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

[alsa-devel] Resent HPI 4.03.36 updated summaries
by linux＠audioscience.com 06 Jul '10

06 Jul '10

Same patches, new summaries.

2 7

[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 06 Jul '10

06 Jul '10

The following changes since commit ce93a3702832121d517ad348817929f22fcce47c: ASoC: Add SmartQ sound driver (2010-07-04 18:04:12 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Mark Brown (4): ASoC: Fix sorting of Makefile and Kconfig ASoC: Remove current WM8960 deemphasis control ASoC: Automatically manage DAC deemphasis rate for WM8960 ASoC: Automatically manage ALC coefficients for WM8960 sound/soc/Kconfig | 2 +- sound/soc/Makefile | 2 +- sound/soc/codecs/wm8960.c | 99 +++++++++++++++++++++++++++++++++++++++++---- 3 files changed, 93 insertions(+), 10 deletions(-)

2 1

[alsa-devel] [PATCH] Updated alsa-kernel asihpi sources so patches are no longer required.
by linux＠audioscience.com 06 Jul '10

06 Jul '10

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- .gitignore | 2 -- pci/asihpi/Makefile | 3 --- pci/asihpi/hpidspcd.c | 2 ++ pci/asihpi/hpidspcd.patch | 21 --------------------- pci/asihpi/hpios.c | 2 ++ pci/asihpi/hpios.patch | 35 ----------------------------------- 6 files changed, 4 insertions(+), 61 deletions(-) create mode 100644 pci/asihpi/hpidspcd.c delete mode 100644 pci/asihpi/hpidspcd.patch create mode 100644 pci/asihpi/hpios.c delete mode 100644 pci/asihpi/hpios.patch diff --git a/.gitignore b/.gitignore index 204b952..db642b2 100644 --- a/.gitignore +++ b/.gitignore @@ -75,8 +75,6 @@ pci/ac97/ac97_bus.c pci/ac97/ac97_codec.c pci/ad1889.c pci/ali5451/ali5451.c -pci/asihpi/hpidspcd.c -pci/asihpi/hpios.c pci/atiixp.c pci/atiixp_modem.c pci/au88x0/au88x0.c diff --git a/pci/asihpi/Makefile b/pci/asihpi/Makefile index 95ee449..50ab325 100644 --- a/pci/asihpi/Makefile +++ b/pci/asihpi/Makefile @@ -5,7 +5,6 @@ endif include $(SND_TOPDIR)/toplevel.config include $(SND_TOPDIR)/Makefile.conf -clean-files := hpidspcd.c hpios.c EXTRA_CFLAGS = -I$(SND_TOPDIR)/alsa-kernel/pci/asihpi @@ -13,5 +12,3 @@ include $(SND_TOPDIR)/alsa-kernel/pci/asihpi/Makefile include $(SND_TOPDIR)/Rules.make -hpidspcd.c: hpidspcd.patch $(SND_TOPDIR)/alsa-kernel/pci/asihpi/hpidspcd.c -hpios.c: hpios.patch $(SND_TOPDIR)/alsa-kernel/pci/asihpi/hpios.c diff --git a/pci/asihpi/hpidspcd.c b/pci/asihpi/hpidspcd.c new file mode 100644 index 0000000..9b7d80a --- /dev/null +++ b/pci/asihpi/hpidspcd.c @@ -0,0 +1,2 @@ +#include "adriver.h" +#include "../../alsa-kernel/pci/asihpi/hpidspcd.c" diff --git a/pci/asihpi/hpidspcd.patch b/pci/asihpi/hpidspcd.patch deleted file mode 100644 index 7831639..0000000 --- a/pci/asihpi/hpidspcd.patch +++ /dev/null @@ -1,21 +0,0 @@ ---- ../../alsa-kernel/pci/asihpi/hpidspcd.c 2010-04-21 18:36:16.613453111 +0200 -+++ hpidspcd.c 2010-04-21 18:41:26.104970371 +0200 -@@ -1,3 +1,4 @@ -+#include "adriver.h" - /***********************************************************************/ - /*! - -@@ -73,8 +74,13 @@ - sprintf(fw_name, "asihpi/dsp%04x.bin", adapter); - HPI_DEBUG_LOG(INFO, "requesting firmware for %s\n", fw_name); - -+#if (LINUX_VERSION_CODE < KERNEL_VERSION(2 , 5 , 0)) -+ err = request_firmware(&ps_firmware, fw_name, -+ ps_dsp_code->ps_dev->slot_name); -+#else - err = request_firmware(&ps_firmware, fw_name, - &ps_dsp_code->ps_dev->dev); -+#endif - if (err != 0) { - HPI_DEBUG_LOG(ERROR, "%d, request_firmware failed for %s\n", - err, fw_name); diff --git a/pci/asihpi/hpios.c b/pci/asihpi/hpios.c new file mode 100644 index 0000000..3f151a3 --- /dev/null +++ b/pci/asihpi/hpios.c @@ -0,0 +1,2 @@ +#include "adriver.h" +#include "../../alsa-kernel/pci/asihpi/hpios.c" diff --git a/pci/asihpi/hpios.patch b/pci/asihpi/hpios.patch deleted file mode 100644 index 1af0f7f..0000000 --- a/pci/asihpi/hpios.patch +++ /dev/null @@ -1,35 +0,0 @@ ---- ../../alsa-kernel/pci/asihpi/hpios.c 2010-04-21 18:36:16.617452068 +0200 -+++ hpios.c 2010-04-21 18:40:32.700700812 +0200 -@@ -1,3 +1,4 @@ -+#include "adriver.h" - /****************************************************************************** - - AudioScience HPI driver -@@ -26,6 +27,7 @@ - #include <linux/delay.h> - #include <linux/sched.h> - -+#if LINUX_VERSION_CODE >= KERNEL_VERSION(2 , 6 , 14) - void hpios_delay_micro_seconds(u32 num_micro_sec) - { - if ((usecs_to_jiffies(num_micro_sec) > 1) && !in_interrupt()) { -@@ -38,6 +40,19 @@ - mdelay(num_micro_sec / 1000); - - } -+#else -+void hpios_delay_micro_seconds(u32 num_micro_sec) -+{ -+ if ((num_micro_sec / 1000 >= 1000000 / HZ) && !in_interrupt()) { -+ set_current_state(TASK_UNINTERRUPTIBLE); -+ schedule_timeout((HZ * num_micro_sec + (HZ - 1)) / 1000000); -+ } else if (num_micro_sec <= 2000) -+ udelay(num_micro_sec); -+ else -+ mdelay(num_micro_sec / 1000); -+ -+} -+#endif - - void hpios_locked_mem_init(void) - { -- 1.7.0.4

3 3

Re: [alsa-devel] Help with dummy.c (where/how to write?)
by Smilen Dimitrov 06 Jul '10

06 Jul '10

Hi Eliot, Thanks so much for your response: >> >> I am trying to work on dummy.c: >> >> http://lxr.linux.no/#linux+v2.6.32/sound/drivers/dummy.c >> >> as an exercise, and I'd like to modify it so it behaves as an >> 'oscillator'; [snip] >> > > You are better to look at the loopback driver in alsa-driver repo: > > http://git.alsa-project.org/?p=alsa-driver.git;a=blob_plain;f=drivers/aloop… > > You would put your generation code in clear_capture_buf() > Fantastic info - that is exactly what I needed :) (Don't know why your response hasn't shown in the alsa-devel archives yet, hopefully my reply will bump it up there) Cheers!

1 0

[alsa-devel] [RFC 00/16] ASoC: multi-component - CODECs
by Liam Girdwood 05 Jul '10

05 Jul '10

This is the second part of the multi-component series and focuses on the changes required to support ASoC multi-component on CODEC drivers. I've split the CODEC driver patches into smaller chunks and tried to group (where possible) similar CODECs together. All the patches will be rebased together at upstream time so we don't break bisect. Most CODEC drivers were extremely straight forward with changes limited to their probe() and remove() sections. Some CODECs did require a little extra work and this is marked in their commit messages. Thanks Liam

5 24

[alsa-devel] [RFC 0/5] ASoC multi-component support : core
by Liam Girdwood 05 Jul '10

05 Jul '10

ASoC is now coming up to it's 5th birthday and is beginning to show it's age a little when it comes to supporting the next generation of smart phones and multimedia devices with it's current single CODEC and single DMA platform design. Historically, ASoC is designed around an embedded sound card with a single CODEC and a single platform DMA driver (with multiple DAIs). This design works great for most of the current audio hardware in production, however we are now seeing hardware designs for the next generation of devices that can have many CODECs, DMAs, DSPs and a host of other audio peripherals connected to the audio subsystem. So we now need ASoC to support new audio hardware architectures with multiple audio CODECs, DMA engines, DSPs, DAIs etc. This patch series allows an ASoC sound card to support multiple components (CODECs, DAI and DMA) and is the first part in a series of work (other parts coming later in the year from Mark and I) designed to enhance ASoC for modern mobile audio hardware. This patch series provides the following important features:- 1) Allow sound cards to have multiple CODECs and platform DMA controllers. 2) Allow CODEC drivers to support more than one CODEC device. 3) Allow all CODEC, platform DMA and DAI drivers to have platform data. This is achieved by: - 1) Separating the component struct driver data from the component struct device data. 2) Making all components kernel devices. (A side effect of this change is that we can now store all our component private data in struct device private data (like everyone else) and there is also now less pointer indirection for components.) The change mainly affects only the probe() and remove() sections of component drivers and the component enumeration section of the core. The core also now handles setting up the CODEC data structs too (meaning CODEC drivers dont have to do this). All component PCM, DAPM and Kcontrol operations remain unchanged. I've so far tested the new multi-component enumeration on the TI OMAP and Marvell PXA platforms with AC97, MFD and I2C based CODECs. I do however need ASoC platform maintainers to test on their systems (as I only have a subset of the hardware). Some ASOC platforms do need a little extra checking/fixups too :- o Samsung platform: required the most work to untangle the components. o i.MX platform: required work to de-couple components. o Injenic J4740 platform: I do not have a toolchain to even build this arch so it will also need someone to build it too. If you find your platform does not work then the most likely reason is that I'm missing a platform_device registration for your CODEC, DAI, etc. device in your arch board.c or device.c files. Please let me know or send a patch :) All the multi-component code is in my multi-component branch here :- git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git This RFC post contains the changes to ASoC core only. I'll post the multi-component CODEC and platform changes as separate RFCs in order not to flood the list (as there are 38 patches in all). This patch series will eventually be rebased into a single patch for upstream so we dont break bisect. The subsequent patches for CODECs and platforms in this series can be found in the git branch above. The intention is to upstream this for 2.6.36 Thanks Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

5 14

[alsa-devel] Help with dummy.c (where/how to write?)
by Smilen Dimitrov 05 Jul '10

05 Jul '10

Hi all, I am trying to work on dummy.c: http://lxr.linux.no/#linux+v2.6.32/sound/drivers/dummy.c as an exercise, and I'd like to modify it so it behaves as an 'oscillator'; that is, when I choose 'dummy' as an input in say Audacity, and press Record - I'd like to be able to write something to the audio (pcm) buffer of the driver directly from the driver itself (say a single sample), and be able to observe those samples recorded in Audacity. So far, by adding some debug statements, the only thing I can figure, is that when 'Record' in Audacity is pressed, 'dummy_hrtimer_callback' is called continuously - but I cannot tell what functions to use to modify / write to output buffers (not even sure what the output buffers there are, I guess dpcm->substream->private_data). Could anyone point to me if there is a similar example existing already, or if not. provide me with some pointers on how to do this? Thanks in advance, Cheers!

1 0

[alsa-devel] [PATCH] ALSA: pcm_lib: avoid timing jitter in snd_pcm_read/write()
by David Dillow 05 Jul '10

05 Jul '10

When using poll() to wait for the next period -- or avail_min samples -- one gets a consistent delay for each system call that is usually just a little short of the selected period time. However, When using snd_pcm_read/write(), one gets a jittery delay that alternates between less than a millisecond and approximately two period times. This is caused by snd_pcm_lib_{read,write}1() transferring any available samples to the user's buffer and adjusting the application pointer prior to sleeping to the end of the current period. When the next period interrupt occurs, there is then less than avail_min samples remaining to be transferred in the period, so we end up sleeping until a second period occurs. This is solved by using runtime->twake as the number of samples needed for a wakeup in addition to selecting the proper wait queue to wake in snd_pcm_update_state(). This requires twake to be non-zero when used by snd_pcm_lib_{read,write}1() even if avail_min is zero. Signed-off-by: Dave Dillow <dave(a)thedillows.org> --- Hans, you may want this one as well for the SIS7019, as it gives more margin to deal with captured data on alternating calls into the stack. diff --git a/include/sound/pcm.h b/include/sound/pcm.h index dd76cde..83c6fa6 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -313,7 +313,7 @@ struct snd_pcm_runtime { struct snd_pcm_mmap_control *control; /* -- locking / scheduling -- */ - unsigned int twake: 1; /* do transfer (!poll) wakeup */ + snd_pcm_uframes_t twake; /* do transfer (!poll) wakeup if non-zero */ wait_queue_head_t sleep; /* poll sleep */ wait_queue_head_t tsleep; /* transfer sleep */ struct fasync_struct *fasync; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index e9d98be..bcf95d3 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -287,8 +287,11 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return -EPIPE; } } - if (avail >= runtime->control->avail_min) - wake_up(runtime->twake ? &runtime->tsleep : &runtime->sleep); + if (runtime->twake) { + if (avail >= runtime->twake) + wake_up(&runtime->tsleep); + } else if (avail >= runtime->control->avail_min) + wake_up(&runtime->sleep); return 0; } @@ -1707,7 +1710,7 @@ EXPORT_SYMBOL(snd_pcm_period_elapsed); * The available space is stored on availp. When err = 0 and avail = 0 * on the capture stream, it indicates the stream is in DRAINING state. */ -static int wait_for_avail_min(struct snd_pcm_substream *substream, +static int wait_for_avail(struct snd_pcm_substream *substream, snd_pcm_uframes_t *availp) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -1757,7 +1760,7 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, avail = snd_pcm_playback_avail(runtime); else avail = snd_pcm_capture_avail(runtime); - if (avail >= runtime->control->avail_min) + if (avail >= runtime->twake) break; } _endloop: @@ -1820,7 +1823,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->twake = 1; + runtime->twake = runtime->control->avail_min ? : 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -1833,7 +1836,9 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, err = -EAGAIN; goto _end_unlock; } - err = wait_for_avail_min(substream, &avail); + runtime->twake = min_t(snd_pcm_uframes_t, size, + runtime->control->avail_min ? : 1); + err = wait_for_avail(substream, &avail); if (err < 0) goto _end_unlock; } @@ -2042,7 +2047,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->twake = 1; + runtime->twake = runtime->control->avail_min ? : 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -2060,7 +2065,9 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, err = -EAGAIN; goto _end_unlock; } - err = wait_for_avail_min(substream, &avail); + runtime->twake = min_t(snd_pcm_uframes_t, size, + runtime->control->avail_min ? : 1); + err = wait_for_avail(substream, &avail); if (err < 0) goto _end_unlock; if (!avail)

5 12

[alsa-devel] [PATCH] ASoC: DaVinci: More accurate calculation for clock divider for McBSP (I2S)
by Raffaele Recalcati 05 Jul '10

05 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> i2s_accurate_sck switch can be used to have a better approximate sampling frequency. The trade off is between more accurate clock (fast clock) and less accurate clock (slow clock). The waveform will be not symmetric. Probably it is possible to get a better algorithm for calculating the divider, trying to keep a slower clock as possible. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 33 ++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-i2s.c | 28 ++++++++++++++++++++----- 2 files changed, 55 insertions(+), 6 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 0847d21..5149abe 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -73,6 +73,39 @@ struct snd_platform_data { */ int clk_input_pin; + /* + * This flag works when both clock and FS are outputs for the cpu + * and makes clock more accurate (FS is not symmetrical and the + * clock is very fast. + * The clock becoming faster is named + * i2s continuous serial clocl (I2S_SCK) and it is an externally + * visible bit clock. + * + * first line : WordSelect + * second line : ContinuousSerialClock + * third line: SerialData + * + * SYMMETRICAL APPROACH: + * _______________________ LEFT + * _| RIGHT |______________________| + * _ _ _ _ _ _ _ _ + * _| |_| |_ x16 _| |_| |_| |_| |_ x16 _| |_| |_ + * _ _ _ _ _ _ _ _ + * _/ \_/ \_ ... _/ \_/ \_/ \_/ \_ ... _/ \_/ \_ + * \_/ \_/ \_/ \_/ \_/ \_/ \_/ \_/ + * + * ACCURATE CLOCK APPROACH: + * ______________ LEFT + * _| RIGHT |_______________________________| + * _ _ _ _ _ _ _ _ _ + * _| |_ x16 _| |_| |_ x16 _| |_| |_| |_| |_| |_| | + * _ _ _ _ dummy cycles + * _/ \_ ... _/ \_/ \_ ... _/ \__________________ + * \_/ \_/ \_/ \_/ + * + */ + bool i2s_accurate_sck; + /* McASP specific fields */ int tdm_slots; u8 op_mode; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 17f594f..e6dcd81 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -122,6 +122,7 @@ static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { }; struct davinci_mcbsp_dev { + struct device *dev; struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 @@ -154,6 +155,7 @@ struct davinci_mcbsp_dev { unsigned int fmt; int clk_div; int clk_input_pin; + bool i2s_accurate_sck; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -296,7 +298,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, pcr |= DAVINCI_MCBSP_PCR_SCLKME; break; default: - dev_err(&pdev->dev, "bad clk_input_pin\n"); + dev_err(dev->dev, "bad clk_input_pin\n"); return -EINVAL; } @@ -447,11 +449,23 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, DAVINCI_MCBSP_SRGR_CLKSM; srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); - /* symmetric waveforms */ - clk_div = freq / (mcbsp_word_length * 16) / - params->rate_num * params->rate_den; - srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * - 16 - 1); + if (dev->i2s_accurate_sck) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + } clk_div &= 0xFF; srgr |= clk_div; break; @@ -662,6 +676,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; dev->clk_input_pin = pdata->clk_input_pin; + dev->i2s_accurate_sck = pdata->i2s_accurate_sck; } dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { @@ -694,6 +709,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) goto err_free_mem; } dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; + dev->dev = &pdev->dev; davinci_i2s_dai.private_data = dev; davinci_i2s_dai.capture.dma_data = dev->dma_params; -- 1.7.0.4

1 0

Re: [alsa-devel] ALSA: hda - Don't check capture source mixer if no ADC is available
by Dan Carpenter 05 Jul '10

05 Jul '10

Commit fbe618f2168 "ALSA: hda - Don't check capture source mixer if no ADC is available" added support for when there is no ADC available and spec->adc_nids is NULL. If spec->adc_nids is NULL, can it cause a problem later when we call snd_hda_add_nid()? sound/pci/hda/patch_realtek.c +2643 alc_build_controls(96) error: we previously assumed 'spec->adc_nids' could be null. 2635 if (spec->cap_mixer) { 2636 const char *kname = kctl ? kctl->id.name : NULL; 2637 for (knew = spec->cap_mixer; knew->name; knew++) { 2638 if (kname && strcmp(knew->name, kname) == 0) 2639 continue; 2640 kctl = snd_hda_find_mixer_ctl(codec, knew->name); 2641 for (i = 0; kctl && i < kctl->count; i++) { 2642 err = snd_hda_add_nid(codec, kctl, i, 2643 spec->adc_nids[i]); ^^^^^^^^^^^^^^^^^ 2644 if (err < 0) 2645 return err; 2646 } 2647 } 2648 } regards, dan carpenter

2 1