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

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[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

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[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

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[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

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