085,38 @@ int snd_pcm_start(snd_pcm_t *pcm) } /** + * \brief Start a PCM at a specified point in the future + * \param pcm PCM handle + * \param start_time Absolute time at which to start the stream + * \return 0 on success otherwise a negative error code + * \retval -ENOSYS operation not supported for the current timestamp type + * \retval -EINVAL timespec invalid + * \retval -ETIME requested start_time cannot be satisfied + * + * This method is non-blocking: It establishes an appropriate timer in the kernel + * that will start the stream on expiry. + * + * The timer is unconditionally cancelled upon any \a attempt to change the stream + * state e.g. drop, drain, start, start_at. + * + * \p start_time is interpreted in the context of the stream snd_pcm_tstamp_type, + * set using \link ::snd_pcm_sw_params_set_tstamp_type() \endlink. + */ +int snd_pcm_start_at(snd_pcm_t *pcm, const snd_htimestamp_t *start_time) +{ + assert(pcm); + assert(start_time); + if (CHECK_SANITY(! pcm->setup)) { + SNDMSG("PCM not set up"); + return -EIO; + } + if (pcm->fast_ops->start_at) { + return pcm->fast_ops->start_at(pcm->fast_op_arg, start_time); + } + return -EINVAL; +} + +/** * \brief Stop a PCM dropping pending frames * \param pcm PCM handle * \return 0 on success otherwise a negative error code @@ -1693,6 +1725,7 @@ static const char *const snd_pcm_tstamp_type_names[] = { TSTAMP_TYPE(GETTIMEOFDAY), TSTAMP_TYPE(MONOTONIC), TSTAMP_TYPE(MONOTONIC_RAW), + TSTAMP_TYPE(AUDIO_WALLCLOCK), }; #endif diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index c34b766..aa0c7ba 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -447,6 +447,11 @@ static int snd_pcm_hw_sw_params(snd_pcm_t *pcm, snd_pcm_sw_params_t * params) hw->mmap_control->avail_min = params->avail_min; return sync_ptr(hw, 0); } + if (params->tstamp_type == SND_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK && + hw->version < SNDRV_PROTOCOL_VERSION(2, 0, 12)) { + SYSMSG("Kernel doesn't support SND_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK"); + return -EINVAL; + } if (params->tstamp_type == SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW && hw->version < SNDRV_PROTOCOL_VERSION(2, 0, 12)) { SYSMSG("Kernel doesn't support SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW"); @@ -620,6 +625,20 @@ static int snd_pcm_hw_start(snd_pcm_t *pcm) return 0; } +static int snd_pcm_hw_start_at(snd_pcm_t *pcm, const snd_htimestamp_t *start_time) +{ + snd_pcm_hw_t *hw = pcm->private_data; + int err; + + sync_ptr(hw, 0); + if (ioctl(hw->fd, SNDRV_PCM_IOCTL_START_AT, start_time) < 0) { + err = -errno; + SYSMSG("SNDRV_PCM_IOCTL_START_AT failed (%i)", err); + return err; + } + return 0; +} + static int snd_pcm_hw_drop(snd_pcm_t *pcm) { snd_pcm_hw_t *hw = pcm->private_data; @@ -1336,6 +1355,7 @@ static const snd_pcm_fast_ops_t snd_pcm_hw_fast_ops = { .prepare = snd_pcm_hw_prepare, .reset = snd_pcm_hw_reset, .start = snd_pcm_hw_start, + .start_at = snd_pcm_hw_start_at, .drop = snd_pcm_hw_drop, .drain = snd_pcm_hw_drain, .pause = snd_pcm_hw_pause, diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 394505f..6829750 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -154,6 +154,7 @@ typedef struct { int (*prepare)(snd_pcm_t *pcm); int (*reset)(snd_pcm_t *pcm); int (*start)(snd_pcm_t *pcm); + int (*start_at)(snd_pcm_t *pcm, const snd_htimestamp_t *start_time); int (*drop)(snd_pcm_t *pcm); int (*drain)(snd_pcm_t *pcm); int (*pause)(snd_pcm_t *pcm, int enable); -- 1.7.10.4

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[alsa-devel] [RFC] kernel: Add snd_pcm_start_at
by Tim Cussins 03 Dec '14

03 Dec '14

snd_pcm_start_at() - kernel-side The headlines: - Provide snd_pcm_start_at ioctl - Add SND_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK - Use the stream tstamp_type to interpret the start_at timespec - snd_pcm_start_at_ops defines handler for a TSTAMP_TYPE - handler requirements clearly defined - If posix clock, use a high-res timer implementation (thx Nick Stoughton!) - If audio wallclock, delegate to the pcm driver if possible, otherwise error - Further implementations easily added - Cancel pending timer on: - Subsequent call to snd_pcm_start_at() - Any /attempt/ to change the stream state Major points for discussion: - snd_pcm_start_at_ops requirements aren't clear enough - callback must not do cleanup - cancel is guaranteed to be called - do cleanup here - snd_pcm_gettime() can't currently return a value when tstamp_type is AUDIO_WALLCLOCK - start_at timer callbacks *should* probably lock the stream, contrary to Nick's note - The user story for start_at cancellation from userspace is a bit weird... - Cancellation on state-change seems necessary, but not sufficient - Maybe add snd_pcm_start_at_cancel()? Future thoughts: - Perhaps snd_pcm_start_at should be a blocking call (EAGAIN when stream is non-blocking) - Maybe tstamp_types and startat_types should be different things I've become convinced that exposing the nature of the pcm clock is troublesome, and unnecessary, and fell back to Takashi's suggestion of a DEVICE_SPECIFIC tstamp_type. I note that AUDIO_WALLCLOCK seems to capture the concept of 'pcm time', so I've called it that. Because I'm using SND_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK instead of SNDRV_PCM_TSTAMP_TYPE_PTP, there's no additional payload required for snd_pcm_sw_set_tstamp_type(). Comments and suggestions welcome :) Tim diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 1e7f74a..b8bfa9b 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -82,6 +82,8 @@ struct snd_pcm_ops { unsigned long offset); int (*mmap)(struct snd_pcm_substream *substream, struct vm_area_struct *vma); int (*ack)(struct snd_pcm_substream *substream); + int (*wall_clock_start_at)(struct snd_pcm_substream *substream, const struct timespec *ts); + int (*wall_clock_start_at_cancel)(struct snd_pcm_substream *substream); }; /* @@ -210,6 +212,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE #endif +#define SNDRV_PCM_START_AT_TSTAMP_TYPE_NULL (SNDRV_PCM_TSTAMP_TYPE_LAST+1) + struct snd_pcm_file { struct snd_pcm_substream *substream; int no_compat_mmap; @@ -364,6 +368,10 @@ struct snd_pcm_runtime { #ifdef CONFIG_SND_PCM_XRUN_DEBUG struct snd_pcm_hwptr_log *hwptr_log; #endif + + int start_at_tstamp_type; /* start_at timer tstamp_type Set to + SNDRV_PCM_START_AT_TSTAMP_TYPE_NULL if not active */ + void *start_at_timer_data; /* start_at timer data */ }; struct snd_pcm_group { /* keep linked substreams */ @@ -1069,6 +1077,7 @@ static inline void snd_pcm_gettime(struct snd_pcm_runtime *runtime, case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW: getrawmonotonic(tv); break; + case SNDRV_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK: default: getnstimeofday(tv); break; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 941d32f..dfe2bac 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -467,8 +467,9 @@ struct snd_xfern { enum { SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ - SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ - SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, + SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ + SNDRV_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK, /* audio wallclock timestamp */ + SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK, }; /* channel positions */ @@ -549,6 +550,7 @@ enum { #define SNDRV_PCM_IOCTL_READN_FRAMES _IOR('A', 0x53, struct snd_xfern) #define SNDRV_PCM_IOCTL_LINK _IOW('A', 0x60, int) #define SNDRV_PCM_IOCTL_UNLINK _IO('A', 0x61) +#define SNDRV_PCM_IOCTL_START_AT _IOW('A', 0x62, struct timespec) /***************************************************************************** * * diff --git a/sound/core/pcm.c b/sound/core/pcm.c index cfc56c8..483f85d 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1003,6 +1003,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, init_waitqueue_head(&runtime->tsleep); runtime->status->state = SNDRV_PCM_STATE_OPEN; + runtime->start_at_tstamp_type = SNDRV_PCM_START_AT_TSTAMP_TYPE_NULL; substream->runtime = runtime; substream->private_data = pcm->private_data; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 095d957..352921d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -35,6 +35,9 @@ #include <sound/timer.h> #include <sound/minors.h> #include <asm/io.h> +#if defined(CONFIG_HIGH_RES_TIMERS) +#include <linux/hrtimer.h> +#endif /* * Compatibility @@ -67,6 +70,8 @@ static int snd_pcm_hw_params_old_user(struct snd_pcm_substream *substream, #endif static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream); +static int snd_pcm_start_at_cancel(struct snd_pcm_substream *substream); + /* * */ @@ -265,6 +270,7 @@ static const char * const snd_pcm_hw_param_names[] = { }; #endif + int snd_pcm_hw_refine(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -834,6 +840,11 @@ static int snd_pcm_action_group(struct action_ops *ops, struct snd_pcm_substream *s1; int res = 0, depth = 1; + /* Any attempt to change state cancels a pending start_at timer */ + res = snd_pcm_start_at_cancel(substream); + if (res < 0) + return res; + snd_pcm_group_for_each_entry(s, substream) { if (do_lock && s != substream) { if (s->pcm->nonatomic) @@ -888,6 +899,11 @@ static int snd_pcm_action_single(struct action_ops *ops, int state) { int res; + + /* Any attempt to change state cancels a pending start_at timer */ + res = snd_pcm_start_at_cancel(substream); + if (res < 0) + return res; res = ops->pre_action(substream, state); if (res < 0) @@ -1015,6 +1031,234 @@ static struct action_ops snd_pcm_action_start = { .post_action = snd_pcm_post_start }; +static inline clockid_t snd_pcm_get_clockid(struct snd_pcm_substream* substream) +{ + switch(substream->runtime->tstamp_type) + { + case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC: + return CLOCK_MONOTONIC; + case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW: + return CLOCK_MONOTONIC_RAW; + default: + return CLOCK_REALTIME; + } +} + +/* snd_pcm_start_at_ops + * There are various mechanisms for supporting snd_pcm_start_at. Posix clocks may + * use hires timers if available. + * + * schedule() may use the runtime member 'start_at_timer_data' to store enough + * information to cancel the timer. When the timer fires, it must be single-shot, + * and must not cancel/delete the timer: cancel() is guaranteed to be called + * before the timer is reused, so resources should be freed within cancel(), not + * by the timer callback. + * + * cancel() returns when the start_at timer callback is guaranteed to be both + * cancelled and not currently running. +*/ + +struct snd_pcm_start_at_ops { + int (*schedule)(struct snd_pcm_substream *substream, const struct timespec *start_time); + int (*cancel)(struct snd_pcm_substream *substream); +}; + +/* To support snd_pcm_start_at for posix tstamp_types, we use high-res timers, if + * kernel is configured appropriately. + */ + +#ifdef CONFIG_HIGH_RES_TIMERS +/* + * hrtimer interface + */ + +struct hrtimer_pcm { + struct hrtimer timer; + struct snd_pcm_substream *substream; +}; + +/* + * called from a hard irq context - no need for locks. + * only problem is that the caller might have gone away and closed the substream + * before the timer expires. + */ +enum hrtimer_restart snd_pcm_do_start_time(struct hrtimer *timer) +{ + struct hrtimer_pcm *pcm_timer; + struct snd_pcm_substream *substream; + int ret; + + pcm_timer = container_of(timer, struct hrtimer_pcm, timer); + substream = pcm_timer->substream; + + ret = snd_pcm_do_start(substream, SNDRV_PCM_STATE_RUNNING); + if (ret == 0) { + snd_pcm_post_start(substream, SNDRV_PCM_STATE_RUNNING); + } + return HRTIMER_NORESTART; +} +#endif + + + +static int start_at_posix_schedule(struct snd_pcm_substream *substream, const struct timespec *start_time) +{ +#ifdef CONFIG_HIGH_RES_TIMERS + struct hrtimer_pcm *pcm_timer; + struct timespec now; + int ret; + + /* Get time now and check if start_time is in the past */ + snd_pcm_gettime(substream->runtime, &now); + if (timespec_compare(&now, start_time) >= 0) { + return -ETIME; + } + + /* Allocate a hrtimer to handle the start_at */ + pcm_timer = kmalloc(sizeof(*pcm_timer), GFP_KERNEL); + if (!pcm_timer) + return -ENOMEM; + + hrtimer_init(&pcm_timer->timer, snd_pcm_get_clockid(substream), HRTIMER_MODE_ABS); + + /* Setup timer */ + pcm_timer->timer.function = snd_pcm_do_start_time; + pcm_timer->substream = substream; + + /* Store timer in start_at info */ + substream->runtime->start_at_timer_data = pcm_timer; + + /* Pre start */ + ret = snd_pcm_pre_start(substream, SNDRV_PCM_STATE_PREPARED); + if (ret < 0) + goto error; + + ret = hrtimer_start(&pcm_timer->timer, timespec_to_ktime(*start_time), HRTIMER_MODE_ABS); + if (ret < 0 ) + goto error; + + return 0; +error: + kfree(pcm_timer); + return ret; +#else + return -ENOSYS; +#endif +} + +static int start_at_posix_cancel(struct snd_pcm_substream *substream) +{ +#ifdef CONFIG_HIGH_RES_TIMERS + struct hrtimer_pcm *pcm_timer = substream->runtime->start_at_timer_data; + hrtimer_cancel(&pcm_timer->timer); /* Cancel existing timer. (NOP if it's not running) */ + kfree(pcm_timer); + return 0; +#else + return -ENOSYS; +#endif +} + +static int start_at_wallclock_schedule(struct snd_pcm_substream *substream, const struct timespec *start_time) +{ + if (substream->ops->wall_clock_start_at) + return substream->ops->wall_clock_start_at(substream, start_time); + else + return -ENOSYS; +} + +static int start_at_wallclock_cancel(struct snd_pcm_substream *substream) +{ + if (substream->ops->wall_clock_start_at_cancel) + return substream->ops->wall_clock_start_at_cancel(substream); + else + return -ENOSYS; +} + +static struct snd_pcm_start_at_ops start_at_ops[SNDRV_PCM_START_AT_TSTAMP_TYPE_NULL+1] = { + [SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY] = { + .schedule = start_at_posix_schedule, + .cancel = start_at_posix_cancel + }, + [SNDRV_PCM_TSTAMP_TYPE_MONOTONIC] = { + .schedule = start_at_posix_schedule, + .cancel = start_at_posix_cancel + }, + [SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW] = { /* hrtimers can't handle CLOCK_MONOTONIC_RAW */ + .schedule = NULL, + .cancel = NULL, + }, + [SNDRV_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK] = { /* delegate start_at to pcm driver */ + .schedule = start_at_wallclock_schedule, + .cancel = start_at_wallclock_cancel, + }, + [SNDRV_PCM_START_AT_TSTAMP_TYPE_NULL] = { /* null handler - required for handling first cancellation */ + .schedule = NULL, + .cancel = NULL, + }, +}; + +/* snd_pcm_start_at_cancel() allows state-transition code to conveniently cancel the pending timer */ +static int snd_pcm_start_at_cancel(struct snd_pcm_substream *substream) +{ + struct snd_pcm_start_at_ops *ops = &start_at_ops[substream->runtime->start_at_tstamp_type]; + int ret = 0; + + /* If ops->cancel is NULL, it's not an error. */ + if (ops->cancel) { + ret = ops->cancel(substream); + if (ret == 0) + substream->runtime->start_at_tstamp_type = SNDRV_PCM_START_AT_TSTAMP_TYPE_NULL; + } + + return ret; +} + +int snd_pcm_start_at(struct snd_pcm_substream *substream, + struct timespec __user *_start_time) +{ + struct timespec start_time; + int new_tstamp_type; + struct snd_pcm_start_at_ops *ops; + int ret; + + if (copy_from_user(&start_time, _start_time, sizeof(start_time))) + return -EFAULT; + + if (!timespec_valid(&start_time)) + return -EINVAL; + + /* If not a playback substream, give up */ + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + /* Cancel any existing timer */ + ret = snd_pcm_start_at_cancel(substream); + if (ret < 0) + return ret; + + /* Get new tstamp_type */ + new_tstamp_type = substream->runtime->tstamp_type; + + /* Save current start_at tstamp_type. This way, it's valid before + schedule() is called */ + substream->runtime->start_at_tstamp_type = new_tstamp_type; + + /* Get apprpriate start_at ops */ + ops = &start_at_ops[new_tstamp_type]; + + /* Schedule start_at. If it doesn't exist, that's an error. */ + if (ops->schedule) { + ret = ops->schedule(substream, &start_time); + /* If successful, mark timer as cancelled */ + if (ret < 0) + /* If schedule() failed, reset tstamp type */ + substream->runtime->start_at_tstamp_type = SNDRV_PCM_START_AT_TSTAMP_TYPE_NULL; + return ret; + } + else + return -ENOSYS; +} + /** * snd_pcm_start - start all linked streams * @substream: the PCM substream instance @@ -2721,6 +2965,8 @@ static int snd_pcm_common_ioctl1(struct file *file, return snd_pcm_action_lock_irq(&snd_pcm_action_start, substream, SNDRV_PCM_STATE_RUNNING); case SNDRV_PCM_IOCTL_LINK: return snd_pcm_link(substream, (int)(unsigned long) arg); + case SNDRV_PCM_IOCTL_START_AT: + return snd_pcm_start_at(substream, arg); case SNDRV_PCM_IOCTL_UNLINK: return snd_pcm_unlink(substream); case SNDRV_PCM_IOCTL_RESUME: -- 1.9.1

1 0

[alsa-devel] [PATCH 00/30 v3] ALSA: Enhancement for existed FireWire drivers
by Takashi Sakamoto 03 Dec '14

03 Dec '14

This patchset is to enable ALSA Dice driver to handle more models, to improve ALSA firewire-speakers driver as OXFW driver. This patchset is for linux-next branch. == Updates from my previous post: [alsa-devel] [PATCH 00/29 v2] ALSA: Enhancement for existed FireWire drivers http://mailman.alsa-project.org/pipermail/alsa-devel/2014-October/082844.ht… - Add a patch to fix some minor bugs (dice) - Improve handling config-rom strings for loud models (oxfw) - Fix a bug of mutual exclusive lock for AMDTP streams (oxfw) - Obsolete atomic_t for reference counters (dice/oxfw) Takashi Sakamoto (30): ALSA: dice: suppress checkpatch.pl warnings ALSA: dice: Rename structure and its members ALSA: dice: Move file to its own directory ALSA: dice: Split transaction functionality into a file ALSA: dice: Split stream functionality into a file ALSA: dice: Split PCM functionality into a file ALSA: dice: Split hwdep functionality into a file ALSA: dice: Split proc interface into a file ALSA: dice: Add new functions for constraints of PCM parameters ALSA: dice: Change the way to start stream ALSA: dice: Add support for duplex streams with synchronization ALSA: dice: Support for non SYT-Match sampling clock source mode ALSA: dice: Add support for capturing PCM samples ALSA: dice: Add support for MIDI capture/playback ALSA: dice: remove experimental state ALSA: speakers: Rename to oxfw and rename some members ALSA: oxfw: Move to its own directory ALSA: oxfw: Split stream functionality to a new file and add a header file ALSA: oxfw: Split PCM functionality to a new file ALSA: oxfw: Split control functionality to a new file ALSA: oxfw: Change the way to name card ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation ALSA: oxfw: Change the way to make PCM rules/constraints ALSA: oxfw: Add proc interface for debugging purpose ALSA: oxfw: Change the way to start stream ALSA: oxfw: Add support for Behringer/Mackie devices ALSA: oxfw: Add support AMDTP in-stream ALSA: oxfw: add support for capturing PCM samples ALSA: oxfw: Add support for capture/playback MIDI messages ALSA: oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 26 +- sound/firewire/Makefile | 7 +- sound/firewire/dice.c | 1511 ---------------------------- sound/firewire/dice/Makefile | 3 + sound/firewire/dice/dice-hwdep.c | 190 ++++ sound/firewire/{ => dice}/dice-interface.h | 0 sound/firewire/dice/dice-midi.c | 194 ++++ sound/firewire/dice/dice-pcm.c | 424 ++++++++ sound/firewire/dice/dice-proc.c | 252 +++++ sound/firewire/dice/dice-stream.c | 407 ++++++++ sound/firewire/dice/dice-transaction.c | 382 +++++++ sound/firewire/dice/dice.c | 361 +++++++ sound/firewire/dice/dice.h | 189 ++++ sound/firewire/oxfw/Makefile | 3 + sound/firewire/oxfw/oxfw-command.c | 153 +++ sound/firewire/oxfw/oxfw-control.c | 283 ++++++ sound/firewire/oxfw/oxfw-hwdep.c | 190 ++++ sound/firewire/oxfw/oxfw-midi.c | 207 ++++ sound/firewire/oxfw/oxfw-pcm.c | 425 ++++++++ sound/firewire/oxfw/oxfw-proc.c | 113 +++ sound/firewire/oxfw/oxfw-stream.c | 685 +++++++++++++ sound/firewire/oxfw/oxfw.c | 315 ++++++ sound/firewire/oxfw/oxfw.h | 145 +++ sound/firewire/speakers.c | 792 --------------- 26 files changed, 4944 insertions(+), 2319 deletions(-) delete mode 100644 sound/firewire/dice.c create mode 100644 sound/firewire/dice/Makefile create mode 100644 sound/firewire/dice/dice-hwdep.c rename sound/firewire/{ => dice}/dice-interface.h (100%) create mode 100644 sound/firewire/dice/dice-midi.c create mode 100644 sound/firewire/dice/dice-pcm.c create mode 100644 sound/firewire/dice/dice-proc.c create mode 100644 sound/firewire/dice/dice-stream.c create mode 100644 sound/firewire/dice/dice-transaction.c create mode 100644 sound/firewire/dice/dice.c create mode 100644 sound/firewire/dice/dice.h create mode 100644 sound/firewire/oxfw/Makefile create mode 100644 sound/firewire/oxfw/oxfw-command.c create mode 100644 sound/firewire/oxfw/oxfw-control.c create mode 100644 sound/firewire/oxfw/oxfw-hwdep.c create mode 100644 sound/firewire/oxfw/oxfw-midi.c create mode 100644 sound/firewire/oxfw/oxfw-pcm.c create mode 100644 sound/firewire/oxfw/oxfw-proc.c create mode 100644 sound/firewire/oxfw/oxfw-stream.c create mode 100644 sound/firewire/oxfw/oxfw.c create mode 100644 sound/firewire/oxfw/oxfw.h delete mode 100644 sound/firewire/speakers.c -- 2.1.0

3 47

[alsa-devel] [alsa-dev] Variable latency issue in alsa-jack-plugins
by Corfu, Valentin (Wind River) 03 Dec '14

03 Dec '14

Hello guys, I recently encountered a problem about variable latency in alsa-jack-plugin. I have successfully checked the latency with jack_iodelay in following way: $ jack_iodelay & $ alsaloop -Pjackplug -Cjackplug & $ jack_connect alsa-jack.jackP.998.0:out_000 jack_delay:in $ jack_connect jack_delay:out alsa-jack.jackC.998.1:in_000 But total roundtrip latency, seems to be variable after each start-up. Q1: What can be the root cause of this variable latency issue? Q2: How can I find theoretical latency of the alsa-jack-plugin in both directions (playback - capture) ? Thanks and Best Regards, Valentin Intel GmbH Dornacher Strasse 1 85622 Feldkirchen/Muenchen, Deutschland Sitz der Gesellschaft: Feldkirchen bei Muenchen Geschaeftsfuehrer: Christian Lamprechter, Hannes Schwaderer, Douglas Lusk Registergericht: Muenchen HRB 47456 Ust.-IdNr./VAT Registration No.: DE129385895 Citibank Frankfurt a.M. (BLZ 502 109 00) 600119052

1 0

[alsa-devel] [PATCH] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 03 Dec '14

03 Dec '14

When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK to save power. Enable runtime power management, and auto enable/disable MCLK in pm_runtime resume and suspend. When wm8960 codec is being used, it will triger resume() to enable MCLK. When codec is not being used, it will triger suspend() to disable MCLK and power down PLL. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- include/sound/wm8960.h | 1 + sound/soc/codecs/wm8960.c | 51 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 52 insertions(+) diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index e8ce8ee..d7e84ed 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -16,6 +16,7 @@ #define WM8960_DRES_MAX 3 struct wm8960_data { + struct clk *mclk; bool capless; /* Headphone outputs configured in capless mode */ bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */ diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 031a1ae..6a529b1 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,8 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -988,6 +990,8 @@ static void wm8960_set_pdata_from_of(struct i2c_client *i2c, if (of_property_read_bool(np, "wlf,shared-lrclk")) pdata->shared_lrclk = true; + + pdata->mclk = devm_clk_get(&i2c->dev, NULL); } static int wm8960_i2c_probe(struct i2c_client *i2c, @@ -1011,6 +1015,14 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, else if (i2c->dev.of_node) wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + /* Mark the mclk pointer to NULL if no mclk assigned */ + if (IS_ERR(wm8960->pdata.mclk)) { + /* But do not ignore the request for probe defer */ + if (PTR_ERR(wm8960->pdata.mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + wm8960->pdata.mclk = NULL; + } + ret = wm8960_reset(wm8960->regmap); if (ret != 0) { dev_err(&i2c->dev, "Failed to issue reset\n"); @@ -1041,6 +1053,9 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8960); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); @@ -1053,6 +1068,41 @@ static int wm8960_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME +static int wm8960_runtime_resume(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + int ret; + + if (wm8960->pdata.mclk) { + ret = clk_prepare_enable(wm8960->pdata.mclk); + if (ret) { + dev_err(dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int wm8960_runtime_suspend(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + + /* Power down PLL to save power */ + regmap_update_bits(wm8960->regmap, WM8960_POWER2, 0x1, 0); + + if (wm8960->pdata.mclk) + clk_disable_unprepare(wm8960->pdata.mclk); + + return 0; +} +#endif + +static const struct dev_pm_ops wm8960_pm = { + SET_RUNTIME_PM_OPS(wm8960_runtime_suspend, wm8960_runtime_resume, NULL) +}; + static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -1070,6 +1120,7 @@ static struct i2c_driver wm8960_i2c_driver = { .name = "wm8960", .owner = THIS_MODULE, .of_match_table = wm8960_of_match, + .pm = &wm8960_pm, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Define the DCAPS preset for the old Intel chipsets
by Takashi Iwai 03 Dec '14

03 Dec '14

Just for improving readability. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 37 ++++++++++++++++++++----------------- 1 file changed, 20 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fc7aff0eb562..53e43d111a3b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -276,6 +276,10 @@ enum { (((chip)->driver_caps & AZX_DCAPS_SNOOP_MASK) >> 10) #define AZX_DCAPS_SNOOP_TYPE(type) ((AZX_SNOOP_TYPE_ ## type) << 10) +/* quirks for old Intel chipsets */ +#define AZX_DCAPS_INTEL_ICH \ + (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_BUFSIZE) + /* quirks for Intel PCH */ #define AZX_DCAPS_INTEL_PCH_NOPM \ (AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ @@ -2054,31 +2058,30 @@ static const struct pci_device_id azx_ids[] = { /* Braswell */ { PCI_DEVICE(0x8086, 0x2284), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, - /* ICH */ + /* ICH6 */ { PCI_DEVICE(0x8086, 0x2668), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH6 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH7 */ { PCI_DEVICE(0x8086, 0x27d8), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH7 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ESB2 */ { PCI_DEVICE(0x8086, 0x269a), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ESB2 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH8 */ { PCI_DEVICE(0x8086, 0x284b), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH8 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH9 */ { PCI_DEVICE(0x8086, 0x293e), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH9 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH9 */ { PCI_DEVICE(0x8086, 0x293f), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH9 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH10 */ { PCI_DEVICE(0x8086, 0x3a3e), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH10 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH10 */ { PCI_DEVICE(0x8086, 0x3a6e), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH10 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, /* Generic Intel */ { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, -- 2.1.3

2 3

[alsa-devel] [PATCH] ALSA: i2sbus: Deletion of unnecessary checks before the function call "release_and_free_resource"
by SF Markus Elfring 03 Dec '14

03 Dec '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 2 Dec 2014 22:50:24 +0100 The release_and_free_resource() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/aoa/soundbus/i2sbus/core.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index a80d5ea..4e2b4fb 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -79,8 +79,7 @@ static void i2sbus_release_dev(struct device *dev) if (i2sdev->out.dbdma) iounmap(i2sdev->out.dbdma); if (i2sdev->in.dbdma) iounmap(i2sdev->in.dbdma); for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) - if (i2sdev->allocated_resource[i]) - release_and_free_resource(i2sdev->allocated_resource[i]); + release_and_free_resource(i2sdev->allocated_resource[i]); free_dbdma_descriptor_ring(i2sdev, &i2sdev->out.dbdma_ring); free_dbdma_descriptor_ring(i2sdev, &i2sdev->in.dbdma_ring); for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) @@ -323,8 +322,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, if (dev->out.dbdma) iounmap(dev->out.dbdma); if (dev->in.dbdma) iounmap(dev->in.dbdma); for (i=0;i<3;i++) - if (dev->allocated_resource[i]) - release_and_free_resource(dev->allocated_resource[i]); + release_and_free_resource(dev->allocated_resource[i]); mutex_destroy(&dev->lock); kfree(dev); return 0; -- 2.1.3

2 1

[alsa-devel] [PATCH] sound: pci: azt3328.c: Cleaning up function that are not used anywhere
by Rickard Strandqvist 03 Dec '14

03 Dec '14

Remove function snd_azf3328_codec_outl() that is not used anywhere. This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/azt3328.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 5a69e26..82da7b9 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -379,15 +379,6 @@ snd_azf3328_codec_inw(const struct snd_azf3328_codec_data *codec, unsigned reg) } static inline void -snd_azf3328_codec_outl(const struct snd_azf3328_codec_data *codec, - unsigned reg, - u32 value -) -{ - outl(value, codec->io_base + reg); -} - -static inline void snd_azf3328_codec_outl_multi(const struct snd_azf3328_codec_data *codec, unsigned reg, const void *buffer, int count ) -- 1.7.10.4

2 1

Re: [alsa-devel] ASoC: core: Call mute for cpu dais as well
by Dan Carpenter 02 Dec '14

02 Dec '14

Hello Ramesh Babu, This is a semi-automatic email about new static checker warnings. The patch ae11601b80b9: "ASoC: core: Call mute for cpu dais as well" from Oct 15, 2014, leads to the following Smatch complaint: sound/soc/soc-pcm.c:777 soc_pcm_prepare() error: we previously assumed 'cpu_dai->driver->ops' could be null (see line 755) sound/soc/soc-pcm.c 754 755 if (cpu_dai->driver->ops && cpu_dai->driver->ops->prepare) { ^^^^^^^^^^^^^^^^^^^^ I don't think this can be NULL. Probably we should remove this check. 756 ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); 757 if (ret < 0) { 758 dev_err(cpu_dai->dev, "ASoC: DAI prepare error: %d\n", 759 ret); 760 goto out; 761 } 762 } 763 764 /* cancel any delayed stream shutdown that is pending */ 765 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && 766 rtd->pop_wait) { 767 rtd->pop_wait = 0; 768 cancel_delayed_work(&rtd->delayed_work); 769 } 770 771 snd_soc_dapm_stream_event(rtd, substream->stream, 772 SND_SOC_DAPM_STREAM_START); 773 774 for (i = 0; i < rtd->num_codecs; i++) 775 snd_soc_dai_digital_mute(rtd->codec_dais[i], 0, 776 substream->stream); 777 snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream); ^^^^^^^ But if it were NULL then this newly introduced function call would Oops. 778 779 out: regards, dan carpenter

3 2

[alsa-devel] [PATCH 1/1] ALSA: trident: Deletion of a check before snd_util_memhdr_free()
by SF Markus Elfring 02 Dec '14

02 Dec '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 2 Dec 2014 18:52:21 +0100 The snd_util_memhdr_free() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/trident/trident_main.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index da875dc..57cd757 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3702,8 +3702,7 @@ static int snd_trident_free(struct snd_trident *trident) free_irq(trident->irq, trident); if (trident->tlb.buffer.area) { outl(0, TRID_REG(trident, NX_TLBC)); - if (trident->tlb.memhdr) - snd_util_memhdr_free(trident->tlb.memhdr); + snd_util_memhdr_free(trident->tlb.memhdr); if (trident->tlb.silent_page.area) snd_dma_free_pages(&trident->tlb.silent_page); vfree(trident->tlb.shadow_entries); -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: echoaudio: Deletion of a check before release_and_free_resource()
by SF Markus Elfring 02 Dec '14

02 Dec '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 2 Dec 2014 18:34:45 +0100 The release_and_free_resource() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/echoaudio/echoaudio.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 60e4003..21228ad 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1875,8 +1875,7 @@ static int snd_echo_free(struct echoaudio *chip) if (chip->dsp_registers) iounmap(chip->dsp_registers); - if (chip->iores) - release_and_free_resource(chip->iores); + release_and_free_resource(chip->iores); pci_disable_device(chip->pci); -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: asihpi: Deletion of an unnecessary check before the function call "vfree"
by SF Markus Elfring 02 Dec '14

02 Dec '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 2 Dec 2014 18:05:32 +0100 The vfree() function performs also input parameter validation. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/asihpi/hpioctl.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index e457eb8..6aa677e 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -547,8 +547,7 @@ void asihpi_adapter_remove(struct pci_dev *pci_dev) if (pa->irq) free_irq(pa->irq, pa); - if (pa->p_buffer) - vfree(pa->p_buffer); + vfree(pa->p_buffer); if (1) dev_info(&pci_dev->dev, -- 2.1.3

2 1

[alsa-devel] [PATCH] ASoC: snd-soc-afeb9260: delete driver as board has just been removed
by Nicolas Ferre 02 Dec '14

02 Dec '14

During the removal of all AT91 !DT boards, we removed the AFEB9260. This driver is !DT and was only used by this board, so we delete it as well. Reported-by: Paul Bolle <pebolle(a)tiscali.nl> Signed-off-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/atmel/Makefile | 1 - sound/soc/atmel/snd-soc-afeb9260.c | 151 ------------------------------------- 2 files changed, 152 deletions(-) delete mode 100644 sound/soc/atmel/snd-soc-afeb9260.c diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 5baabc8bde3a..466a821da98c 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -17,4 +17,3 @@ snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o -obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c deleted file mode 100644 index 9579799ace54..000000000000 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ /dev/null @@ -1,151 +0,0 @@ -/* - * afeb9260.c -- SoC audio for AFEB9260 - * - * Copyright (C) 2009 Sergey Lapin <slapin(a)ossfans.org> - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/kernel.h> -#include <linux/clk.h> -#include <linux/platform_device.h> - -#include <linux/atmel-ssc.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> -#include <mach/hardware.h> -#include <linux/gpio.h> - -#include "../codecs/tlv320aic23.h" -#include "atmel-pcm.h" -#include "atmel_ssc_dai.h" - -#define CODEC_CLOCK 12000000 - -static int afeb9260_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int err; - - /* Set the codec system clock for DAC and ADC */ - err = - snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); - - if (err < 0) { - printk(KERN_ERR "can't set codec system clock\n"); - return err; - } - - return err; -} - -static struct snd_soc_ops afeb9260_ops = { - .hw_params = afeb9260_hw_params, -}; - -static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route afeb9260_audio_map[] = { - {"Headphone Jack", NULL, "LHPOUT"}, - {"Headphone Jack", NULL, "RHPOUT"}, - - {"LLINEIN", NULL, "Line In"}, - {"RLINEIN", NULL, "Line In"}, - - {"MICIN", NULL, "Mic Jack"}, -}; - - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link afeb9260_dai = { - .name = "TLV320AIC23", - .stream_name = "AIC23", - .cpu_dai_name = "atmel-ssc-dai.0", - .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "atmel_pcm-audio", - .codec_name = "tlv320aic23-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &afeb9260_ops, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_machine_afeb9260 = { - .name = "AFEB9260", - .owner = THIS_MODULE, - .dai_link = &afeb9260_dai, - .num_links = 1, - - .dapm_widgets = tlv320aic23_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), - .dapm_routes = afeb9260_audio_map, - .num_dapm_routes = ARRAY_SIZE(afeb9260_audio_map), -}; - -static struct platform_device *afeb9260_snd_device; - -static int __init afeb9260_soc_init(void) -{ - int err; - struct device *dev; - - if (!(machine_is_afeb9260())) - return -ENODEV; - - - afeb9260_snd_device = platform_device_alloc("soc-audio", -1); - if (!afeb9260_snd_device) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(afeb9260_snd_device, &snd_soc_machine_afeb9260); - err = platform_device_add(afeb9260_snd_device); - if (err) - goto err1; - - dev = &afeb9260_snd_device->dev; - - return 0; -err1: - platform_device_put(afeb9260_snd_device); - return err; -} - -static void __exit afeb9260_soc_exit(void) -{ - platform_device_unregister(afeb9260_snd_device); -} - -module_init(afeb9260_soc_init); -module_exit(afeb9260_soc_exit); - -MODULE_AUTHOR("Sergey Lapin <slapin(a)ossfans.org>"); -MODULE_DESCRIPTION("ALSA SoC for AFEB9260"); -MODULE_LICENSE("GPL"); - -- 2.1.3

3 4

[alsa-devel] [PATCH 2/2] ARM: tegra: Enable the mic-detect gpio on Acer Chromebook 13
by Dylan Reid 02 Dec '14

02 Dec '14

Enables the gpio-base mic detection on the Acer Chromebook 13. This gpio is set by the jack-detection chip when it notices either of the TRRS type headsets with a microphone. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- arch/arm/boot/dts/tegra124-nyan-big.dts | 2 ++ 1 file changed, 2 insertions(+) diff --git a/arch/arm/boot/dts/tegra124-nyan-big.dts b/arch/arm/boot/dts/tegra124-nyan-big.dts index 7d0784c..1c68ff9 100644 --- a/arch/arm/boot/dts/tegra124-nyan-big.dts +++ b/arch/arm/boot/dts/tegra124-nyan-big.dts @@ -1130,6 +1130,8 @@ clock-names = "pll_a", "pll_a_out0", "mclk"; nvidia,hp-det-gpios = <&gpio TEGRA_GPIO(I, 7) GPIO_ACTIVE_HIGH>; + nvidia,mic-det-gpios = + <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_HIGH>; }; }; -- 2.1.2.330.g565301e

3 2

[alsa-devel] [PATCH v7.1 00/19] Rework OMAP4+ HDMI audio support
by Jyri Sarha 02 Dec '14

02 Dec '14

The patches are based on: git://git.kernel.org/pub/scm/linux/kernel/git/tomba/linux.git for-next The base, the patches, and couple of additional not-to-be-merged omap2plus_defconfig patches can be found here: https://github.com/jsarha/linux.git omap-hdmi-audio It would make the most sense to get these in trough fbdev tree. So it would be nice to get acked-bys (if the patches are Ok) for ASoC side changes from appropriate maintainers. Changes since v7: - Squash: - "OMAPDSS: omapdss.h: Remove audio_state member of struct omap_dss_device" - into "OMAPDSS: Remove all references to obsolete HDMI audio callbacks" Changes since v6: - "ASoC: omap-hdmi-audio: Add platform device for OMAP HDMI audio support" - Fix Kconfig help text - Remove #include <sound/simple_card.h> - Use snd-soc-dummy codec instead of hdmi-audio-codec - Add: OMAPDSS: hdmi_wp: Protect reserved bits in hdmi_wp_audio_config_format() - Add: OMAPDSS: hdmi5_core: Initialize mandatory sample_order parameter - Add: "OMAPDSS: omapdss.h: Remove audio_state member of struct omap_dss_device" - "OMAPDSS: hdmi4: Register ASoC platform device for omap hdmi audio" - Register "omap-hdmi-audio" with PLATFORM_DEVID_AUTO - "OMAPDSS: hdmi5: Register ASoC platform device for omap hdmi audio" - Register "omap-hdmi-audio" with PLATFORM_DEVID_AUTO - Add: "OMAPDSS: hdmi5: Change hdmi_wp idlemode to to no_idle for audio playback" Jyri Sarha (19): OMAPDSS: hdmi_wp: Protect reserved bits in hdmi_wp_audio_config_format() OMAPDSS: hdmi5_core: Initialize mandatory sample_order parameter OMAPDSS: hdmi.h: Add HDMI_AUDIO_LAYOUT_6CH enum value OMAPDSS: hdmi: Remove most of OMAP[45]_DSS_HDMI_AUDIO ifdefs OMAPDSS: hdmi4_core: Remove unused hdmi4_audio_get_dma_port() OMAPDSS: hdmi_wp: Add function for getting audio dma address OMAPDSS: hdmi: Make hdmi structure public OMAPDSS: hdmi: Add pdev pointer for audio_pdev in HDMI DRV data ASoC: omap-hdmi-audio: Add platform device for OMAP HDMI audio support OMAPDSS: Kconfig: Remove HDMI audio booleans from Kconfig OMAPDSS: hdmi: Make hdmi_mode_has_audio() more user friedly OMAPDSS: hdmi.h: Add members to hdmi drvdata for audio implementation OMAPDSS: hdmi4: Remove callbacks for the old ASoC DAI driver OMAPDSS: hdmi4: Register ASoC platform device for omap hdmi audio OMAPDSS: hdmi5: Remove callbacks for the old ASoC DAI driver OMAPDSS: hdmi5: Register ASoC platform device for omap hdmi audio ASoC: omap: Remove obsolete HDMI audio code and Kconfig options OMAPDSS: Remove all references to obsolete HDMI audio callbacks OMAPDSS: hdmi5: Change hdmi_wp idlemode to to no_idle for audio playback .../fbdev/omap2/displays-new/connector-hdmi.c | 99 ----- .../fbdev/omap2/displays-new/encoder-tpd12s015.c | 56 --- drivers/video/fbdev/omap2/dss/Kconfig | 7 - drivers/video/fbdev/omap2/dss/hdmi.h | 38 +- drivers/video/fbdev/omap2/dss/hdmi4.c | 269 +++++++------ drivers/video/fbdev/omap2/dss/hdmi4_core.c | 14 - drivers/video/fbdev/omap2/dss/hdmi4_core.h | 4 - drivers/video/fbdev/omap2/dss/hdmi5.c | 265 ++++++------- drivers/video/fbdev/omap2/dss/hdmi5_core.c | 9 +- drivers/video/fbdev/omap2/dss/hdmi5_core.h | 2 - drivers/video/fbdev/omap2/dss/hdmi_common.c | 2 - drivers/video/fbdev/omap2/dss/hdmi_wp.c | 16 +- include/sound/omap-hdmi-audio.h | 43 +++ include/video/omapdss.h | 40 -- sound/soc/omap/Kconfig | 26 +- sound/soc/omap/Makefile | 6 +- sound/soc/omap/omap-hdmi-audio.c | 407 ++++++++++++++++++++ sound/soc/omap/omap-hdmi-card.c | 87 ----- sound/soc/omap/omap-hdmi.c | 364 ----------------- sound/soc/omap/omap-hdmi.h | 38 -- 20 files changed, 781 insertions(+), 1011 deletions(-) create mode 100644 include/sound/omap-hdmi-audio.h create mode 100644 sound/soc/omap/omap-hdmi-audio.c delete mode 100644 sound/soc/omap/omap-hdmi-card.c delete mode 100644 sound/soc/omap/omap-hdmi.c delete mode 100644 sound/soc/omap/omap-hdmi.h -- 1.7.9.5

5 51

[alsa-devel] [PATCHv6 1/9] ASOC: wm8971: clean for checkpatch --strict
by Xavier Hsu 02 Dec '14

02 Dec '14

Before adding wm8973 support, clean up existing wm8971 style. Signed-off-by: Xavier Hsu <xavier.hsu(a)linaro.org> Signed-off-by: Andy Green <andy.green(a)linaro.org> --- sound/soc/codecs/wm8971.c | 110 +++++++++++++++++++++++---------------------- sound/soc/codecs/wm8971.h | 30 ++++++------- 2 files changed, 72 insertions(+), 68 deletions(-) diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 39ddb9b..01c76a4 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -29,9 +29,9 @@ #include "wm8971.h" -#define WM8971_REG_COUNT 43 +#define WM8971_REG_COUNT 43 -static struct workqueue_struct *wm8971_workq = NULL; +static struct workqueue_struct *wm8971_workq; /* codec private data */ struct wm8971_priv { @@ -92,25 +92,28 @@ static const struct reg_default wm8971_reg_defaults[] = { #define wm8971_reset(c) snd_soc_write(c, WM8971_RESET, 0) /* WM8971 Controls */ -static const char *wm8971_bass[] = { "Linear Control", "Adaptive Boost" }; -static const char *wm8971_bass_filter[] = { "130Hz @ 48kHz", - "200Hz @ 48kHz" }; -static const char *wm8971_treble[] = { "8kHz", "4kHz" }; -static const char *wm8971_alc_func[] = { "Off", "Right", "Left", "Stereo" }; -static const char *wm8971_ng_type[] = { "Constant PGA Gain", - "Mute ADC Output" }; -static const char *wm8971_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; -static const char *wm8971_mono_mux[] = {"Stereo", "Mono (Left)", - "Mono (Right)", "Digital Mono"}; -static const char *wm8971_dac_phase[] = { "Non Inverted", "Inverted" }; -static const char *wm8971_lline_mux[] = {"Line", "NC", "NC", "PGA", - "Differential"}; -static const char *wm8971_rline_mux[] = {"Line", "Mic", "NC", "PGA", - "Differential"}; -static const char *wm8971_lpga_sel[] = {"Line", "NC", "NC", "Differential"}; -static const char *wm8971_rpga_sel[] = {"Line", "Mic", "NC", "Differential"}; -static const char *wm8971_adcpol[] = {"Normal", "L Invert", "R Invert", - "L + R Invert"}; +static const char const *wm8971_bass[] = {"Linear Control", "Adaptive Boost"}; +static const char const *wm8971_bass_filter[] = {"130Hz @ 48kHz", + "200Hz @ 48kHz"}; +static const char const *wm8971_treble[] = {"8kHz", "4kHz"}; +static const char const *wm8971_alc_func[] = {"Off", "Right", + "Left", "Stereo"}; +static const char const *wm8971_ng_type[] = {"Constant PGA Gain", + "Mute ADC Output"}; +static const char const *wm8971_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz"}; +static const char const *wm8971_mono_mux[] = {"Stereo", "Mono (Left)", + "Mono (Right)", "Digital Mono"}; +static const char const *wm8971_dac_phase[] = {"Non Inverted", "Inverted"}; +static const char const *wm8971_lline_mux[] = {"Line", "NC", "NC", + "PGA", "Differential"}; +static const char const *wm8971_rline_mux[] = {"Line", "Mic", "NC", + "PGA", "Differential"}; +static const char const *wm8971_lpga_sel[] = {"Line", "NC", "NC", + "Differential"}; +static const char const *wm8971_rpga_sel[] = {"Line", "Mic", "NC", + "Differential"}; +static const char const *wm8971_adcpol[] = {"Normal", "L Invert", + "R Invert", "L + R Invert"}; static const struct soc_enum wm8971_enum[] = { SOC_ENUM_SINGLE(WM8971_BASS, 7, 2, wm8971_bass), /* 0 */ @@ -136,24 +139,24 @@ static const struct snd_kcontrol_new wm8971_snd_controls[] = { SOC_DOUBLE_R("Capture Switch", WM8971_LINVOL, WM8971_RINVOL, 7, 1, 1), SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8971_LOUT1V, - WM8971_ROUT1V, 7, 1, 0), + WM8971_ROUT1V, 7, 1, 0), SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8971_LOUT2V, - WM8971_ROUT2V, 7, 1, 0), + WM8971_ROUT2V, 7, 1, 0), SOC_SINGLE("Mono Playback ZC Switch", WM8971_MOUTV, 7, 1, 0), SOC_DOUBLE_R("PCM Volume", WM8971_LDAC, WM8971_RDAC, 0, 255, 0), SOC_DOUBLE_R("Bypass Left Playback Volume", WM8971_LOUTM1, - WM8971_LOUTM2, 4, 7, 1), + WM8971_LOUTM2, 4, 7, 1), SOC_DOUBLE_R("Bypass Right Playback Volume", WM8971_ROUTM1, - WM8971_ROUTM2, 4, 7, 1), + WM8971_ROUTM2, 4, 7, 1), SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8971_MOUTM1, - WM8971_MOUTM2, 4, 7, 1), + WM8971_MOUTM2, 4, 7, 1), SOC_DOUBLE_R("Headphone Playback Volume", WM8971_LOUT1V, - WM8971_ROUT1V, 0, 127, 0), + WM8971_ROUT1V, 0, 127, 0), SOC_DOUBLE_R("Speaker Playback Volume", WM8971_LOUT2V, - WM8971_ROUT2V, 0, 127, 0), + WM8971_ROUT2V, 0, 127, 0), SOC_ENUM("Bass Boost", wm8971_enum[0]), SOC_ENUM("Bass Filter", wm8971_enum[1]), @@ -238,14 +241,14 @@ SOC_DAPM_ENUM("Route", wm8971_enum[13]); static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, - &wm8971_left_mixer_controls[0], - ARRAY_SIZE(wm8971_left_mixer_controls)), + &wm8971_left_mixer_controls[0], + ARRAY_SIZE(wm8971_left_mixer_controls)), SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, - &wm8971_right_mixer_controls[0], - ARRAY_SIZE(wm8971_right_mixer_controls)), + &wm8971_right_mixer_controls[0], + ARRAY_SIZE(wm8971_right_mixer_controls)), SND_SOC_DAPM_MIXER("Mono Mixer", WM8971_PWR2, 2, 0, - &wm8971_mono_mixer_controls[0], - ARRAY_SIZE(wm8971_mono_mixer_controls)), + &wm8971_mono_mixer_controls[0], + ARRAY_SIZE(wm8971_mono_mixer_controls)), SND_SOC_DAPM_PGA("Right Out 2", WM8971_PWR2, 3, 0, NULL, 0), SND_SOC_DAPM_PGA("Left Out 2", WM8971_PWR2, 4, 0, NULL, 0), @@ -260,18 +263,18 @@ static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = { SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8971_PWR1, 3, 0), SND_SOC_DAPM_MUX("Left PGA Mux", WM8971_PWR1, 5, 0, - &wm8971_left_pga_controls), + &wm8971_left_pga_controls), SND_SOC_DAPM_MUX("Right PGA Mux", WM8971_PWR1, 4, 0, - &wm8971_right_pga_controls), + &wm8971_right_pga_controls), SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, - &wm8971_left_line_controls), + &wm8971_left_line_controls), SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, - &wm8971_right_line_controls), + &wm8971_right_line_controls), SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, - &wm8971_monomux_controls), + &wm8971_monomux_controls), SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, - &wm8971_monomux_controls), + &wm8971_monomux_controls), SND_SOC_DAPM_OUTPUT("LOUT1"), SND_SOC_DAPM_OUTPUT("ROUT1"), @@ -431,7 +434,7 @@ static int get_coeff(int mclk, int rate) } static int wm8971_set_dai_sysclk(struct snd_soc_dai *codec_dai, - int clk_id, unsigned int freq, int dir) + int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); @@ -449,7 +452,7 @@ static int wm8971_set_dai_sysclk(struct snd_soc_dai *codec_dai, } static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai, - unsigned int fmt) + unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; u16 iface = 0; @@ -507,8 +510,8 @@ static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); @@ -553,7 +556,7 @@ static int wm8971_mute(struct snd_soc_dai *dai, int mute) } static int wm8971_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) + enum snd_soc_bias_level level) { u16 pwr_reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; @@ -580,11 +583,12 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec, } #define WM8971_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops wm8971_dai_ops = { .hw_params = wm8971_pcm_hw_params, @@ -616,6 +620,7 @@ static void wm8971_work(struct work_struct *work) container_of(work, struct snd_soc_dapm_context, delayed_work.work); struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); + wm8971_set_bias_level(codec, codec->dapm.bias_level); } @@ -637,7 +642,7 @@ static int wm8971_resume(struct snd_soc_codec *codec) snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_ON; queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, - msecs_to_jiffies(1000)); + msecs_to_jiffies(1000)); } return 0; @@ -660,7 +665,7 @@ static int wm8971_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, - msecs_to_jiffies(1000)); + msecs_to_jiffies(1000)); /* set the update bits */ snd_soc_update_bits(codec, WM8971_LDAC, 0x0100, 0x0100); @@ -675,7 +680,6 @@ static int wm8971_probe(struct snd_soc_codec *codec) return ret; } - /* power down chip */ static int wm8971_remove(struct snd_soc_codec *codec) { @@ -729,8 +733,8 @@ static int wm8971_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8971); - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_wm8971, &wm8971_dai, 1); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8971, + &wm8971_dai, 1); return ret; } diff --git a/sound/soc/codecs/wm8971.h b/sound/soc/codecs/wm8971.h index f31c38f..dc38e21 100644 --- a/sound/soc/codecs/wm8971.h +++ b/sound/soc/codecs/wm8971.h @@ -22,25 +22,25 @@ #define WM8971_ADCDAC 0x05 #define WM8971_IFACE 0x07 #define WM8971_SRATE 0x08 -#define WM8971_LDAC 0x0a -#define WM8971_RDAC 0x0b -#define WM8971_BASS 0x0c +#define WM8971_LDAC 0x0a +#define WM8971_RDAC 0x0b +#define WM8971_BASS 0x0c #define WM8971_TREBLE 0x0d #define WM8971_RESET 0x0f -#define WM8971_ALC1 0x11 -#define WM8971_ALC2 0x12 -#define WM8971_ALC3 0x13 +#define WM8971_ALC1 0x11 +#define WM8971_ALC2 0x12 +#define WM8971_ALC3 0x13 #define WM8971_NGATE 0x14 -#define WM8971_LADC 0x15 -#define WM8971_RADC 0x16 -#define WM8971_ADCTL1 0x17 -#define WM8971_ADCTL2 0x18 -#define WM8971_PWR1 0x19 -#define WM8971_PWR2 0x1a -#define WM8971_ADCTL3 0x1b +#define WM8971_LADC 0x15 +#define WM8971_RADC 0x16 +#define WM8971_ADCTL1 0x17 +#define WM8971_ADCTL2 0x18 +#define WM8971_PWR1 0x19 +#define WM8971_PWR2 0x1a +#define WM8971_ADCTL3 0x1b #define WM8971_ADCIN 0x1f -#define WM8971_LADCIN 0x20 -#define WM8971_RADCIN 0x21 +#define WM8971_LADCIN 0x20 +#define WM8971_RADCIN 0x21 #define WM8971_LOUTM1 0x22 #define WM8971_LOUTM2 0x23 #define WM8971_ROUTM1 0x24 -- 1.7.9.5

2 10

Re: [alsa-devel] Add new machine support for Dell
by Takashi Iwai 02 Dec '14

02 Dec '14

At Tue, 2 Dec 2014 07:03:45 +0000, Kailang wrote: > > Hi Takashi, > > Dell has new machine for ALC293. > It supports headset Mic and Headphone Mic. Thanks, applied with Cc to stable now. Takashi > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0000-add-headset-mic-support-for-new-dell-machine.patch <application/octet-stream (base64)>] >

1 0

[alsa-devel] [PATCH v2] ASoC: Intel: Move capture PCM pin to PCM0 for Broadwell/Haswell
by Jie Yang 02 Dec '14

02 Dec '14

Move capture PCM pin from PCM4 to PCM0 for Broadwell/Haswell. This will allow us to integrate with pulseaudio better for usually default device is set to 0. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/broadwell.c | 15 ++------------- sound/soc/intel/haswell.c | 14 ++------------ sound/soc/intel/sst-haswell-pcm.c | 34 ++++++++++++++++------------------ 3 files changed, 20 insertions(+), 43 deletions(-) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 52cb764..c256764 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -171,7 +171,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { /* Front End DAI links */ { .name = "System PCM", - .stream_name = "System Playback", + .stream_name = "System Playback/Capture", .cpu_dai_name = "System Pin", .platform_name = "haswell-pcm-audio", .dynamic = 1, @@ -180,6 +180,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .init = broadwell_rtd_init, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_playback = 1, + .dpcm_capture = 1, }, { .name = "Offload0", @@ -214,18 +215,6 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_capture = 1, }, - { - .name = "Capture PCM", - .stream_name = "Capture", - .cpu_dai_name = "Capture Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - }, - /* Back End DAI links */ { /* SSP0 - Codec */ diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c index 3981982..cb8a482 100644 --- a/sound/soc/intel/haswell.c +++ b/sound/soc/intel/haswell.c @@ -109,7 +109,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { /* Front End DAI links */ { .name = "System", - .stream_name = "System Playback", + .stream_name = "System Playback/Capture", .cpu_dai_name = "System Pin", .platform_name = "haswell-pcm-audio", .dynamic = 1, @@ -118,6 +118,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .init = haswell_rtd_init, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_playback = 1, + .dpcm_capture = 1, }, { .name = "Offload0", @@ -152,17 +153,6 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_capture = 1, }, - { - .name = "Capture", - .stream_name = "Capture", - .cpu_dai_name = "Capture Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - }, /* Back End DAI links */ { diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 5f70d81..0180b38 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -309,7 +309,7 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = { ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Mic Capture volume */ - SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, + SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 0, 0, 8, ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), }; @@ -396,8 +396,14 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, /* DSP stream type depends on DAI ID */ switch (rtd->cpu_dai->id) { case 0: - stream_type = SST_HSW_STREAM_TYPE_SYSTEM; - module_id = SST_HSW_MODULE_PCM_SYSTEM; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + stream_type = SST_HSW_STREAM_TYPE_SYSTEM; + module_id = SST_HSW_MODULE_PCM_SYSTEM; + } + else { + stream_type = SST_HSW_STREAM_TYPE_CAPTURE; + module_id = SST_HSW_MODULE_PCM_CAPTURE; + } break; case 1: case 2: @@ -410,10 +416,6 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, path_id = SST_HSW_STREAM_PATH_SSP0_OUT; module_id = SST_HSW_MODULE_PCM_REFERENCE; break; - case 4: - stream_type = SST_HSW_STREAM_TYPE_CAPTURE; - module_id = SST_HSW_MODULE_PCM_CAPTURE; - break; default: dev_err(rtd->dev, "error: invalid DAI ID %d\n", rtd->cpu_dai->id); @@ -779,6 +781,13 @@ static struct snd_soc_dai_driver hsw_dais[] = { .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, + .capture = { + .stream_name = "Analog Capture", + .channels_min = 2, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, + }, }, { /* PCM */ @@ -815,17 +824,6 @@ static struct snd_soc_dai_driver hsw_dais[] = { .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, - { - .name = "Capture Pin", - .id = HSW_PCM_DAI_ID_CAPTURE, - .capture = { - .stream_name = "Analog Capture", - .channels_min = 2, - .channels_max = 4, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, - }, - }, }; static const struct snd_soc_dapm_widget widgets[] = { -- 1.9.1

3 4

[alsa-devel] [PATCH] ASoC: fsl_ssi: free irq before irq_dispose_mapping()
by Jiada Wang 01 Dec '14

01 Dec '14

irq_dispose_mapping() in turns calls unregister_irq_proc(), which will remove irq proc entry, if IRQ is not freed before calling of irq_dispose_mapping(), then it will cause kernel warning. By free IRQ before irq_dispose_mapping(), this patch fix the following kernel warning found when remove of fsl_ssi driver: [ 31.515336] ------------[ cut here ]------------ [ 31.520091] WARNING: CPU: 2 PID: 434 at fs/proc/generic.c:521 remove_proc_entry+0x14c/0x16c() [ 31.528708] remove_proc_entry: removing non-empty directory 'irq/79', leaking at least '202c000.ss' [ 31.537911] Modules linked in: snd_soc_wm8962 snd_soc_imx_wm8962 snd_soc_fsl_ssi(-) evbug [ 31.546249] CPU: 2 PID: 434 Comm: rmmod Not tainted 3.18.0-rc6-00028-g3314bf6-dirty #1 [ 31.554235] Backtrace: [ 31.556816] [<80011ea8>] (dump_backtrace) from [<80012044>] (show_stack+0x18/0x1c) [ 31.564416] r6:80142c88 r5:00000000 r4:00000000 r3:00000000 [ 31.570267] [<8001202c>] (show_stack) from [<806980ec>] (dump_stack+0x88/0xa4) [ 31.577588] [<80698064>] (dump_stack) from [<80029d78>] (warn_slowpath_common+0x70/0x94) [ 31.585711] r5:00000009 r4:bb61fd90 [ 31.589423] [<80029d08>] (warn_slowpath_common) from [<80029e40>] (warn_slowpath_fmt+0x38/0x40) [ 31.598187] r8:bb61fdfe r7:be05d76d r6:be05d9a8 r5:00000002 r4:be05d700 [ 31.605054] [<80029e0c>] (warn_slowpath_fmt) from [<80142c88>] (remove_proc_entry+0x14c/0x16c) [ 31.613709] r3:806a79c0 r2:808229a0 [ 31.617371] [<80142b3c>] (remove_proc_entry) from [<80070380>] (unregister_irq_proc+0x94/0xb8) [ 31.625989] r10:00000000 r8:8000ede4 r7:80955f2c r6:0000004f r5:8118e738 r4:be00af00 [ 31.633952] [<800702ec>] (unregister_irq_proc) from [<80069dac>] (free_desc+0x2c/0x64) [ 31.641898] r6:0000004f r5:80955f38 r4:be00af00 [ 31.646604] [<80069d80>] (free_desc) from [<80069e68>] (irq_free_descs+0x4c/0x8c) [ 31.654092] r7:00000081 r6:00000001 r5:0000004f r4:00000001 [ 31.659863] [<80069e1c>] (irq_free_descs) from [<8006fc3c>] (irq_dispose_mapping+0x40/0x5c) [ 31.668247] r6:be17b844 r5:be17b800 r4:0000004f r3:802c5ec0 [ 31.673998] [<8006fbfc>] (irq_dispose_mapping) from [<7f004ea4>] (fsl_ssi_remove+0x58/0x70 [snd_so) [ 31.683948] r4:bb5bba10 r3:00000001 [ 31.687618] [<7f004e4c>] (fsl_ssi_remove [snd_soc_fsl_ssi]) from [<803720a0>] (platform_drv_remove) [ 31.697564] r5:7f0064f8 r4:be17b810 [ 31.701195] [<80372080>] (platform_drv_remove) from [<80370494>] (__device_release_driver+0x78/0xc) [ 31.710361] r5:7f0064f8 r4:be17b810 [ 31.713987] [<8037041c>] (__device_release_driver) from [<80370d20>] (driver_detach+0xbc/0xc0) [ 31.722631] r5:7f0064f8 r4:be17b810 [ 31.726259] [<80370c64>] (driver_detach) from [<80370304>] (bus_remove_driver+0x54/0x98) [ 31.734382] r6:00000800 r5:00000000 r4:7f0064f8 r3:bb67f500 [ 31.740149] [<803702b0>] (bus_remove_driver) from [<80371398>] (driver_unregister+0x30/0x50) [ 31.748617] r4:7f0064f8 r3:bd9f7080 [ 31.752245] [<80371368>] (driver_unregister) from [<80371f3c>] (platform_driver_unregister+0x14/0x) [ 31.761498] r4:7f00655c r3:7f005a70 [ 31.765130] [<80371f28>] (platform_driver_unregister) from [<7f005a84>] (fsl_ssi_driver_exit+0x14/) [ 31.776147] [<7f005a70>] (fsl_ssi_driver_exit [snd_soc_fsl_ssi]) from [<8008ed80>] (SyS_delete_mod) [ 31.786553] [<8008ec64>] (SyS_delete_module) from [<8000ec20>] (ret_fast_syscall+0x0/0x48) [ 31.794824] r6:00c46d18 r5:00000800 r4:00c46d18 [ 31.799530] ---[ end trace 954e8a3a15379e52 ]--- Moreover replace devm_request_irq() with request_irq() since there is no need to use it as now driver always frees IRQ manually. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/fsl/fsl_ssi.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index e695517..cfdb337 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1400,9 +1400,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) } if (ssi_private->use_dma) { - ret = devm_request_irq(&pdev->dev, ssi_private->irq, - fsl_ssi_isr, 0, dev_name(&pdev->dev), - ssi_private); + ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, + dev_name(&pdev->dev), ssi_private); if (ret < 0) { dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); @@ -1412,7 +1411,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) ret = fsl_ssi_debugfs_create(&ssi_private->dbg_stats, &pdev->dev); if (ret) - goto error_asoc_register; + goto error_debugfs_create; /* * If codec-handle property is missing from SSI node, we assume @@ -1453,6 +1452,10 @@ done: error_sound_card: fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); +error_debugfs_create: + if (ssi_private->use_dma) + free_irq(ssi_private->irq, ssi_private); + error_irq: snd_soc_unregister_component(&pdev->dev); @@ -1473,6 +1476,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); + if (ssi_private->use_dma) + free_irq(ssi_private->irq, ssi_private); + if (ssi_private->pdev) platform_device_unregister(ssi_private->pdev); snd_soc_unregister_component(&pdev->dev); -- 1.9.3

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[alsa-devel] [PATCH] ALSA: usb: Add support for Zoom R16/24 capture and midi interfaces
by Panu Matilainen 01 Dec '14

01 Dec '14

This makes the midi interface and capture work out of the box with R16 (and presumably R24 too but untested). Playback stream would also seem to function fine except for one caveat: no sound is produced, so it is disabled for now. Signed-off-by: Panu Matilainen <pmatilai(a)laiskiainen.org> --- sound/usb/quirks-table.h | 34 ++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 8 ++++++++ 2 files changed, 42 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c657752..73621c9 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3202,6 +3202,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { + /* ZOOM R16/24 in audio interface mode */ + USB_DEVICE(0x1686, 0x00dd), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + /* Control */ + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + /* Playback, sound is missing atm */ + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + /* Capture */ + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE, + }, + { + /* Midi */ + .ifnum = 3, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + .ifnum = -1 + }, + } + } +}, + +{ /* * Some USB MIDI devices don't have an audio control interface, * so we have to grab MIDI streaming interfaces here. diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a5941f8..2e49193 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1160,6 +1160,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, break; } } + + /* Zoom R16/24 needs a tiny delay here, otherwise requests like + * get/set frequency return as failed despite actually succeeding. + */ + if ((le16_to_cpu(dev->descriptor.idVendor) == 0x1686) && + (le16_to_cpu(dev->descriptor.idProduct) == 0x00dd) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + mdelay(1); } /* -- 1.9.3

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[alsa-devel] [PATCH v2] ALSA: usb: Add support for Zoom R16/24 capture and midi interfaces
by Panu Matilainen 01 Dec '14

01 Dec '14

This makes the midi interface and capture work out of the box with R16 (and presumably R24 too but untested). Playback stream would also seem to function fine except for one caveat: no sound is produced, so it is disabled for now. Mixer descriptors are garbage and will require further quirks to enable functionality, also disabled here. Signed-off-by: Panu Matilainen <pmatilai(a)laiskiainen.org> --- sound/usb/quirks-table.h | 40 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 8 ++++++++ 2 files changed, 48 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c657752..c955663 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3203,6 +3203,46 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* + * ZOOM R16/24 in audio interface mode. + * Mixer descriptors are garbage, further quirks will be needed + * to make any of it functional, thus disabled for now. + * Playback stream appears to start and run fine but no sound + * is produced, so also disabled for now. + */ + USB_DEVICE(0x1686, 0x00dd), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + /* Mixer */ + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + /* Playback */ + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + /* Capture */ + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE, + }, + { + /* Midi */ + .ifnum = 3, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + .ifnum = -1 + }, + } + } +}, + +{ + /* * Some USB MIDI devices don't have an audio control interface, * so we have to grab MIDI streaming interfaces here. */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a5941f8..2e49193 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1160,6 +1160,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, break; } } + + /* Zoom R16/24 needs a tiny delay here, otherwise requests like + * get/set frequency return as failed despite actually succeeding. + */ + if ((le16_to_cpu(dev->descriptor.idVendor) == 0x1686) && + (le16_to_cpu(dev->descriptor.idProduct) == 0x00dd) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + mdelay(1); } /* -- 1.9.3

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[alsa-devel] [GIT PULL] Add sst audio firmware for baytrail platforms
by Vinod Koul 01 Dec '14

01 Dec '14

Hello Linux firmware maintainers, Please PULL to receive the Intel audio firmware for audio DSP in Baytrail platforms. This firmware is using the same license as another version of the Baytrail firmware already available in linux-firmware git. The following changes since commit 0e5f63771d0df6d7859f7c4100a74d737c62ac88: are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git byt Vinod Koul (1): linux-firmware: add sst audio firmware for baytrail platforms WHENCE | 8 ++++++++ intel/fw_sst_0f28.bin | Bin 0 -> 647078 bytes 2 files changed, 8 insertions(+), 0 deletions(-) create mode 100644 intel/fw_sst_0f28.bin Thanks -- ~Vinod

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