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[alsa-devel] [PATCH 2/3] ASoC: DaVinci: Added selection of clk input pin for McBSP
by Raffaele Recalcati 06 Jul '10

06 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> When McBSP peripheral gets the clock from an external pin, there are three possible chooses, MCBSP_CLKX, MCBSP_CLKR and MCBSP_CLKS. evm-dm365 uses MCBSP_CLKR, instead in bmx board I have a different hardware connection and I use MCBSP_CLKS, so I have added this possibility. 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) 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 | 15 +++++++++++++++ sound/soc/davinci/davinci-i2s.c | 29 ++++++++++++++++++++++++----- 2 files changed, 39 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..0847d21 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -63,6 +63,16 @@ struct snd_platform_data { unsigned sram_size_playback; unsigned sram_size_capture; + /* + * If McBSP peripheral gets the clock from an external pin, + * there are three chooses, that are MCBSP_CLKX, MCBSP_CLKR + * and MCBSP_CLKS. + * Depending on different hardware connections it is possible + * to use this setting to change the behaviour of McBSP + * driver. The dm365_clk_input_pin enum is available for dm365 + */ + int clk_input_pin; + /* McASP specific fields */ int tdm_slots; u8 op_mode; @@ -78,6 +88,11 @@ enum { MCASP_VERSION_2, /* DA8xx/OMAPL1x */ }; +enum dm365_clk_input_pin { + MCBSP_CLKR = 0, /* DM365 */ + MCBSP_CLKS, +}; + #define INACTIVE_MODE 0 #define TX_MODE 1 #define RX_MODE 2 diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index c8f038c..ba5644b 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 @@ -153,6 +154,7 @@ struct davinci_mcbsp_dev { unsigned int fmt; int clk_div; + int clk_input_pin; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -279,11 +281,26 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, DAVINCI_MCBSP_PCR_CLKRM; break; case SND_SOC_DAIFMT_CBM_CFS: - /* McBSP CLKR pin is the input for the Sample Rate Generator. - * McBSP FSR and FSX are driven by the Sample Rate Generator. */ - pcr = DAVINCI_MCBSP_PCR_SCLKME | - DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM; + pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM; + /* + * Selection of the clock input pin that is the + * input for the Sample Rate Generator. + * McBSP FSR and FSX are driven by the Sample Rate + * Generator. + */ + switch (dev->clk_input_pin) { + case MCBSP_CLKS: + pcr |= DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; + break; + case MCBSP_CLKR: + pcr |= DAVINCI_MCBSP_PCR_SCLKME; + break; + default: + dev_err(dev->dev, "bad clk_input_pin\n"); + return -EINVAL; + } + break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is master */ @@ -644,6 +661,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; + dev->clk_input_pin = pdata->clk_input_pin; } dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { @@ -676,6 +694,7 @@ 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

[PATCH - Fix Fast Track Ultra series: 44.1/88.1 kHz 1/Fix playback/capture issues and related chrashes at 44100 Hz and 88200 Hz for M-Audio Fast Track Ultra series.
by Felix 06 Jul '10

06 Jul '10

Signed-off-by: Felix <fexpop(a)web.de> diff --git a/usb/endpoint.c b/usb/endpoint.c index 1a701f1..bb9f938 100644 --- a/usb/endpoint.c +++ b/usb/endpoint.c @@ -404,8 +404,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ - case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ /* doesn't set the sample rate attribute, but supports it */ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; diff --git a/usb/quirks-table.h b/usb/quirks-table.h index f8797f6..af9c2a1 100644 --- a/usb/quirks-table.h +++ b/usb/quirks-table.h @@ -1827,10 +1827,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE(0x0763, 0x2080), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + USB_DEVICE(0x0763, 0x2081), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ - /* .product_name = "Fast Track Ultra 8", */ + /* .product_name = "Fast Track Ultra", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = & (const struct snd_usb_audio_quirk[]) { @@ -1840,11 +1840,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { @@ -1855,7 +1895,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE(0x0763, 0x2081), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ /* .product_name = "Fast Track Ultra 8R", */ .ifnum = QUIRK_ANY_INTERFACE, @@ -1867,11 +1907,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { -- 1.7.0.4 --------------090100050500090907030802--

1 0

[PATCH - Fix Fast Track Ultra series: 44.1/88.1 kHz 1/Fix playback/capture issues and related chrashes at 44100 Hz and 88200 Hz for M-Audio Fast Track Ultra series.
by Felix 06 Jul '10

06 Jul '10

Signed-off-by: Felix <fexpop(a)web.de> diff --git a/usb/endpoint.c b/usb/endpoint.c index 1a701f1..bb9f938 100644 --- a/usb/endpoint.c +++ b/usb/endpoint.c @@ -404,8 +404,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ - case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ /* doesn't set the sample rate attribute, but supports it */ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; diff --git a/usb/quirks-table.h b/usb/quirks-table.h index f8797f6..af9c2a1 100644 --- a/usb/quirks-table.h +++ b/usb/quirks-table.h @@ -1827,10 +1827,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE(0x0763, 0x2080), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + USB_DEVICE(0x0763, 0x2081), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ - /* .product_name = "Fast Track Ultra 8", */ + /* .product_name = "Fast Track Ultra", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = & (const struct snd_usb_audio_quirk[]) { @@ -1840,11 +1840,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { @@ -1855,7 +1895,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE(0x0763, 0x2081), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ /* .product_name = "Fast Track Ultra 8R", */ .ifnum = QUIRK_ANY_INTERFACE, @@ -1867,11 +1907,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { -- 1.7.0.4 --------------040009020804090202030406--

1 0

[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

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[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!

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