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

[alsa-devel] [RFC] ALSA vs. dedicated char device for a USB Audio Class gadget driver
by Laurent Pinchart 14 May '09

14 May '09

Hi everybody. Linux provides a USB Audio Class (UAC) driver on the USB host side, but a UAC gadget driver on the USB device side is currently missing. Considering plugging in this hole, I'm looking at userspace <-> kernelspace APIs. In a nutshell (let's consider recording, where the device appears to the host as a USB microphone), the kernel driver needs to be fed with audio data from a userspace application and stream those data on USB. Audio data can come from a real microphone (through an ALSA driver), a file, a network socket, ... the driver will not care. I need an API to transfer audio data from userspace to kernelspace. I initially thought about ALSA, but it turns out some assumptions made by ALSA are not fulfilled by my system. One of the most serious problems is that the UAC gadget driver doesn't have any audio clock. The only hardware clock available is the USB device controller interrupts generated at the USB transfer rate, and those are much faster than the audio sample rate. This will cause buffer underruns that I need to handle. I can thing of 3 solutions to solve this issue, and I'd like your opinion on them. 1. Get the userspace application to provide a clock to the driver. This could be done by writing to a /proc file, a special char device, or even by adding an ioctl to ALSA. Every time the driver receives a clock tick it would transfer one period of data. 2. Find out how much data is available in the ALSA ring buffer. I could then throttle USB interrupts when no data is available in the buffer and try again later. ALSA maintains a read and a write pointer (hw_ptr and appl_ptr) that can be accessed through the snd_pcm_*_avail() functions. However, I'm not sure how to handle proper locking, and I don't understand how runtime->boundary interacts with those functions. I've also heard that there's a free-running mode that will make those functions return bogus values. I'd appreciate help on this. 3. Use another API (dedicated character device, socket, ...) to transfer audio data from userspace to kernelspace. ALSA is way too big for my requirements (2 layers of abstraction, architectured around assumptions that make sense for real sound cards but not for my hardware, ...). On the other hand, it would be handy to use existing ALSA applications to driver the device. I'd appreciate comments regarding the API choice. How do you all expect to push data from a userspace application to a UAC driver on the device side ? If ALSA is *the* way, is one of the above solutions practical or is there a better way ? Best regards, Laurent Pinchart

1 0

[alsa-devel] [PATCH] keywest: Get rid of useless i2c_device_name() macro
by Jean Delvare 14 May '09

14 May '09

The i2c_device_name() macro is used only once and doesn't have much value, it hurts redability more than it helps. Get rid of it. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Benjamin Herrenschmidt <benh(a)kernel.crashing.org> --- sound/ppc/keywest.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) --- linux-2.6.30-rc1.orig/sound/ppc/keywest.c 2009-04-10 16:22:08.000000000 +0200 +++ linux-2.6.30-rc1/sound/ppc/keywest.c 2009-04-10 16:25:16.000000000 +0200 @@ -33,10 +33,6 @@ static struct pmac_keywest *keywest_ctx; -#ifndef i2c_device_name -#define i2c_device_name(x) ((x)->name) -#endif - static int keywest_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -56,7 +52,7 @@ static int keywest_attach_adapter(struct if (! keywest_ctx) return -EINVAL; - if (strncmp(i2c_device_name(adapter), "mac-io", 6)) + if (strncmp(adapter->name, "mac-io", 6)) return 0; /* ignored */ memset(&info, 0, sizeof(struct i2c_board_info)); -- Jean Delvare

2 1

[alsa-devel] AC97 reset fail after suspend
by Tomas 'Sleep_Walker' Cech 14 May '09

14 May '09

Hi, I've got following error on Palm LifeDrive and PalmTX after suspend & resume: pxa2xx_ac97_try_warm_reset: warm reset timeout (GSR=0x0) pxa2xx_ac97_read: read error (ac97_reg=0 GSR=0x0) pxa2xx_ac97_try_cold_reset: cold reset timeout (GSR=0x44) pxa2xx_ac97_read: read error (ac97_reg=0 GSR=0x44) WM9712 AC97 reset failed could not reset AC97 codec That completely disable whole AC97 communication, so touchscreen, sound and battery doesn't work any longer. I also tried this with Treo680 (code will be sent soon). HW is very similar, only codec is different - LD/T5/TX use WM9712, Treo680 use WM9713. On Treo680 there is no problem with that. Thanks for any hint about that. Tomas 'Sleep_Walker' Cech Attaching config for LD/TX and dmesg output.

4 10

[alsa-devel] DMA over run on playback
by Jon Smirl 14 May '09

14 May '09

There's a long thread over on the pulse list about glitch free playback. The glitches they are encountering are caused by CPU scheduling latency. They are trying to fix this by setting HZ up to 1000 and constantly polling the audio DMA queue to keep it 99% full. This doesn't seem like the right solution to me. It is fixing the symptom not the cause. The cause is 200-300ms scheduling latency. The source of that needs to be tracked down and fixed in the kernel. But we have to live with the latencies until they are fixed. The strategy of checking the queue at 1000Hz works but it is very inefficient. The underlying problem is that the buffer ALSA is using is too small on systems with 300ms latency. The buffer is just big enough to cover 300ms so they rapidly check and fill it at 1000Hz to ensure that it is full if the 300ms latency strikes. On my hardware with period interrupts ALSA is only checking the buffer at 8Hz. Since I'm checking appl_ptr I know when DMA over runs the buffers. This allows me to insert silence and I could indicated this condition to ALSA if there was a mechanism for doing so. ALSA could use this over run knowledge to measure scheduling latency and adjust the buffering. But the DMA interface between ALSA and the driver has been fixed at stream creation time. There's no way to dynamically alter it (like window size changes in TCP/IP). With networking you get a list of buffers to send. As you send these buffers you mark them sent. The core is free to hand you buffers straight from user space or do copies and use internal ring buffers. The network driver just gets a list of physical addresses to send. This buffer bookkeeping could occur in snd_pcm_period_elapsed(). A dynamic chaining mechanism allows you to alter the buffering mid-stream. If the driver indication a DMA over run error this tells ALSA that it needs to insert another buffer. After a while these errors will stop and ALSA will have measured worst case CPU scheduling latency. From then on it will know the exact size of buffering needed for the kernel it it running on and it can use this knowledge at stream creation time. Now filling the buffer at 8Hz or lower will work and you don't have to spend the power associated with 1000Hz timer interrupts. -- Jon Smirl jonsmirl(a)gmail.com

4 10

[alsa-devel] [PATCH 1/4 v2] pxa2xx-i2s: Proper hw initialization
by Karl Beldan 14 May '09

14 May '09

Make sure we are in a know good state at end of probe : Reset FIFO logic and registers, and make sure REC and RPL functions along with FIFO service are disabled (SACR0_RST enables REC and RPL). Resetting loses current settings so remove reset from stream startup. Now reset occurs only at probe. Signed-off-by: Karl Beldan <karl.beldan(a)mobile-devices.fr> --- sound/soc/pxa/pxa2xx-i2s.c | 17 ++++++++++++++--- 1 files changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 2f4b6e4..59cc4f5 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -106,10 +106,8 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, if (IS_ERR(clk_i2s)) return PTR_ERR(clk_i2s); - if (!cpu_dai->active) { - SACR0 |= SACR0_RST; + if (!cpu_dai->active) SACR0 = 0; - } return 0; } @@ -346,6 +344,19 @@ static int pxa2xx_i2s_probe(struct platform_device *dev) if (ret != 0) clk_put(clk_i2s); + /* + * PXA Developer's Manual: + * If SACR0[ENB] is toggled in the middle of a normal operation, + * the SACR0[RST] bit must also be set and cleared to reset all + * I2S controller registers. + */ + SACR0 = SACR0_RST; + SACR0 = 0; + /* Make sure RPL and REC are disabled */ + SACR1 = SACR1_DRPL | SACR1_DREC; + /* Along with FIFO servicing */ + SAIMR &= ~(SAIMR_RFS | SAIMR_TFS); + return ret; } -- 1.6.3.rc1.34.g0be9b -- Karl

2 2

[alsa-devel] [PATCH] riptide: postfix increment and off by one
by Roel Kluin 14 May '09

14 May '09

With a postfix increment these variables are incremented beyond CMDIF_TIMEOUT / MAX_WRITE_RETRY. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 6f10344..e51a5ef 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -889,7 +889,7 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm, spin_lock_irqsave(&cif->lock, irqflags); while (i++ < CMDIF_TIMEOUT && !IS_READY(cif->hwport)) udelay(10); - if (i >= CMDIF_TIMEOUT) { + if (i > CMDIF_TIMEOUT) { err = -EBUSY; goto errout; } @@ -907,8 +907,10 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm, WRITE_PORT_ULONG(cmdport->data1, cmd); /* write cmd */ if ((flags & RESP) && ret) { while (!IS_DATF(cmdport) && - time++ < CMDIF_TIMEOUT) + time < CMDIF_TIMEOUT) { udelay(10); + time++; + } if (time < CMDIF_TIMEOUT) { /* read response */ ret->retlongs[0] = READ_PORT_ULONG(cmdport->data1); @@ -1454,7 +1456,7 @@ static int snd_riptide_trigger(struct snd_pcm_substream *substream, int cmd) SEND_GPOS(cif, 0, data->id, &rptr); udelay(1); } while (i != rptr.retlongs[1] && j++ < MAX_WRITE_RETRY); - if (j >= MAX_WRITE_RETRY) + if (j > MAX_WRITE_RETRY) snd_printk(KERN_ERR "Riptide: Could not stop stream!"); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: @@ -1783,7 +1785,7 @@ snd_riptide_codec_write(struct snd_ac97 *ac97, unsigned short reg, SEND_SACR(cif, val, reg); SEND_RACR(cif, reg, &rptr); } while (rptr.retwords[1] != val && i++ < MAX_WRITE_RETRY); - if (i == MAX_WRITE_RETRY) + if (i > MAX_WRITE_RETRY) snd_printdd("Write AC97 reg failed\n"); }

2 1

[alsa-devel] WM9712 - Big and little endian format conversion.
by Jon Smirl 14 May '09

14 May '09

I have three areas specifying audio formats. The WM9712 AC97 codec is S16_LE. My DMA hardware is S32_BE. All samples need to be in 32b words even if they are 8,16,24b. The cpu's AC97 system sends the first 20 big endian bits down the AC97 link in the proper order. How can I force the DMA engine setting to override what the codec is saying? Or should the codec be changed to specify all bit lengths and both endians? My DMA platform: static const struct snd_pcm_hardware psc_dma_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_S32_BE, .rate_min = 8000, .rate_max = 48000, .channels_min = 1, .channels_max = 2, .period_bytes_max = 1024 * 1024, .period_bytes_min = 32, .periods_min = 2, .periods_max = 256, .buffer_bytes_max = 2 * 1024 * 1024, .fifo_size = 512, }; CPU DAI: static struct snd_soc_dai psc_ac97_dai_template[] = { { .name = "%s analog", .suspend = psc_ac97_suspend, .resume = psc_ac97_resume, .playback = { .channels_min = 1, .channels_max = 6, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, }, .capture = { .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, }, .ops = &psc_ac97_analog_ops, }, Codec: struct snd_soc_dai wm9712_dai[] = { { .name = "AC97 HiFi", .ac97_control = 1, .playback = { .stream_name = "HiFi Playback", .channels_min = 1, .channels_max = 2, .rates = WM9712_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "HiFi Capture", .channels_min = 1, .channels_max = 2, .rates = WM9712_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &wm9712_dai_ops_hifi, }, { -- Jon Smirl jonsmirl(a)gmail.com

2 7

[alsa-devel] Avoiding buffer underruns in the hardware interrupt handler
by Laurent Pinchart 14 May '09

14 May '09

Hi everybody. Disclaimer: I'm new with ALSA development and I'm trying to digest the large amount of information I'm overflowed with, so please bear with possibly stupid help requests. I'm writing an ALSA driver for a playback device. Unlike most sound cards, this device uses private memory for its audio buffer. I thus need to manually copy data from the ALSA buffer to the device buffer in the driver. The device audio buffer is quite small, and for this reason I'd rather not use the copy/silence callbacks, but copy data from the ALSA buffer to the device in interrupt. The device might generate interrupts a bit faster than what would be expected from sampling rate, so I need to check in the interrupt handler how much data the ALSA buffer holds. To do so I need to maintain a "hardware pointer" in the driver and to access the "user pointer" on the other side of the buffer. I haven't found a clear API to access that second pointer. What is the best/preferred way to find out how much data the ALSA buffer holds ? I suppose I will need to perform some locking when accessing the "user pointer", I will also appreciate advices regarding which mutex/spinlock I should use for that purpose. Best regards, Laurent Pinchart

1 0

[alsa-devel] [PATCH 4/4 v2] pxa2xx-i2s: Fix suspend/resume
by Karl Beldan 13 May '09

13 May '09

pxa2xx_i2s_resume is : - unconditionnaly setting SACR0_ENB - unsetting SACR0_ENB in saved SACR0 pxa_i2s.sacr0 fix these. In pxa2xx_i2s_{resume,suspend}, save/restore registers even when !dai->active. Signed-off-by: Karl Beldan <karl.beldan(a)mobile-devices.fr> --- sound/soc/pxa/pxa2xx-i2s.c | 11 +++-------- 1 files changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index a423637..efe499e 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -262,9 +262,6 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, #ifdef CONFIG_PM static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai) { - if (!dai->active) - return 0; - /* store registers */ pxa_i2s.sacr0 = SACR0; pxa_i2s.sacr1 = SACR1; @@ -279,16 +276,14 @@ static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai) static int pxa2xx_i2s_resume(struct snd_soc_dai *dai) { - if (!dai->active) - return 0; - pxa_i2s_wait(); - SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB; + SACR0 = pxa_i2s.sacr0 & ~SACR0_ENB; SACR1 = pxa_i2s.sacr1; SAIMR = pxa_i2s.saimr; SADIV = pxa_i2s.sadiv; - SACR0 |= SACR0_ENB; + + SACR0 = pxa_i2s.sacr0; return 0; } -- 1.6.3.rc1.34.g0be9b -- Karl

1 0

[alsa-devel] [PATCH 3/4 v2] pxa2xx-i2s: Fix inappropriate release of i2s clock
by Karl Beldan 13 May '09

13 May '09

i2s_clk is 'put' for no reason in pxa2xx_i2s_shutdown. Now we 'get' i2s_clk at probe and 'put' it at driver removal or when probe fails. Signed-off-by: Karl Beldan <karl.beldan(a)mobile-devices.fr> --- sound/soc/pxa/pxa2xx-i2s.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index a461c8d..a423637 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -257,8 +257,6 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, pxa_i2s_wait(); clk_disable(clk_i2s); } - - clk_put(clk_i2s); } #ifdef CONFIG_PM -- 1.6.3.rc1.34.g0be9b -- Karl

1 0

[alsa-devel] [PATCH 2/4 v2] pxa2xx-i2s: Handle SACR1_DRPL and SACR1_DREC separately
by Karl Beldan 13 May '09

13 May '09

- hw_params enables both RPL and REC functions each time : Enable the appropriate function in pxa2xx_i2s_trigger. - pxa2xx_i2s_shutdown disables i2s anytime one of RPL or REC function is off : Turn it off only when both functions are off. Signed-off-by: Karl Beldan <karl.beldan(a)mobile-devices.fr> --- sound/soc/pxa/pxa2xx-i2s.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 59cc4f5..a461c8d 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -176,9 +176,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, /* is port used by another stream */ if (!(SACR0 & SACR0_ENB)) { - SACR0 = 0; - SACR1 = 0; if (pxa_i2s.master) SACR0 |= SACR0_BCKD; @@ -224,6 +222,10 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + SACR1 &= ~SACR1_DRPL; + else + SACR1 &= ~SACR1_DREC; SACR0 |= SACR0_ENB; break; case SNDRV_PCM_TRIGGER_RESUME: @@ -250,7 +252,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, SAIMR &= ~SAIMR_RFS; } - if (SACR1 & (SACR1_DREC | SACR1_DRPL)) { + if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) { SACR0 &= ~SACR0_ENB; pxa_i2s_wait(); clk_disable(clk_i2s); -- 1.6.3.rc1.34.g0be9b -- Karl

1 0

[alsa-devel] [PATCH 0/4 v2] pxa2xx-i2s: Rework
by Karl Beldan 13 May '09

13 May '09

Following is a rework of the i2s code for the pxa. I tested it with the ac97 clock as the master clock. Feedbacks are welcome. Diff with initial version: - hw init from 1/4 is moved from pxa2xx_i2s_init to pxa2xx_i2s_probe - moved 'removal of hw reset from pxa2xx_i2s_startup' from 3/4 to 1/4v2 - moved 'clk_put(clk_i2s) removal' from 2/4 to 3/4v2 - 3/4 was also removing FIFOs flushes, 3/4v2 consists of 'clk_put(clk_i2s) removal' only, -- Karl

1 1

[alsa-devel] [PATCH 4/4] pxa2xx-i2s: Fix suspend/resume
by Karl Beldan 13 May '09

13 May '09

pxa2xx_i2s_resume is : - unconditionnaly setting SACR0_ENB - unsetting SACR0_ENB in saved SACR0 pxa_i2s.sacr0 fix these. In pxa2xx_i2s_{resume,suspend}, save/restore registers even when !dai->active. Signed-off-by: Karl Beldan <karl.beldan(a)mobile-devices.fr> --- sound/soc/pxa/pxa2xx-i2s.c | 11 +++-------- 1 files changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index a423637..efe499e 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -262,9 +262,6 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, #ifdef CONFIG_PM static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai) { - if (!dai->active) - return 0; - /* store registers */ pxa_i2s.sacr0 = SACR0; pxa_i2s.sacr1 = SACR1; @@ -279,16 +276,14 @@ static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai) static int pxa2xx_i2s_resume(struct snd_soc_dai *dai) { - if (!dai->active) - return 0; - pxa_i2s_wait(); - SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB; + SACR0 = pxa_i2s.sacr0 & ~SACR0_ENB; SACR1 = pxa_i2s.sacr1; SAIMR = pxa_i2s.saimr; SADIV = pxa_i2s.sadiv; - SACR0 |= SACR0_ENB; + + SACR0 = pxa_i2s.sacr0; return 0; } -- 1.6.3.rc1.34.g0be9b -- Karl

1 0

[alsa-devel] [PATCH 3/4] pxa2xx-i2s: Fix inappropriate release of i2s clock
by Karl Beldan 13 May '09

13 May '09

i2s_clk is 'put' for no reason in pxa2xx_i2s_shutdown. Now we 'get' i2s_clk at probe and 'put' it at driver removal or when probe fails. Signed-off-by: Karl Beldan <karl.beldan(a)mobile-devices.fr> --- sound/soc/pxa/pxa2xx-i2s.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index a461c8d..a423637 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -257,8 +257,6 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, pxa_i2s_wait(); clk_disable(clk_i2s); } - - clk_put(clk_i2s); } #ifdef CONFIG_PM -- 1.6.3.rc1.34.g0be9b -- Karl

1 0

[alsa-devel] [PATCH 2/4] pxa2xx-i2s: Handle SACR1_DRPL and SACR1_DREC separately
by Karl Beldan 13 May '09

13 May '09

- hw_params enables both RPL and REC functions each time : Enable the appropriate function in pxa2xx_i2s_trigger. - pxa2xx_i2s_shutdown disables i2s anytime one of RPL or REC function is off : Turn it off only when both functions are off. Signed-off-by: Karl Beldan <karl.beldan(a)mobile-devices.fr> --- sound/soc/pxa/pxa2xx-i2s.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 59cc4f5..a461c8d 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -176,9 +176,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, /* is port used by another stream */ if (!(SACR0 & SACR0_ENB)) { - SACR0 = 0; - SACR1 = 0; if (pxa_i2s.master) SACR0 |= SACR0_BCKD; @@ -224,6 +222,10 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + SACR1 &= ~SACR1_DRPL; + else + SACR1 &= ~SACR1_DREC; SACR0 |= SACR0_ENB; break; case SNDRV_PCM_TRIGGER_RESUME: @@ -250,7 +252,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, SAIMR &= ~SAIMR_RFS; } - if (SACR1 & (SACR1_DREC | SACR1_DRPL)) { + if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) { SACR0 &= ~SACR0_ENB; pxa_i2s_wait(); clk_disable(clk_i2s); -- 1.6.3.rc1.34.g0be9b -- Karl

1 0

[alsa-devel] [PATCH 1/4] pxa2xx-i2s: Proper hw initialization
by Karl Beldan 13 May '09

13 May '09

Make sure we are in a know good state at end of probe : Reset FIFO logic and registers, and make sure REC and RPL functions along with FIFO service are disabled (SACR0_RST enables REC and RPL). Resetting loses current settings so remove reset from stream startup. Now reset occurs only at probe. Signed-off-by: Karl Beldan <karl.beldan(a)mobile-devices.fr> --- sound/soc/pxa/pxa2xx-i2s.c | 17 ++++++++++++++--- 1 files changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 2f4b6e4..59cc4f5 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -106,10 +106,8 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, if (IS_ERR(clk_i2s)) return PTR_ERR(clk_i2s); - if (!cpu_dai->active) { - SACR0 |= SACR0_RST; + if (!cpu_dai->active) SACR0 = 0; - } return 0; } @@ -346,6 +344,19 @@ static int pxa2xx_i2s_probe(struct platform_device *dev) if (ret != 0) clk_put(clk_i2s); + /* + * PXA Developer's Manual: + * If SACR0[ENB] is toggled in the middle of a normal operation, + * the SACR0[RST] bit must also be set and cleared to reset all + * I2S controller registers. + */ + SACR0 = SACR0_RST; + SACR0 = 0; + /* Make sure RPL and REC are disabled */ + SACR1 = SACR1_DRPL | SACR1_DREC; + /* Along with FIFO servicing */ + SAIMR &= ~(SAIMR_RFS | SAIMR_TFS); + return ret; } -- 1.6.3.rc1.34.g0be9b -- Karl

1 0

[alsa-devel] [PATCH 0/1] Add AIC3106 support.
by Ambrose, Martin 13 May '09

13 May '09

Hello, I patched in support for the AIC3106 codec which is the IC on the DA830 EVM. This patch applies to the ASoC tree: ddc4097b77bbb227851a44287acb2fb8a9896cc1 But it also applies cleanly to the Davinci tree: Unfortunately there are missing pieces needed (e.g. edma/mcasp) for this board on the Davinci tree to test directly. But it passes the smoke test of building and running on the Davinci EVM. -Martin

2 3

[alsa-devel] asoundconf removed from alsa-utils?
by Susan Cragin 13 May '09

13 May '09

The message says that asoundconf is not present because alsa-utils is not installed. However, I have just installed the latest. 20090512. And the following alsa-utils work: arecord works, as well as alsamixer and alsactl. I hesitate to even report this, because I'm on karmic with 2.6.30-5-generic kernel, and I know there are bugs, and that I am pre-alpha. So this is FYI -- no response needed or expected. Susan

3 2

[alsa-devel] emu10k1 TRAM access ?
by mAdVax 13 May '09

13 May '09

Hello I would like to send some data to a DSP patch. I can use alsa controls (easy) but I need more : TRAM and lookup table. I read many lines of code, but all this is still quite obscure... Here are 3 questions : How can I fill the TRAM ? (from user space) How can I know which TRAM area has been allocated to a given patch ? Do 'lookup tables' work ? Some code samples will be greatly appreciated :) Thanks in advance. Xavier

1 0

[alsa-devel] ALSA 1.0.20 PCM Capture Problem (Does not happen on ALSA 1.0.19)
by Robert Vincent Krakora 13 May '09

13 May '09

Hello, I am utilizing an external mic on my machine in conjunction with a VF0560 Live! Cam Optia AF webcam. I am running CentOS 5.3 with the kernel listed below and ALSA 1.0.20. I using the machine to stream video with VLC 9.9 (see further below). VLC spits out the following on the console of my machine whist capturing A/V and serving it up on the network...ALSA 1.0.19 did not exhibit this problem while streaming it did lock up after 20 minutes or so (audio could be heard on the VLC or mplayer client with ALSA 1.0.19...just bits and pieces of audio every now and then with ALSA 1.0.20. V4L is not the culprit here as video streams and can be received without incident if I do not also capture audio. The mic is plugged into the onboard sound on the motherboard. I have not yet tried my USB mic with ALSA 1.0.20, but I will try it later tonight. Any and all help would be greatly appreciate. Thanks in advance. [00000473] main audio output warning: audio drift is too big (-146765), clearing out [00000473] main audio output warning: mixer start isn't output start (-29715) [00000473] main audio output warning: audio drift is too big (-150692), clearing out [00000473] main audio output warning: mixer start isn't output start (-30980) [00000493] main spu text debug: looking for text renderer module: 3 candidates [00000493] freetype spu text error: failed to load font file /usr/share/fonts/truetype/freefont/FreeSerifBold.ttf [00000493] main spu text debug: using text renderer module "svg" [00000493] main spu text debug: TIMER module_Need() : 29.750 ms - Total 29.750 ms / 1 intvls (Avg 29.750 ms) [00000492] main blend debug: looking for video blending module: 1 candidate [00000492] blend blend debug: chroma: YUVA -> I420 [00000492] main blend debug: using video blending module "blend" [00000492] main blend debug: TIMER module_Need() : 0.433 ms - Total 0.433 ms / 1 intvls (Avg 0.433 ms) [00000473] main audio output warning: audio drift is too big (-147017), clearing out [00000473] main audio output warning: mixer start isn't output start (-29763) [00000473] main audio output warning: audio drift is too big (-147157), clearing out [00000473] main audio output warning: mixer start isn't output start (-30302) [00000473] main audio output warning: audio drift is too big (-150434), clearing out [00000473] main audio output warning: mixer start isn't output start (-30931) [00000473] main audio output warning: buffer is 114796 in advance, triggering downsampling [00000473] main audio output warning: audio drift is too big (-261628), clearing out [00000473] main audio output warning: timing screwed, stopping resampling [00000473] main audio output warning: mixer start isn't output start (-53788) [00000473] main audio output warning: audio drift is too big (-146580), clearing out [00000473] main audio output warning: mixer start isn't output start (-29711) [00000473] main audio output warning: audio drift is too big (-150686), clearing out [00000473] main audio output warning: mixer start isn't output start (-30979) [00000473] main audio output warning: audio drift is too big (-146733), clearing out [00000473] main audio output warning: mixer start isn't output start (-29708) [00000473] main audio output warning: audio drift is too big (-146978), clearing out [00000473] main audio output warning: mixer start isn't output start (-30267) [00000473] main audio output warning: audio drift is too big (-150569), clearing out [00000473] main audio output warning: mixer start isn't output start (-30445) [00000473] main audio output warning: buffer is 114650 in advance, triggering downsampling [00000473] main audio output warning: audio drift is too big (-261447), clearing out [00000473] main audio output warning: timing screwed, stopping resampling [00000473] main audio output warning: mixer start isn't output start (-54265) [00000473] main audio output warning: audio drift is too big (-146518), clearing out [00000473] main audio output warning: mixer start isn't output start (-29699) [00000473] main audio output warning: audio drift is too big (-146731), clearing out [00000473] main audio output warning: mixer start isn't output start (-30220) [00000473] main audio output warning: audio drift is too big (-150880), clearing out [00000473] main audio output warning: mixer start isn't output start (-30505) [00000473] main audio output warning: audio drift is too big (-146644), clearing out [00000473] main audio output warning: mixer start isn't output start (-30203) [00000473] main audio output warning: audio drift is too big (-146701), clearing out [00000473] main audio output warning: mixer start isn't output start (-29702) [00000473] main audio output warning: audio drift is too big (-150683), clearing out [00000473] main audio output warning: mixer start isn't output start (-30979) [00000473] main audio output warning: audio drift is too big (-146706), clearing out [00000473] main audio output warning: mixer start isn't output start (-29703) [00000473] main audio output warning: audio drift is too big (-146916), clearing out [00000473] main audio output warning: mixer start isn't output start (-30255) [00000473] main audio output warning: audio drift is too big (-150819), clearing out [00000473] main audio output warning: mixer start isn't output start (-31005) [00000473] main audio output warning: audio drift is too big (-147003), clearing out [00000473] main audio output warning: mixer start isn't output start (-29760) [00000473] main audio output warning: audio drift is too big (-146886), clearing out [00000473] main audio output warning: mixer start isn't output start (-30250) [00000473] main audio output warning: audio drift is too big (-150552), clearing out [root@vizioroom105 ~]# uname -a Linux vizioroom105.msgnet.com 2.6.27.10 #2 SMP Thu Jan 22 09:09:44 EST 2009 x86_64 x86_64 x86_64 GNU/Linux [root@vizioroom105 ~]# arecord -l **** List of CAPTURE Hardware Devices **** card 0: AF [VF0560 Live! Cam Optia AF], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: SB [HDA ATI SB], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: SB [HDA ATI SB], device 1: ALC883 Digital [ALC883 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: SB [HDA ATI SB], device 2: ALC883 Analog [ALC883 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 [root@vizioroom105 ~]# su silentm -c "/usr/bin/vcamera -vvvvv -I dummy -V dummy --dscp=0x01 v4l2:// :v4l2-dev=/dev/video0 :v4l2-adev=hw.1,0 :v4l2-width=720 :v4l2-height=480 :v4l2-standard=0 --sout '#transcode{vcodec=mp4v,vb=6000,fps=30,scale=1,acodec=mpga,ab=32,channels=1}:duplicate{dst=rtp{mux=ts,port=1234,sdp=rtsp://192.168.1.179:8080/test.sdp}}' vlc://quit" Best Regards, -- Rob Krakora Senior Software Engineer MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext. 206 (317)663-0808 Fax

3 2

[alsa-devel] patch to support Samsung NC10 mini notebook
by Chris Pockele 13 May '09

13 May '09

Hello, The attached file is a patch to support the sound hardware in the Samsung NC10 mini notebook. The Samsung NC10 is and Intel Atom-based notebook with Realtek ALC272 HDA codecs. Sound is supported partly in Linux depending on ALSA version, but there were always some issues (wrong mixer behaviour, PC beep always muted, ...). This patch tries to solve most of these issues. I updated the patch so it would work using the alsa-driver-20090509 snapshot from ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ (originally developed from alsa-driver-20090416 snapshot). Fixes: - mute internal speaker / microphone when plugging in external headphone / microphone - stereo capture from external microphone - separate volume controls for headphone and internal speakers + "Master" for both - "PC Beep" mixer control - "Mic boost" and "External mic boost" controls - ... A few issues remaining: - Recording PCM output does not work yet. It works on the version of Windows that comes with the NC10. - When choosing recording source "Internal Mic" while an external microphone is plugged in, the internal microphone will be used instead of the external one, as it should. However, when plugging in the external microphone after choosing the recording source setting, it WILL override the internal microphone. When replying, please note I am not on list. Regards, Chris http://users.skynet.be/chrisp/linux/

2 4

[alsa-devel] [PATCH] aoa: remove driver_data direct access of struct device
by Roel Kluin 13 May '09

13 May '09

To avoid direct access to the driver_data pointer in struct device, the functions dev_get_drvdata() and dev_set_drvdata() should be used. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index fbf5c93..586965f 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -1037,7 +1037,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) } ldev->selfptr_headphone.ptr = ldev; ldev->selfptr_lineout.ptr = ldev; - sdev->ofdev.dev.driver_data = ldev; + dev_set_drvdata(&sdev->ofdev.dev, ldev); list_add(&ldev->list, &layouts_list); layouts_list_items++; @@ -1081,7 +1081,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) static int aoa_fabric_layout_remove(struct soundbus_dev *sdev) { - struct layout_dev *ldev = sdev->ofdev.dev.driver_data; + struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev); int i; for (i=0; i<MAX_CODECS_PER_BUS; i++) { @@ -1114,7 +1114,7 @@ static int aoa_fabric_layout_remove(struct soundbus_dev *sdev) #ifdef CONFIG_PM static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t state) { - struct layout_dev *ldev = sdev->ofdev.dev.driver_data; + struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev); if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off) ldev->gpio.methods->all_amps_off(&ldev->gpio); @@ -1124,7 +1124,7 @@ static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t sta static int aoa_fabric_layout_resume(struct soundbus_dev *sdev) { - struct layout_dev *ldev = sdev->ofdev.dev.driver_data; + struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev); if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off) ldev->gpio.methods->all_amps_restore(&ldev->gpio); diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 418c84c..4e3b819 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -358,14 +358,14 @@ static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) return -ENODEV; } - dev->ofdev.dev.driver_data = control; + dev_set_drvdata(&dev->ofdev.dev, control); return 0; } static int i2sbus_remove(struct macio_dev* dev) { - struct i2sbus_control *control = dev->ofdev.dev.driver_data; + struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev); struct i2sbus_dev *i2sdev, *tmp; list_for_each_entry_safe(i2sdev, tmp, &control->list, item) @@ -377,7 +377,7 @@ static int i2sbus_remove(struct macio_dev* dev) #ifdef CONFIG_PM static int i2sbus_suspend(struct macio_dev* dev, pm_message_t state) { - struct i2sbus_control *control = dev->ofdev.dev.driver_data; + struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev); struct codec_info_item *cii; struct i2sbus_dev* i2sdev; int err, ret = 0; @@ -407,7 +407,7 @@ static int i2sbus_suspend(struct macio_dev* dev, pm_message_t state) static int i2sbus_resume(struct macio_dev* dev) { - struct i2sbus_control *control = dev->ofdev.dev.driver_data; + struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev); struct codec_info_item *cii; struct i2sbus_dev* i2sdev; int err, ret = 0;

3 5

[alsa-devel] pxa2xx-pcm and dma_request
by Robert Jarzmik 13 May '09

13 May '09

Hi Mark and Dimitry, I've been playing for some time with DMA on the PXA series. I was wondering if the behaviour of pxa2xx-pcm is correct. As far as I understand, in an asoc context, the __pxa2xx_pcm_open() is called, while in "arm context", pxa2xx_pcm_open() is called, which calls __pxa2xx_pcm_open(). This was introduced by commit a6d77317678148c973bb0131cc5a3a772f756d23 I think. One main difference is that in "arm context", pxa_dma_request() is called, while in "asoc context", it is not. I'd like one of you to cross-check, as I must admit I'm not following all the impacts of the pcm code split-up. Cheers. -- Robert

3 2

[alsa-devel] aplay as a Zombie shows up in systemsupervisor when starting Totem or Totem-xine.
by Kristoffer Grundström 13 May '09

13 May '09

I think I've identified the issue with my sound as aplay being a Zombie everytime I'm starting totem or totem-xine. When restarting ALSA with service alsa restart as root I get what's shown in this link: http://pastebin.mandriva.com/11255 3319 is Pulseaudio. Pulseaudio disappears when closing totem or totem-xine. This problem has to do with my earlier mentioned bugreport in this link: https://qa.mandriva.com/show_bug.cgi?id=49048 Thank you. /Kristoffer

1 0

[alsa-devel] [PATCH 2/4] pxa2xx-i2s: Handle SACR1_DRPL and SACR1_DREC separately
by Karl Beldan 13 May '09

13 May '09

- hw_params enables both RPL and REC functions each time, enable the appropriate function in pxa2xx_i2s_trigger. - pxa2xx_i2s_shutdown disables i2s anytime one of RPL or REC function is off, turn it off only when both functions are off. - do not put the clk_i2s for no reason in pxa2xx_i2s_shutdown. Signed-off-by: Karl Beldan <karl.beldan(a)mobile-devices.fr> --- sound/soc/pxa/pxa2xx-i2s.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 52862dc..0e53d07 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -178,9 +178,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, /* is port used by another stream */ if (!(SACR0 & SACR0_ENB)) { - SACR0 = 0; - SACR1 = 0; if (pxa_i2s.master) SACR0 |= SACR0_BCKD; @@ -226,6 +224,10 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + SACR1 &= ~SACR1_DRPL; + else + SACR1 &= ~SACR1_DREC; SACR0 |= SACR0_ENB; break; case SNDRV_PCM_TRIGGER_RESUME: @@ -252,13 +254,11 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, SAIMR &= ~SAIMR_RFS; } - if (SACR1 & (SACR1_DREC | SACR1_DRPL)) { + if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) { SACR0 &= ~SACR0_ENB; pxa_i2s_wait(); clk_disable(clk_i2s); } - - clk_put(clk_i2s); } #ifdef CONFIG_PM -- 1.6.3.rc1.34.g0be9b -- Karl

3 14

[alsa-devel] storm of emu10k1: unhandled interrupt: 0x00400000
by Daniel Drake 13 May '09

13 May '09

Hi, Martin von Gagern reported a bug in the emu10k1 driver while running 2.6.29: https://bugs.gentoo.org/show_bug.cgi?id=266657 This message is logged seemingly infinitely, to the point where the system became unusable. emu10k1: unhandled interrupt: 0x00400000 This is INTE_ADLIBENABLE. The bug is a one-off and unreproducible, this mail is really just for information only... or would people be happy with this flag being added to the "mask the interrupts that we don't handle" code below the code that prints the "unhandled interrupt" message? Daniel

1 0

[alsa-devel] [PATCH] control, hw, rawmidi: accept control device node as card identifier when opening
by Lennart Poettering 12 May '09

12 May '09

Heya! Traditionally low-level ALSA devices are opened with device strings like 'hw:CARD' where CARD is either a numeric index or a card name. This patch introduces a third syntax that allows you to specify the file name of the control device node. The purpose for this is to make ALSA a bit more like other Linux devices and allow implementaiton of udev device symlinks similar to how disks are currently handled, with /dev/disk/by-path/xxxx /dev/disk/by-id/xxxx and so on. With this patch this line: aplay -f CD -D hw:/dev/snd/controlC0 < /dev/urandom becomes equivalent to this line: aplay -f CD -D hw:0 < /dev/urandom A more useful example is this: aplay -f CD -D hw:/dev/snd/by-path/pci-0000:00:1a.7-usb-0:2:1.2 (this line requires a a few additional udev rules) This only works for control devices nodes, not for PCM or any other device nodes. Since the control device node is used as 'entry point' when opening PCM devices this is no limitation. Lennart --- include/pcm_plugin.h | 4 ++- src/control/cards.c | 20 ++++++++++-- src/control/control_hw.c | 69 +++++++++++++++++++++++++++++++++---------- src/control/control_local.h | 3 +- src/pcm/pcm_hw.c | 29 ++++++++++++------ src/rawmidi/rawmidi_hw.c | 30 ++++++++++++------ src/rawmidi/rawmidi_local.h | 6 ++- 7 files changed, 118 insertions(+), 43 deletions(-) diff --git a/include/pcm_plugin.h b/include/pcm_plugin.h index eea1d82..758144e 100644 --- a/include/pcm_plugin.h +++ b/include/pcm_plugin.h @@ -1,3 +1,4 @@ +/* -*- Mode: C; tab-width: 8; indent-tabs-mode: t; c-basic-offset: 8 -*- */ /** * \file include/pcm_plugin.h * \brief Common PCM plugin code @@ -66,7 +67,8 @@ typedef int snd_pcm_route_ttable_entry_t; * Hardware plugin */ int snd_pcm_hw_open(snd_pcm_t **pcmp, const char *name, - int card, int device, int subdevice, + int card, const char *control_device_node, + int device, int subdevice, snd_pcm_stream_t stream, int mode, int mmap_emulation, int sync_ptr_ioctl); int _snd_pcm_hw_open(snd_pcm_t **pcmp, const char *name, diff --git a/src/control/cards.c b/src/control/cards.c index 4d2c739..912e8c8 100644 --- a/src/control/cards.c +++ b/src/control/cards.c @@ -1,3 +1,4 @@ +/* -*- Mode: C; tab-width: 8; indent-tabs-mode: t; c-basic-offset: 8 -*- */ /** * \file control/cards.c * \brief Basic Soundcard Operations @@ -127,12 +128,25 @@ int snd_card_get_index(const char *string) return card; return err; } + + if (string[0] == '/') { + err = snd_ctl_hw_open(&handle, NULL, -1, string, 0); + + if (err < 0) + return err; + + card = snd_ctl_hw_card(handle); + snd_ctl_close(handle); + + return card; + } + for (card = 0; card < 32; card++) { #ifdef SUPPORT_ALOAD if (! snd_card_load(card)) continue; #endif - if (snd_ctl_hw_open(&handle, NULL, card, 0) < 0) + if (snd_ctl_hw_open(&handle, NULL, card, NULL, 0) < 0) continue; if (snd_ctl_card_info(handle, &info) < 0) { snd_ctl_close(handle); @@ -159,7 +173,7 @@ int snd_card_get_name(int card, char **name) if (name == NULL) return -EINVAL; - if ((err = snd_ctl_hw_open(&handle, NULL, card, 0)) < 0) + if ((err = snd_ctl_hw_open(&handle, NULL, card, NULL, 0)) < 0) return err; if ((err = snd_ctl_card_info(handle, &info)) < 0) { snd_ctl_close(handle); @@ -186,7 +200,7 @@ int snd_card_get_longname(int card, char **name) if (name == NULL) return -EINVAL; - if ((err = snd_ctl_hw_open(&handle, NULL, card, 0)) < 0) + if ((err = snd_ctl_hw_open(&handle, NULL, card, NULL, 0)) < 0) return err; if ((err = snd_ctl_card_info(handle, &info)) < 0) { snd_ctl_close(handle); diff --git a/src/control/control_hw.c b/src/control/control_hw.c index e9a6be2..3cead5e 100644 --- a/src/control/control_hw.c +++ b/src/control/control_hw.c @@ -1,4 +1,5 @@ -/* +/* -*- Mode: C; tab-width: 8; indent-tabs-mode: t; c-basic-offset: 8 -*- + * * Control Interface - Hardware * Copyright (c) 1998,1999,2000 by Jaroslav Kysela <perex(a)perex.cz> * Copyright (c) 2000 by Abramo Bagnara <abramo(a)alsa-project.org> @@ -49,6 +50,12 @@ typedef struct { } snd_ctl_hw_t; #endif /* DOC_HIDDEN */ +int snd_ctl_hw_card(snd_ctl_t *handle) { + snd_ctl_hw_t *hw = handle->private_data; + + return hw->card; +} + static int snd_ctl_hw_close(snd_ctl_t *handle) { snd_ctl_hw_t *hw = handle->private_data; @@ -361,7 +368,7 @@ static const snd_ctl_ops_t snd_ctl_hw_ops = { .read = snd_ctl_hw_read, }; -int snd_ctl_hw_open(snd_ctl_t **handle, const char *name, int card, int mode) +int snd_ctl_hw_open(snd_ctl_t **handle, const char *name, int card, const char *devnode, int mode) { int fd, ver; char filename[sizeof(SNDRV_FILE_CONTROL) + 10]; @@ -370,13 +377,23 @@ int snd_ctl_hw_open(snd_ctl_t **handle, const char *name, int card, int mode) snd_ctl_hw_t *hw; int err; - *handle = NULL; + *handle = NULL; - if (CHECK_SANITY(card < 0 || card >= 32)) { + if (CHECK_SANITY(!!devnode == (card >= 0))) { + SNDMSG("Specified both card index and device node"); + return -EINVAL; + } + + if (CHECK_SANITY(card >= 32)) { SNDMSG("Invalid card index %d", card); return -EINVAL; } - sprintf(filename, SNDRV_FILE_CONTROL, card); + + if (!devnode) { + sprintf(filename, SNDRV_FILE_CONTROL, card); + devnode = filename; + } + if (mode & SND_CTL_READONLY) fmode = O_RDONLY; else @@ -385,13 +402,15 @@ int snd_ctl_hw_open(snd_ctl_t **handle, const char *name, int card, int mode) fmode |= O_NONBLOCK; if (mode & SND_CTL_ASYNC) fmode |= O_ASYNC; - fd = snd_open_device(filename, fmode); - if (fd < 0) { + fd = snd_open_device(devnode, fmode); + if (fd < 0 && card >= 0) { snd_card_load(card); - fd = snd_open_device(filename, fmode); - if (fd < 0) - return -errno; + fd = snd_open_device(devnode, fmode); } + + if (fd < 0) + return -errno; + #if 0 /* * this is bogus, an application have to care about open filedescriptors @@ -412,6 +431,19 @@ int snd_ctl_hw_open(snd_ctl_t **handle, const char *name, int card, int mode) close(fd); return -SND_ERROR_INCOMPATIBLE_VERSION; } + + if (card < 0) { + snd_ctl_card_info_t info; + + if (ioctl(fd, SNDRV_CTL_IOCTL_CARD_INFO, &info) < 0) { + err = -errno; + close(fd); + return err; + } + + card = info.card; + } + hw = calloc(1, sizeof(snd_ctl_hw_t)); if (hw == NULL) { close(fd); @@ -437,7 +469,7 @@ int _snd_ctl_hw_open(snd_ctl_t **handlep, char *name, snd_config_t *root ATTRIBU { snd_config_iterator_t i, next; long card = -1; - const char *str; + const char *str, *devnode = NULL; int err; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -454,16 +486,21 @@ int _snd_ctl_hw_open(snd_ctl_t **handlep, char *name, snd_config_t *root ATTRIBU err = snd_config_get_string(n, &str); if (err < 0) return -EINVAL; - card = snd_card_get_index(str); - if (card < 0) - return card; + if (str[0] == '/') + devnode = name; + else { + card = snd_card_get_index(str); + if (card < 0) + return card; + } + } continue; } return -EINVAL; } - if (card < 0) + if (card < 0 && !devnode) return -EINVAL; - return snd_ctl_hw_open(handlep, name, card, mode); + return snd_ctl_hw_open(handlep, name, card, devnode, mode); } SND_DLSYM_BUILD_VERSION(_snd_ctl_hw_open, SND_CONTROL_DLSYM_VERSION); diff --git a/src/control/control_local.h b/src/control/control_local.h index fd9f941..5088b5f 100644 --- a/src/control/control_local.h +++ b/src/control/control_local.h @@ -95,6 +95,7 @@ struct _snd_hctl { int snd_ctl_new(snd_ctl_t **ctlp, snd_ctl_type_t type, const char *name); int _snd_ctl_poll_descriptor(snd_ctl_t *ctl); #define _snd_ctl_async_descriptor _snd_ctl_poll_descriptor -int snd_ctl_hw_open(snd_ctl_t **handle, const char *name, int card, int mode); +int snd_ctl_hw_open(snd_ctl_t **handle, const char *name, int card, const char *devnode, int mode); +int snd_ctl_hw_card(snd_ctl_t *handle); int snd_ctl_shm_open(snd_ctl_t **handlep, const char *name, const char *sockname, const char *sname, int mode); int snd_ctl_async(snd_ctl_t *ctl, int sig, pid_t pid); diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index c46d14f..59734a1 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -1,3 +1,4 @@ +/* -*- Mode: C; tab-width: 8; indent-tabs-mode: t; c-basic-offset: 8 -*- */ /** * \file pcm/pcm_hw.c * \ingroup PCM_Plugins @@ -1260,7 +1261,8 @@ int snd_pcm_hw_open_fd(snd_pcm_t **pcmp, const char *name, * \brief Creates a new hw PCM * \param pcmp Returns created PCM handle * \param name Name of PCM - * \param card Number of card + * \param card Number of card (must be -1 if control_device_node is non-NULL) + * \param control_device_node Control device node in the file system (must be NULL if card is != -1) * \param device Number of device * \param subdevice Number of subdevice * \param stream PCM Stream @@ -1273,7 +1275,8 @@ int snd_pcm_hw_open_fd(snd_pcm_t **pcmp, const char *name, * changed in future. */ int snd_pcm_hw_open(snd_pcm_t **pcmp, const char *name, - int card, int device, int subdevice, + int card, const char *control_device_node, + int device, int subdevice, snd_pcm_stream_t stream, int mode, int mmap_emulation ATTRIBUTE_UNUSED, int sync_ptr_ioctl) @@ -1288,9 +1291,11 @@ int snd_pcm_hw_open(snd_pcm_t **pcmp, const char *name, assert(pcmp); - if ((ret = snd_ctl_hw_open(&ctl, NULL, card, 0)) < 0) + if ((ret = snd_ctl_hw_open(&ctl, NULL, card, control_device_node, 0)) < 0) return ret; + card = snd_ctl_hw_card(ctl); + switch (stream) { case SND_PCM_STREAM_PLAYBACK: filefmt = SNDRV_FILE_PCM_STREAM_PLAYBACK; @@ -1399,7 +1404,7 @@ int _snd_pcm_hw_open(snd_pcm_t **pcmp, const char *name, { snd_config_iterator_t i, next; long card = -1, device = 0, subdevice = -1; - const char *str; + const char *str, *devnode = NULL; int err, sync_ptr_ioctl = 0; int rate = 0, channels = 0; snd_pcm_format_t format = SND_PCM_FORMAT_UNKNOWN; @@ -1428,10 +1433,14 @@ int _snd_pcm_hw_open(snd_pcm_t **pcmp, const char *name, SNDERR("Invalid type for %s", id); return -EINVAL; } - card = snd_card_get_index(str); - if (card < 0) { - SNDERR("Invalid value for %s", id); - return card; + if (str[0] == '/') + devnode = str; + else { + card = snd_card_get_index(str); + if (card < 0) { + SNDERR("Invalid value for %s", id); + return card; + } } } continue; @@ -1498,11 +1507,11 @@ int _snd_pcm_hw_open(snd_pcm_t **pcmp, const char *name, SNDERR("Unknown field %s", id); return -EINVAL; } - if (card < 0) { + if (card < 0 && !devnode) { SNDERR("card is not defined"); return -EINVAL; } - err = snd_pcm_hw_open(pcmp, name, card, device, subdevice, stream, + err = snd_pcm_hw_open(pcmp, name, card, devnode, device, subdevice, stream, mode | (nonblock ? SND_PCM_NONBLOCK : 0), 0, sync_ptr_ioctl); if (err < 0) diff --git a/src/rawmidi/rawmidi_hw.c b/src/rawmidi/rawmidi_hw.c index 87829fd..f02d4cf 100644 --- a/src/rawmidi/rawmidi_hw.c +++ b/src/rawmidi/rawmidi_hw.c @@ -1,4 +1,5 @@ -/* +/* -*- Mode: C; tab-width: 8; indent-tabs-mode: t; c-basic-offset: 8 -*- + * * RawMIDI - Hardware * Copyright (c) 2000 by Jaroslav Kysela <perex(a)perex.cz> * Abramo Bagnara <abramo(a)alsa-project.org> @@ -170,7 +171,9 @@ static const snd_rawmidi_ops_t snd_rawmidi_hw_ops = { int snd_rawmidi_hw_open(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp, - const char *name, int card, int device, int subdevice, + const char *name, + int card, const char *control_device_node, + int device, int subdevice, int mode) { int fd, ver, ret; @@ -186,9 +189,12 @@ int snd_rawmidi_hw_open(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp, *inputp = NULL; if (outputp) *outputp = NULL; - - if ((ret = snd_ctl_hw_open(&ctl, NULL, card, 0)) < 0) + + if ((ret = snd_ctl_hw_open(&ctl, NULL, card, control_device_node, 0)) < 0) return ret; + + card = snd_ctl_hw_card(ctl); + sprintf(filename, SNDRV_FILE_RAWMIDI, card, device); __again: @@ -330,7 +336,7 @@ int _snd_rawmidi_hw_open(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp, { snd_config_iterator_t i, next; long card = -1, device = 0, subdevice = -1; - const char *str; + const char *str, *devnode = NULL; int err; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -345,9 +351,13 @@ int _snd_rawmidi_hw_open(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp, err = snd_config_get_string(n, &str); if (err < 0) return -EINVAL; - card = snd_card_get_index(str); - if (card < 0) - return card; + if (str[0] == '/') + devnode = str; + else { + card = snd_card_get_index(str); + if (card < 0) + return card; + } } continue; } @@ -365,8 +375,8 @@ int _snd_rawmidi_hw_open(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp, } return -EINVAL; } - if (card < 0) + if (card < 0 && !devnode) return -EINVAL; - return snd_rawmidi_hw_open(inputp, outputp, name, card, device, subdevice, mode); + return snd_rawmidi_hw_open(inputp, outputp, name, card, devnode, device, subdevice, mode); } SND_DLSYM_BUILD_VERSION(_snd_rawmidi_hw_open, SND_RAWMIDI_DLSYM_VERSION); diff --git a/src/rawmidi/rawmidi_local.h b/src/rawmidi/rawmidi_local.h index 3388502..ed2b7da 100644 --- a/src/rawmidi/rawmidi_local.h +++ b/src/rawmidi/rawmidi_local.h @@ -1,4 +1,4 @@ -/* +/* -*- Mode: C; tab-width: 8; indent-tabs-mode: t; c-basic-offset: 8 -*- * Rawmidi interface - local header file * Copyright (c) 2000 by Abramo Bagnara <abramo(a)alsa-project.org> * @@ -51,7 +51,9 @@ struct _snd_rawmidi { }; int snd_rawmidi_hw_open(snd_rawmidi_t **input, snd_rawmidi_t **output, - const char *name, int card, int device, int subdevice, + const char *name, + int card, const char *control_device_node, + int device, int subdevice, int mode); int snd_rawmidi_virtual_open(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp, -- 1.6.2.2 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

2 9

[alsa-devel] [RESEND][PATCH 1/4] pxa2xx-i2s: Proper initialization
by Karl Beldan 12 May '09

12 May '09

Reset FIFO logic and registers, and make sure REC and RPL functions along with FIFO service are disabled at probe. Signed-off-by: Karl Beldan <karl.beldan(a)mobile-devices.fr> --- sound/soc/pxa/pxa2xx-i2s.c | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 2f4b6e4..450fca6 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -346,6 +346,19 @@ static int pxa2xx_i2s_probe(struct platform_device *dev) if (ret != 0) clk_put(clk_i2s); + /* + * PXA Developer's Manual: + * If SACR0[ENB] is toggled in the middle of a normal operation, + * the SACR0[RST] bit must also be set and cleared to reset all + * I2S controller registers. + */ + SACR0 = SACR0_RST; + SACR0 = 0; + /* Make sure RPL and REC are disabled */ + SACR1 = SACR1_DRPL | SACR1_DREC; + /* Along with FIFO servicing */ + SAIMR &= ~(SAIMR_RFS | SAIMR_TFS); + return ret; } -- 1.6.3.rc1.34.g0be9b -- Karl Beldan

2 1

[alsa-devel] Master vs. Front/Rear/LFE/... elements
by Lennart Poettering 12 May '09

12 May '09

Heya! Some cards expose 'Master' volume sliders. Others expose seperate (stereo) sliders for 'Front', 'Rear' and so on. I have trouble dealing with them properly in PulseAudio: First of all, on some cards 'Master' seems not to have any effect on the actual analog output, only 'Front' and friends do. Is this a bug or intended behaviour? Can I assume that 'Master' and 'Front' are always independant? Secondly, I have trouble supporting the 'Front'/'Rear'/'Side'/... elements properly, since they split up the surround channels into seperate elements. Now, this is confusing in many ways, even for "amixer" which will then show channels such as "Rear Front Left" and so on, which obviously make no sense. snd_mixer_selem_has_playback_channel() just returns bogus data for these cases. Why are those elements seperate anyway? Why aren't they combined into a single multi-channel event? Looking at the APIs I get the idea that the problem appears to be that elements can only control all channels the same are all independantly which doesn't really match 1:1 on my multichannel sound cards. However, wouldn't it be possible to use the 'index' value of a selem_id for this? I.e. have a series of controls by the same name but different indexes which would then implement snd_mixer_selem_has_playback_channel() correctly? i.e. foo,0 would do front-left/right, foo,1 would do rear, foo,2 would do lfe, and so? I have no clue how this implemented internally, so not sure how feasible this might be. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4

7 16

[alsa-devel] 2.6.29 Vs 2.6.30
by Harsha, Priya 12 May '09

12 May '09

Hi, I was developing a driver on 2.6.29 and it was playing using aplay properly but when I moved to 2.6.30, same code base is giving me underrun messages. What is the changes with respect to this in 2.6.30? Primarily, why do underrun messages come and how does driver need to handle it? Any clue will be of great help. Thanks, Harsha

1 0

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 12 May '09

12 May '09

The following changes since commit bec4c99e8637b5b8bd4b0513eacb51da25885e3b: Mark Brown (1): ASoC: Fix file patterns for PXA sound drivers are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Joonyoung Shim (1): ASoC: TWL4030: change DAPM for analog microphone selection Mark Brown (3): ASoC: Fix up CODEC DAI formats for big endian CPUs ASoC: Enforce symmetric rates for PXA2xx I2S Merge branch 'for-2.6.30' into for-2.6.31 Mike Rapoport (3): ASoC: soc-core: fix crash when removing not instantiated card pxa2xx-ac97: fix reset gpio mode setting ASoC: em-x270: make the driver support also eXeda and CM-X300 machines Roel Kluin (1): ASoC: remove driver_data direct access of struct device Sergey Lapin (1): ASoC: AFEB9260 driver sound/arm/pxa2xx-ac97-lib.c | 2 +- sound/soc/atmel/Kconfig | 8 ++ sound/soc/atmel/Makefile | 1 + sound/soc/atmel/snd-soc-afeb9260.c | 203 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl4030.c | 52 ++++------ sound/soc/codecs/wm8400.c | 4 +- sound/soc/codecs/wm8731.c | 4 +- sound/soc/codecs/wm8753.c | 4 +- sound/soc/pxa/Kconfig | 4 +- sound/soc/pxa/em-x270.c | 9 +- sound/soc/pxa/pxa2xx-i2s.c | 1 + sound/soc/soc-core.c | 43 ++++++++ 12 files changed, 289 insertions(+), 46 deletions(-) create mode 100644 sound/soc/atmel/snd-soc-afeb9260.c

1 0

[alsa-devel] ALSA for SoC audio support full duplex problem
by jinjin xu 12 May '09

12 May '09

Dear All. I use 2.6.21.5-cfs-v19 on S3C6410(arm),use wm8987 sound card(i2s interface) It is not use in full duplex. I grep and diff the drivers ens1370.c: snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ensoniq_playback1_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ensoniq_capture_ops); soc-core.c: if (playback) { snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); } if (capture) { snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); } ens1370.c use two different fops for playback and record(I test it in Vmware,It can run on full duplex) soc-core.c use the same fops for playback and record It that the problem ? So how to fix it for support full duplex? Thanks!

2 1

Re: [alsa-devel] [PATCH 1/1] hdsp: allow firmware loading from inside the kernel
by Takashi Iwai 12 May '09

12 May '09

At Tue, 12 May 2009 09:41:36 +0200, Raphaël Doursenaud wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Takashi Iwai a écrit : > > At Tue, 12 May 2009 09:25:27 +0200, > > Raphaël Doursenaud wrote: > >> -----BEGIN PGP SIGNED MESSAGE----- > >> Hash: SHA1 > >> > >> Takashi Iwai a écrit : > >>> At Tue, 12 May 2009 09:05:19 +0200, > >>> Raphaël Doursenaud wrote: > >>>> -----BEGIN PGP SIGNED MESSAGE----- > >>>> Hash: SHA1 > >>>> > >>>> Takashi Iwai a écrit : > >>>>> At Tue, 12 May 2009 08:47:29 +0200, > >>>>> Raphaël Doursenaud wrote: > >>>>>> -----BEGIN PGP SIGNED MESSAGE----- > >>>>>> Hash: SHA1 > >>>>>> > >>>>>> Takashi Iwai a écrit : > >>>>>>> At Tue, 12 May 2009 08:16:08 +0200, > >>>>>>> Raphaël Doursenaud wrote: > >>>>>>>> From: Raphaël Doursenaud <rdoursenaud(a)free.fr> > >>>>>>>> > >>>>>>>> Allow the use of the FIRMWARE_IN_KERNEL option with hdsp cards and > >>>>>>>> in-kernel driver. > >>>>>>> Did it really work without problems? > >>>>>>> > >>>>>>> > >>>>>>> Takashi > >>>>>> Tested over the weekend with two multifaces in my DAW. > >>>>>> Got no problem. > >>>>> Interesting. > >>>>> Did you build the firmware file into the kernel, or not? > >>>>> > >>>>> > >>>>> Takashi > >>>> Yes I built all hdsp fimware files (multiface_firmware.bin > >>>> multiface_firmware_rev11.bin digiface_firmware.bin > >>>> digiface_firmware_rev11.bin) into the kernel. > >>>> It's the aim of this patch. > >>> Well, the problem I'm concerned is that the driver can be compiled > >>> in even if you have no built-in firmware. And there is no restriction > >>> or dependency check in Kconfig, so far. > >>> > >>> Could you test how the kernel behaves without the built-in firmware? > >>> Does it hang or give any critical error? > >>> > >>> > >>> thanks, > >>> > >>> Takashi > >> Could you be more specific? > >> I'm not sure to understand why it could be a problem. > >> Do you think that if I set FIRMWARE_IN_KERNEL without compiling the > >> firmware(s) in-kernel the request_firmware() will not resolve and cause > >> an error? > > > > Yes, exactly. > > request_firmware() shall fail in that case likely after a long > > time-out (unless you have the firmware files in initrd) because there > > is really no file / data available at the time it's called. > > And I'm not sure whether this could lead to a fatal operation error. > > > > > > Takashi > AFAIK this is handled in the code (from line 5080) and should lead to > "Hammerfall-DSP: couldn't get firmware from userspace. try using hdsploader" Yep, a fallback mechanism is found in hdsp driver itself. But the driver probe hangs for too long time, and I'm just worried about its influence on the whole kernel boot-up. > I'm building a kernel to test that case. Thanks. Takashi

2 3

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 12 May '09

12 May '09

The following changes since commit 97a775c49c7e1b47b016a492463486a5b86da479: Jinyoung Park (1): ASoC: Fix errors in WM8990 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Mike Rapoport (2): ASoC: soc-core: fix crash when removing not instantiated card pxa2xx-ac97: fix reset gpio mode setting sound/arm/pxa2xx-ac97-lib.c | 2 +- sound/soc/soc-core.c | 3 +++ 2 files changed, 4 insertions(+), 1 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: TWL4030: change DAPMs for analog microphone selection
by Joonyoung Shim 12 May '09

12 May '09

The mixer DAPM is suitable than the mux DAPM for the analog microphone registers(0x05, 0x06) because each bit for path is switchable. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 52 ++++++++++++++++--------------------------- 1 files changed, 19 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e23c20c..a7875c2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -397,36 +397,18 @@ static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control = SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum); /* Left analog microphone selection */ -static const char *twl4030_analoglmic_texts[] = - {"Off", "Main mic", "Headset mic", "AUXL", "Carkit mic"}; - -static const unsigned int twl4030_analoglmic_values[] = - {0x0, 0x1, 0x2, 0x4, 0x8}; - -static const struct soc_enum twl4030_analoglmic_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICL, 0, 0xf, - ARRAY_SIZE(twl4030_analoglmic_texts), - twl4030_analoglmic_texts, - twl4030_analoglmic_values); - -static const struct snd_kcontrol_new twl4030_dapm_analoglmic_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_analoglmic_enum); +static const struct snd_kcontrol_new twl4030_dapm_analoglmic_controls[] = { + SOC_DAPM_SINGLE("Main mic", TWL4030_REG_ANAMICL, 0, 1, 0), + SOC_DAPM_SINGLE("Headset mic", TWL4030_REG_ANAMICL, 1, 1, 0), + SOC_DAPM_SINGLE("AUXL", TWL4030_REG_ANAMICL, 2, 1, 0), + SOC_DAPM_SINGLE("Carkit mic", TWL4030_REG_ANAMICL, 3, 1, 0), +}; /* Right analog microphone selection */ -static const char *twl4030_analogrmic_texts[] = - {"Off", "Sub mic", "AUXR"}; - -static const unsigned int twl4030_analogrmic_values[] = - {0x0, 0x1, 0x4}; - -static const struct soc_enum twl4030_analogrmic_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICR, 0, 0x5, - ARRAY_SIZE(twl4030_analogrmic_texts), - twl4030_analogrmic_texts, - twl4030_analogrmic_values); - -static const struct snd_kcontrol_new twl4030_dapm_analogrmic_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_analogrmic_enum); +static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = { + SOC_DAPM_SINGLE("Sub mic", TWL4030_REG_ANAMICR, 0, 1, 0), + SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 1, 1, 0), +}; /* TX1 L/R Analog/Digital microphone selection */ static const char *twl4030_micpathtx1_texts[] = @@ -1034,11 +1016,15 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD| SND_SOC_DAPM_POST_REG), - /* Analog input muxes with switch for the capture amplifiers */ - SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route", - TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_control), - SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route", - TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_control), + /* Analog input mixers for the capture amplifiers */ + SND_SOC_DAPM_MIXER("Analog Left Capture Route", + TWL4030_REG_ANAMICL, 4, 0, + &twl4030_dapm_analoglmic_controls[0], + ARRAY_SIZE(twl4030_dapm_analoglmic_controls)), + SND_SOC_DAPM_MIXER("Analog Right Capture Route", + TWL4030_REG_ANAMICR, 4, 0, + &twl4030_dapm_analogrmic_controls[0], + ARRAY_SIZE(twl4030_dapm_analogrmic_controls)), SND_SOC_DAPM_PGA("ADC Physical Left", TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0), -- 1.6.0.4

3 9

Re: [alsa-devel] [PATCH 1/1] hdsp: allow firmware loading from inside the kernel
by Takashi Iwai 12 May '09

12 May '09

At Tue, 12 May 2009 09:05:19 +0200, Raphaël Doursenaud wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Takashi Iwai a écrit : > > At Tue, 12 May 2009 08:47:29 +0200, > > Raphaël Doursenaud wrote: > >> -----BEGIN PGP SIGNED MESSAGE----- > >> Hash: SHA1 > >> > >> Takashi Iwai a écrit : > >>> At Tue, 12 May 2009 08:16:08 +0200, > >>> Raphaël Doursenaud wrote: > >>>> From: Raphaël Doursenaud <rdoursenaud(a)free.fr> > >>>> > >>>> Allow the use of the FIRMWARE_IN_KERNEL option with hdsp cards and > >>>> in-kernel driver. > >>> Did it really work without problems? > >>> > >>> > >>> Takashi > >> Tested over the weekend with two multifaces in my DAW. > >> Got no problem. > > > > Interesting. > > Did you build the firmware file into the kernel, or not? > > > > > > Takashi > Yes I built all hdsp fimware files (multiface_firmware.bin > multiface_firmware_rev11.bin digiface_firmware.bin > digiface_firmware_rev11.bin) into the kernel. > It's the aim of this patch. Well, the problem I'm concerned is that the driver can be compiled in even if you have no built-in firmware. And there is no restriction or dependency check in Kconfig, so far. Could you test how the kernel behaves without the built-in firmware? Does it hang or give any critical error? thanks, Takashi

2 3

[alsa-devel] Audio blank samples after some seconds of playing
by javier Martin 12 May '09

12 May '09

I have a .wav file which lasts about 2 minutes. I am playing it with aplay. It works fine, but after some seconds of playing, I hear two blanks suddenly. After that, it continues playing fine till the end of the file. I have tried to increment buffer size and period size but I don't see any difference. Possition of blank samples seems to be quite random, but it always happens near the beggining of the file. Can you please give me some tip for debugging this problem or an idea of what more can I try? Thank you.

1 0

[alsa-devel] [PATCH 1/1] hdsp: allow firmware loading from inside the kernel
by Raphaël Doursenaud 12 May '09

12 May '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 From: Raphaël Doursenaud <rdoursenaud(a)free.fr> Allow the use of the FIRMWARE_IN_KERNEL option with hdsp cards and in-kernel driver. Also corrected a typo in the comment. Signed-off-by: Raphael Doursenaud <rdoursenaud(a)free.fr> - --- sound/pci/rme9652/hdsp.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 314e735..d0d4629 100644 - --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -402,9 +402,9 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_DMA_AREA_BYTES ((HDSP_MAX_CHANNELS+1) * HDSP_CHANNEL_BUFFER_BYTES) #define HDSP_DMA_AREA_KILOBYTES (HDSP_DMA_AREA_BYTES/1024) - -/* use hotplug firmeare loader? */ +/* use hotplug firmware loader? */ #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) - -#if !defined(HDSP_USE_HWDEP_LOADER) && !defined(CONFIG_SND_HDSP) +#if !defined(HDSP_USE_HWDEP_LOADER) #define HDSP_FW_LOADER #endif #endif - -- Raphaël Doursenaud -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkoJFCQACgkQaZKmNAdXaVVliACaA5BMknMAXgOEN/7YQmWGPwUh uRgAn0BiIwAtGU/AOi+QRn81k8apvHfi =+4Kw -----END PGP SIGNATURE-----

3 6

[alsa-devel] [PATCH] soc: remove driver_data direct access of struct device
by Roel Kluin 12 May '09

12 May '09

diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 510efa6..e4547de 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1473,8 +1473,8 @@ static int wm8400_codec_probe(struct platform_device *dev) codec = &priv->codec; codec->private_data = priv; - codec->control_data = dev->dev.driver_data; - priv->wm8400 = dev->dev.driver_data; + codec->control_data = dev_get_drvdata(&dev->dev); + priv->wm8400 = dev_get_drvdata(&dev->dev); ret = regulator_bulk_get(priv->wm8400->dev, ARRAY_SIZE(power), &power[0]); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index e043e3f..7a20587 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -666,14 +666,14 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi) codec->hw_write = (hw_write_t)wm8731_spi_write; codec->dev = &spi->dev; - spi->dev.driver_data = wm8731; + dev_set_drvdata(&spi->dev, wm8731); return wm8731_register(wm8731); } static int __devexit wm8731_spi_remove(struct spi_device *spi) { - struct wm8731_priv *wm8731 = spi->dev.driver_data; + struct wm8731_priv *wm8731 = dev_get_drvdata(&spi->dev); wm8731_unregister(wm8731); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a6e8f3f..d121e58 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1822,14 +1822,14 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) codec->hw_write = (hw_write_t)wm8753_spi_write; codec->dev = &spi->dev; - spi->dev.driver_data = wm8753; + dev_set_drvdata(&spi->dev, wm8753); return wm8753_register(wm8753); } static int __devexit wm8753_spi_remove(struct spi_device *spi) { - struct wm8753_priv *wm8753 = spi->dev.driver_data; + struct wm8753_priv *wm8753 = dev_get_drvdata(&spi->dev); wm8753_unregister(wm8753); return 0; }

2 1

[alsa-devel] MacbookPro5,1 & hda-analyzer
by Ozan Çağlayan 11 May '09

11 May '09

Hi, I just bought a MacbookPro5,1 and having difficulties with the onboard HDA chipset. * No sound can be heard of it if model= is not set to mbp3: [ 12.009422] hda_codec: Unknown model for ALC883, trying auto-probe from BIOS... http://www.alsa-project.org/db/?f=4b2f257e72523bccafd3ace1d3c7044fb5f09ff8 * If model=mbp3 is given, it's possible to get sound from it but with many annoying problems: - You have to switch to 6ch mode for getting sound, doesn't work with 2ch mode, - The sound quality is very very bad, - Once you switch to 6ch mode, - You have to increment the volume of *Line-in* to have hearable sound from *internal speakers*, - You have to increment the volume of *Front* to have hearable sound from *headphones*, - The two controls are not mutually exclusive, you have to mute one of them, if you're using the other. And then I decided to play with hda-analyzer. After a bit of playing, I started to get sound from the board without the model=mbp3. But the mixers are still behaving somewhat strange. Before actually reporting any feedback on what I've switched and did, what do you suggest me to do? Is setting the model= to something, something similar to what I was doing with hda-analyzer? Thanks Ozan Caglayan

1 2

[alsa-devel] [PATCH] ASoC: em-x270: make the driver support also eXeda and CM-X300 machines
by Mike Rapoport 11 May '09

11 May '09

Signed-off-by: Mike Rapoport <mike(a)compulab.co.il> --- sound/soc/pxa/Kconfig | 4 ++-- sound/soc/pxa/em-x270.c | 9 +++++---- 2 files changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index ad8a10f..f8cc974 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -89,13 +89,13 @@ config SND_PXA2XX_SOC_E800 Toshiba e800 PDA config SND_PXA2XX_SOC_EM_X270 - tristate "SoC Audio support for CompuLab EM-x270" + tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" depends on SND_PXA2XX_SOC && MACH_EM_X270 select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help Say Y if you want to add support for SoC audio on - CompuLab EM-x270. + CompuLab EM-x270, eXeda and CM-X300 machines. config SND_PXA2XX_SOC_PALM27X bool "SoC Audio support for Palm T|X, T5 and LifeDrive" diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index 949be9c..f4756e4 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -1,7 +1,7 @@ /* - * em-x270.c -- SoC audio for EM-X270 + * SoC audio driver for EM-X270, eXeda and CM-X300 * - * Copyright 2007 CompuLab, Ltd. + * Copyright 2007, 2009 CompuLab, Ltd. * * Author: Mike Rapoport <mike(a)compulab.co.il> * @@ -68,7 +68,8 @@ static int __init em_x270_init(void) { int ret; - if (!machine_is_em_x270()) + if (!(machine_is_em_x270() || machine_is_exeda() + || machine_is_cm_x300())) return -ENODEV; em_x270_snd_device = platform_device_alloc("soc-audio", -1); @@ -95,5 +96,5 @@ module_exit(em_x270_exit); /* Module information */ MODULE_AUTHOR("Mike Rapoport"); -MODULE_DESCRIPTION("ALSA SoC EM-X270"); +MODULE_DESCRIPTION("ALSA SoC EM-X270, eXeda and CM-X300"); MODULE_LICENSE("GPL"); -- 1.5.6.4

2 1

[alsa-devel] [PATCH] ASoC: soc-core: fix crash when removing not instantiated card
by Mike Rapoport 11 May '09

11 May '09

If the card was not instantiated in snd_soc_instantiate_card, calling soc-remove will crash because some of codec, cpu_dai and card .remove methods are called twice. Fix this by returning from soc_remove immediately. Signed-off-by: Mike Rapoport <mike(a)compulab.co.il> --- sound/soc/soc-core.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 99712f6..1cd149b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -954,6 +954,9 @@ static int soc_remove(struct platform_device *pdev) struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; + if (!card->instantiated) + return 0; + run_delayed_work(&card->delayed_work); if (platform->remove) -- 1.5.6.4

2 1

[alsa-devel] Problem with sound on a Dell Dimension 4550 when it comes to using the ALSA-driver.
by Kristoffer Grundström 11 May '09

11 May '09

I was wondering if there will be a fix out for my problem that's described here: https://qa.mandriva.com/show_bug.cgi?id=49048 I'm getting sick and tired about having this problem in all the Mandriva-versions. Can you help me in any way? Do I have to add more info to that bugreport? /Kristoffer Grundstroem Unofficial Swedish translator of Mandriva.

2 1

[alsa-devel] [PATCH] pxa2xx-ac97: fix reset gpio mode setting
by Mike Rapoport 11 May '09

11 May '09

Signed-off-by: Mike Rapoport <mike(a)compulab.co.il> --- sound/arm/pxa2xx-ac97-lib.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index a2c12d1..6fdca97 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -65,7 +65,7 @@ static void set_resetgpio_mode(int resetgpio_action) switch (resetgpio_action) { case RESETGPIO_NORMAL_ALTFUNC: if (reset_gpio == 113) - mode = 113 | GPIO_OUT | GPIO_DFLT_LOW; + mode = 113 | GPIO_ALT_FN_2_OUT; if (reset_gpio == 95) mode = 95 | GPIO_ALT_FN_1_OUT; break; -- 1.5.6.4

5 6

[alsa-devel] [PATCH 0/1] ASoC: TWL4030: Possible fix for the Beep at the end of the audio file.
by Peter Ujfalusi 11 May '09

11 May '09

Hello, I have finally managed to reproduce and test the 'tuck' phenomena. The following patch seamingly fixes the issue at least in my hardware setup. Note that I still can not test this on Beagle and on the HW that I have testing the headset output is kind of tricky, so this might very well not fix the issue. Anuj, Jarkko, Arun: Would you be so kind and test it on your boards? --- Peter Ujfalusi (1): ASoC: TWL4030: Headset VMID ramp fix sound/soc/codecs/twl4030.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-)

4 11

[alsa-devel] spdif/iec958 outputs at VT1708S not recognized
by Peter Niemayer 11 May '09

11 May '09

Hi, I recently installed an Asus M4N78 mainboard, that uses a nVidia GeForce 8300 chipset and a VT1708S soundchip. Alas, the code in pci/hda/patch_via.c did not recognize the 3 existing iec958/spdif outputs, and presented only the analog devices. I had to short circuit the pin definition in vt1708S_parse_auto_config() to get the digital outputs to work: spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + // test fails for Asus M4N78 if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = VT1708S_DIGOUT_NID; spec->extra_dig_out_nid = 0x15; The digital sound output works just fine for me with this change, but I guess there should be a cleaner way to make this work? Several people in forums on the InterNet reported similar problems with getting their digital outputs on VT1708S to work under Linux. Regards, Peter Niemayer

4 9

[alsa-devel] speaker-test cleanups and fixes
by Dan McGee 11 May '09

11 May '09

This is a series of 3 (mostly unrelated) patches to clean up and fix up a few shortcomings I found in the speaker-test program. Please let me know of any concerns or comments on the patches. -Dan

2 4

[alsa-devel] How to test new SDPIF implementation?
by Jon Smirl 10 May '09

10 May '09

I would like to test my SPDIF driver implementation. How can I used command line tools to play sample 5.1 music over SPDIF? I have a file 5.1 AC3 encoded file -- w.ac3 This doesn't work: iecset -D hw:0 audio off aplay -c 2 -r 48000 -f S16_BE -D hw:0,1 w.ac3 I can play uncompressed music on SDPIF without problem. I know my audio system will play the compressed music, I can use A52 from my desktop. -- Jon Smirl jonsmirl(a)gmail.com

2 1

[alsa-devel] [PATCH 1/2] speaker-test: move existing endian macros up in the file
by Dan McGee 10 May '09

10 May '09

This is necessary for a later patch removing the various endianness checks sprinkled throughout the code. Signed-off-by: Dan McGee <dpmcgee(a)gmail.com> --- speaker-test/speaker-test.c | 20 ++++++++++---------- 1 files changed, 10 insertions(+), 10 deletions(-) diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index d983a4f..c1d53b6 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -68,6 +68,16 @@ enum { #define MAX_CHANNELS 16 +#if __BYTE_ORDER == __LITTLE_ENDIAN +#define COMPOSE_ID(a,b,c,d) ((a) | ((b)<<8) | ((c)<<16) | ((d)<<24)) +#define LE_SHORT(v) (v) +#define LE_INT(v) (v) +#else +#define COMPOSE_ID(a,b,c,d) ((d) | ((c)<<8) | ((b)<<16) | ((a)<<24)) +#define LE_SHORT(v) bswap_16(v) +#define LE_INT(v) bswap_32(v) +#endif + static char *device = "default"; /* playback device */ static snd_pcm_format_t format = SND_PCM_FORMAT_S16; /* sample format */ static unsigned int rate = 48000; /* stream rate */ @@ -520,16 +530,6 @@ struct wave_header { } chunk; }; -#if __BYTE_ORDER == __LITTLE_ENDIAN -#define COMPOSE_ID(a,b,c,d) ((a) | ((b)<<8) | ((c)<<16) | ((d)<<24)) -#define LE_SHORT(v) (v) -#define LE_INT(v) (v) -#else -#define COMPOSE_ID(a,b,c,d) ((d) | ((c)<<8) | ((b)<<16) | ((a)<<24)) -#define LE_SHORT(v) bswap_16(v) -#define LE_INT(v) bswap_32(v) -#endif - #define WAV_RIFF COMPOSE_ID('R','I','F','F') #define WAV_WAVE COMPOSE_ID('W','A','V','E') #define WAV_FMT COMPOSE_ID('f','m','t',' ') -- 1.6.2.4

1 1

Re: [alsa-devel] ASoC: pxa2xx-i2s
by Mark Brown 09 May '09

09 May '09

On Wed, Jan 21, 2009 at 10:46:17AM -0600, Brian G Rhodes wrote: > Mark Brown wrote: > >What is the clock master in your system? > The master is the i2s controller sysclk. The dac2x is in slave mode. > Though the i2s controller isn't directly supplying the bit clock to the > codec in this case. We're using the PLL on the dac2x. Just to confirm, the codec is the clock master on the I2S bus itself?

3 6

[alsa-devel] [PATCH 3/4] pxa2xx-i2s: Cleaner use of the FIFOs
by Karl Beldan 09 May '09

09 May '09

The FIFO logic and the registers are reset at stream startup with SACR0_RST => the sibling function might suffer and the FIFOs might unpleasantly fill up whence the calls to pxa_i2s_wait all over the place. This gets rid of it. Signed-off-by: Karl Beldan <karl.beldan(a)mobile-devices.fr> --- sound/soc/pxa/pxa2xx-i2s.c | 20 +------------------- 1 files changed, 1 insertions(+), 19 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 0e53d07..755e2d1 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -106,25 +106,12 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, if (IS_ERR(clk_i2s)) return PTR_ERR(clk_i2s); - if (!cpu_dai->active) { - SACR0 |= SACR0_RST; + if (!cpu_dai->active) SACR0 = 0; - } return 0; } -/* wait for I2S controller to be ready */ -static int pxa_i2s_wait(void) -{ - int i; - - /* flush the Rx FIFO */ - for(i = 0; i < 16; i++) - SADR; - return 0; -} - static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -169,7 +156,6 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, BUG_ON(IS_ERR(clk_i2s)); clk_enable(clk_i2s); - pxa_i2s_wait(); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out; @@ -256,7 +242,6 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) { SACR0 &= ~SACR0_ENB; - pxa_i2s_wait(); clk_disable(clk_i2s); } } @@ -275,7 +260,6 @@ static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai) /* deactivate link */ SACR0 &= ~SACR0_ENB; - pxa_i2s_wait(); return 0; } @@ -284,8 +268,6 @@ static int pxa2xx_i2s_resume(struct snd_soc_dai *dai) if (!dai->active) return 0; - pxa_i2s_wait(); - SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB; SACR1 = pxa_i2s.sacr1; SAIMR = pxa_i2s.saimr; -- 1.6.3.rc1.34.g0be9b -- Karl

2 3

[alsa-devel] [PATCH 1/4] pxa2xx-i2s: Proper initialization
by Karl Beldan 08 May '09

08 May '09

Reset FIFO logic and registers, and make sure REC and RPL functions and FIFO service are disabled. Signed-off-by: Karl Beldan <karl.beldan(a)mobile-devices.fr> --- sound/soc/pxa/pxa2xx-i2s.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 2f4b6e4..52862dc 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -370,6 +370,18 @@ static struct platform_driver pxa2xx_i2s_driver = { static int __init pxa2xx_i2s_init(void) { clk_i2s = ERR_PTR(-ENOENT); + + /* + * PXA Developer's Manual: + * If SACR0[ENB] is toggled in the middle of a normal operation, + * the SACR0[RST] bit must also be set and cleared to reset all + * I2S controller registers. + */ + SACR0 = SACR0_RST; + SACR0 = 0; + SACR1 = SACR1_DRPL | SACR1_DREC; + SAIMR &= ~(SAIMR_RFS | SAIMR_TFS); + return platform_driver_register(&pxa2xx_i2s_driver); } -- 1.6.3.rc1.34.g0be9b -- Karl

2 2

[alsa-devel] [PATCH] AFEB9260 ASoC driver
by Sergey Lapin 08 May '09

08 May '09

ASoC driver for AT91SAM9260-based AFEB9260 board Signed-off-by: Sergey Lapin <slapin(a)ossfans.org> --- arch/arm/mach-at91/board-afeb-9260v1.c | 4 + sound/soc/atmel/Kconfig | 8 ++ sound/soc/atmel/Makefile | 1 + sound/soc/atmel/afeb9260.c | 217 ++++++++++++++++++++++++++++++++ 4 files changed, 230 insertions(+), 0 deletions(-) create mode 100644 sound/soc/atmel/afeb9260.c diff --git a/arch/arm/mach-at91/board-afeb-9260v1.c b/arch/arm/mach-at91/board-afeb-9260v1.c index e263fda..e3f1c56 100644 --- a/arch/arm/mach-at91/board-afeb-9260v1.c +++ b/arch/arm/mach-at91/board-afeb-9260v1.c @@ -164,6 +164,8 @@ static struct at91_mmc_data __initdata afeb9260_mmc_data = { static struct i2c_board_info __initdata afeb9260_i2c_devices[] = { { + I2C_BOARD_INFO("tlv320aic23", 0x1a), + }, { I2C_BOARD_INFO("fm3130", 0x68), }, { I2C_BOARD_INFO("24c64", 0x50), @@ -196,6 +198,8 @@ static void __init afeb9260_board_init(void) /* I2C */ at91_add_device_i2c(afeb9260_i2c_devices, ARRAY_SIZE(afeb9260_i2c_devices)); + /* Audio */ + at91_add_device_ssc(AT91SAM9260_ID_SSC, ATMEL_SSC_TX); } MACHINE_START(AFEB9260, "Custom afeb9260 board") diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index a608d70..e720d5e 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -41,3 +41,11 @@ config SND_AT32_SOC_PLAYPAQ_SLAVE and FRAME signals on the PlayPaq. Unless you want to play with the AT32 as the SSC master, you probably want to say N here, as this will give you better sound quality. + +config SND_AT91_SOC_AFEB9260 + tristate "SoC Audio support for AFEB9260 board" + depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_SOC_TLV320AIC23 + help + Say Y here to support sound on AFEB9260 board. diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index f54a7cc..ac2df98 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -13,3 +13,4 @@ snd-soc-playpaq-objs := playpaq_wm8510.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o +obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += afeb9260.o diff --git a/sound/soc/atmel/afeb9260.c b/sound/soc/atmel/afeb9260.c new file mode 100644 index 0000000..1514d8d --- /dev/null +++ b/sound/soc/atmel/afeb9260.c @@ -0,0 +1,217 @@ +/* + * afeb9260.c -- SoC audio for AFEB9260 + * + * Copyright (C) 2008 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 <sound/soc-dapm.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_startup(struct snd_pcm_substream *substream) +{ + return 1; +} + +static void afeb9260_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + + dev_dbg(rtd->socdev->dev, "shutdown"); +} + +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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err; + + /* Set codec DAI configuration */ + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S| + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return err; + } + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return 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 = { + .startup = afeb9260_startup, + .hw_params = afeb9260_hw_params, + .shutdown = afeb9260_shutdown, +}; + +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 audio_map[] = { + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic Jack"}, +}; + +static int afeb9260_tlv320aic23_init(struct snd_soc_codec *codec) +{ + + /* Add osk5912 specific widgets */ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* Set up osk5912 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link afeb9260_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &atmel_ssc_dai[0], + .codec_dai = &tlv320aic23_dai, + .init = afeb9260_tlv320aic23_init, + .ops = &afeb9260_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_machine_afeb9260 = { + .name = "AFEB9260", + .platform = &atmel_soc_platform, + .dai_link = &afeb9260_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device afeb9260_snd_devdata = { + .card = &snd_soc_machine_afeb9260, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *afeb9260_snd_device; + +static int __init afeb9260_soc_init(void) +{ + int err; + struct device *dev; + struct atmel_ssc_info *ssc_p = afeb9260_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + if (!(machine_is_afeb9260())) + return -ENODEV; + + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + printk(KERN_ERR "ASoC: Failed to request SSC 0\n"); + err = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + 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, &afeb9260_snd_devdata); + afeb9260_snd_devdata.dev = &afeb9260_snd_device->dev; + err = platform_device_add(afeb9260_snd_device); + if (err) + goto err1; + + dev = &afeb9260_snd_device->dev; + + return 0; +err1: + platform_device_del(afeb9260_snd_device); + platform_device_put(afeb9260_snd_device); +err_ssc: + 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"); + -- 1.5.6.5

2 5

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 08 May '09

08 May '09

The following changes since commit bbd993077d788589a86a718ba7a7895ba5e71a17: Mark Brown (1): ASoC: Remove redundant codec pointer from DAIs are available in the git repository at: ssh://master.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 (git:// doesn't seem to be updating so...). Daniel Mack (2): ASoC: cs4270: introduce CS4270_I2C_INCR ASoC: cs4270: add power management support Mark Brown (2): ASoC: Add headers to match patterns in MAINTAINERS ASoC: Fix file patterns for PXA sound drivers Peter Ujfalusi (1): ASoC: TWL4030: Fix typo in twl4030_codec_mute function MAINTAINERS | 4 ++- sound/soc/codecs/cs4270.c | 61 ++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/twl4030.c | 2 +- 3 files changed, 63 insertions(+), 4 deletions(-)

2 1

[alsa-devel] [PATCH 1/1] ASoC: TWL4030: Fix typo in twl4030_codec_mute function
by Peter Ujfalusi 08 May '09

08 May '09

Copy-paste error: TWL4030_PRECKL_GAIN >> TWL4030_PRECKR_GAIN It has not caused problems, since TWL4030_PRECKL_GAIN == TWL4030_PRECKR_GAIN == 0x30 Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 1a00e4b..fd392c6 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -237,7 +237,7 @@ static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute) TWL4030_REG_PRECKL_CTL); reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL); twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, - reg_val & (~TWL4030_PRECKL_GAIN), + reg_val & (~TWL4030_PRECKR_GAIN), TWL4030_REG_PRECKR_CTL); /* Disable PLL */ -- 1.6.2.3

2 1

[alsa-devel] [PATCH 1/1] ASoC: Added OMAP3 EVM support in ASoC.
by Anuj Aggarwal 08 May '09

08 May '09

Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/Kconfig | 8 +++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3evm.c | 147 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 157 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3evm.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 675732e..b771238 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -39,6 +39,14 @@ config SND_OMAP_SOC_OMAP2EVM help Say Y if you want to add support for SoC audio on the omap2evm board. +config SND_OMAP_SOC_OMAP3EVM + tristate "SoC Audio support for OMAP3EVM board" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the omap3evm board. + config SND_OMAP_SOC_SDP3430 tristate "SoC Audio support for Texas Instruments SDP3430" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 0c9e4ac..a37f498 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -10,6 +10,7 @@ snd-soc-n810-objs := n810.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o +snd-soc-omap3evm-objs := omap3evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o @@ -18,6 +19,7 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o +obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c new file mode 100644 index 0000000..2af5d93 --- /dev/null +++ b/sound/soc/omap/omap3evm.c @@ -0,0 +1,147 @@ +/* + * omap3evm.c -- ALSA SoC support for OMAP3 EVM + * + * Author: Anuj Aggarwal <anuj.aggarwal(a)ti.com> + * + * Based on sound/soc/omap/beagle.c by Steve Sakoman + * + * Copyright (C) 2008 Texas Instruments, Incorporated + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind, + * whether express or implied; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int omap3evm_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "Can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "Can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "Can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap3evm_ops = { + .hw_params = omap3evm_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3evm_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap3evm_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_omap3evm = { + .name = "omap3evm", + .platform = &omap_soc_platform, + .dai_link = &omap3evm_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3evm_snd_devdata = { + .card = &snd_soc_omap3evm, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3evm_snd_device; + +static int __init omap3evm_soc_init(void) +{ + int ret; + + if (!machine_is_omap3evm()) { + pr_debug("Not OMAP3 EVM!\n"); + return -ENODEV; + } + pr_info("OMAP3 EVM SoC init\n"); + + omap3evm_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap3evm_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap3evm_snd_device, &omap3evm_snd_devdata); + omap3evm_snd_devdata.dev = &omap3evm_snd_device->dev; + *(unsigned int *)omap3evm_dai.cpu_dai->private_data = 1; + + ret = platform_device_add(omap3evm_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap3evm_snd_device); + + return ret; +} + +static void __exit omap3evm_soc_exit(void) +{ + platform_device_unregister(omap3evm_snd_device); +} + +module_init(omap3evm_soc_init); +module_exit(omap3evm_soc_exit); + +MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal(a)ti.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM"); +MODULE_LICENSE("GPLv2"); -- 1.6.2.4

4 3

[alsa-devel] asus xonar DX
by Dmitri Seletski 08 May '09

08 May '09

Hello boys and girls. I have successfully installed asus xonar DX onto PCIE x16 slot on Linux 2.6.28-gentoo-r5 SMP on amd64 architecture. Version of alsa package - alsa-driver1.0.20, alsa firmware 1.0.19. I used alsa package and not kernel modules. Several channels seem to work fine(my speakers are broken, no back channels available, started holy war with creative about it, guess with fiasco with faulty speakers and xfi drivers i am supporter of creative no more, i am not gonna bad mouth them either just yet). So far so good. Microphone works, surround sound works too, I am pretty happy. Thanks for all people who advised me something. If i come across this problem - you will hear from me again ;) Specially

2 1

Re: [alsa-devel] 2.6.29 regression: left audio channel broken after resume from suspend with Intel HDA
by Takashi Iwai 08 May '09

08 May '09

At Tue, 21 Apr 2009 20:39:36 +0200, Tino Keitel wrote: > > On Tue, Apr 21, 2009 at 09:39:13 +0200, Takashi Iwai wrote: > > At Tue, 21 Apr 2009 09:32:19 +0200, > > Tino Keitel wrote: > > > > > > On Tue, Apr 21, 2009 at 07:24:40 +0200, Takashi Iwai wrote: > > > > The commit is fcad94a4c71c36a05f4d5c6dcb174534b4e0b136: > > > > ALSA: hda - Fix the cmd cache keys for amp verbs > > > > > > Thanks for the reply. However, the patch doesn't work. I just got > > > distorted sound again. > > > > Then please run alsa-info.sh (with --no-upload option) before and > > after suspend, and attach the generated files. Also, check the latest > > Linus tree whether the problem still appears. > > My first impression with suspend seems to be wrong. It also sometimes > happens just after a while, event without suspend. > > Attached are 2 alsa-info.sh outputs. One was taken after I noticed the > distorted sound, one after I reloaded snd-hda-intel to fix the > distortion. The mixer seems to be quite different after module reload. The codec communication looks a bit unstable on this device... Does this still happen on the latest 2.6.30 version? Takashi

5 17

[alsa-devel] [PATCH 4/4] pxa2xx-i2s: Fix suspend/resume
by Karl Beldan 08 May '09

08 May '09

pxa2xx_i2s_resume is : - unconditionnaly setting SACR0_ENB - unsetting SACR0_ENB in pxa_i2s.sacr0 fix these. In pxa2xx_i2s_{resume,suspend}, save/restore registers even when !dai->active. Signed-off-by: Karl Beldan <karl.beldan(a)mobile-devices.fr> --- sound/soc/pxa/pxa2xx-i2s.c | 11 +++-------- 1 files changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 755e2d1..5b0f0bf 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -249,9 +249,6 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, #ifdef CONFIG_PM static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai) { - if (!dai->active) - return 0; - /* store registers */ pxa_i2s.sacr0 = SACR0; pxa_i2s.sacr1 = SACR1; @@ -265,14 +262,12 @@ static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai) static int pxa2xx_i2s_resume(struct snd_soc_dai *dai) { - if (!dai->active) - return 0; - - SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB; + SACR0 = pxa_i2s.sacr0 & ~SACR0_ENB; SACR1 = pxa_i2s.sacr1; SAIMR = pxa_i2s.saimr; SADIV = pxa_i2s.sadiv; - SACR0 |= SACR0_ENB; + + SACR0 = pxa_i2s.sacr0; return 0; } -- 1.6.3.rc1.34.g0be9b -- Karl

1 0

[alsa-devel] [PATCH 0/4] pxa2xx-i2s: Rework
by Karl Beldan 08 May '09

08 May '09

Following is a rework of the i2s code for the pxa. I tested it with the ac97 clock as the master clock. Tests feedbacks are welcome. -- Karl

1 0

[alsa-devel] [PATCH 0/1] ASoC: Added OMAP3 EVM Support
by Anuj Aggarwal 07 May '09

07 May '09

With refernce to the discussion at: http://marc.info/?l=linux-omap&m=124169330310075&w=2 I am resending the patch to add OMAP3 EVM support in ASoC. It has been built/checked against the tip. Anuj Aggarwal (1): ASoC: Added OMAP3 EVM support in ASoC. sound/soc/omap/Kconfig | 8 +++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3evm.c | 147 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 157 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3evm.c

1 0

[alsa-devel] Laptop (Samsung Q45) speakers unmute when master channel is unmuted even with headphones connected
by Sam Morris 07 May '09

07 May '09

Since Linux 2.6.28 and through to 2.6.30-rc4, the hda-intel driver has had a bug on my laptop (a Samsung Q45). If I connect my headphones, the laptop's internal speakers are muted--fine. But if I then mute and unmute the master mixer channel, the speakers come back to life, even though the headphones are still connected. Please keep 517241-forwarded(a)bugs.debian.org CC'd in follow-ups. -- Sam Morris https://robots.org.uk/ PGP key id 1024D/5EA01078 3412 EA18 1277 354B 991B C869 B219 7FDB 5EA0 1078

2 3

[alsa-devel] SNDRV_PCM_FORMAT_GSM
by Harsha, Priya 07 May '09

07 May '09

Hi, If I use the format SNDRV_PCM_FORMAT_GSM to identify if the audio stream is voice stream or not, would it be appropriate? Thanks, Harsha

2 2

[alsa-devel] [PATCH 0/2] ASoC updates
by Mark Brown 07 May '09

07 May '09

The following changes since commit 32c8dabc97d436582298ebd0e33af041c69f5a4b: Mark Brown (1): ASoC: Remove obsolete declaration of struct snd_soc_clock_info are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Daniel Mack (1): ASoC: tlv320aic3x: control additions and cleanups Grazvydas Ignotas (1): ASoC: Add support for OMAP3 Pandora sound/soc/codecs/tlv320aic3x.c | 18 ++- sound/soc/omap/Kconfig | 8 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3pandora.c | 311 ++++++++++++++++++++++++++++++++++++++++ 4 files changed, 334 insertions(+), 5 deletions(-) create mode 100644 sound/soc/omap/omap3pandora.c

7 20

Re: [alsa-devel] Beep sound in the end of audio file
by Peter Ujfalusi 07 May '09

07 May '09

Hello, On Tuesday 05 May 2009 20:03:57 ext Aggarwal, Anuj wrote: > Hi, > > After playing out any audio file on OMAP3 EVM, having TWL4030 codec, > I am hearing a beep sound. I have also tried implementing a mute function > in which I disabled all the inputs/outputs but still that didn't help. > > Any idea how this can be avoided? I have not heard from this kind of problem so far, which does not mean, that it does not exist ;) Can you describe the beep sound? After a quick look I can not pin point the soc board file used with the omap3evm board. Is it in the tree? Does the beep happens in these cases also (after stopping it with Ctrl+C): aplay -f dat /dev/zero aplay -f dat /dev/urandom CC-ing alsa-devel... > > Thanks and Regards, > Anuj Aggarwal > > Platform Support Products > Texas Instruments Incorporated -- Péter

7 14

[alsa-devel] What about my problem on the Dell Dimension 4550, when will that be fixed?
by Kristoffer Grundström 07 May '09

07 May '09

Yes, as the title says: When will my problem with the ALSA-driver that's displayed here? https://qa.mandriva.com/show_bug.cgi?id=49048 /Kristoffer > From: alsa-devel-request(a)alsa-project.org > Subject: Alsa-devel Digest, Vol 27, Issue 23 > To: alsa-devel(a)alsa-project.org > Date: Thu, 7 May 2009 12:00:01 +0200 > > Send Alsa-devel mailing list submissions to > alsa-devel(a)alsa-project.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > or, via email, send a message with subject or body 'help' to > alsa-devel-request(a)alsa-project.org > > You can reach the person managing the list at > alsa-devel-owner(a)alsa-project.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Alsa-devel digest..." > > > Today's Topics: > > 1. Re: Beep sound in the end of audio file (Todd Fischer) > 2. Re: 2.6.29 regression: left audio channel broken after resume > from suspend with Intel HDA (Tino Keitel) > 3. Re: 2.6.29 regression: left audio channel broken after resume > from suspend with Intel HDA (Justin P. Mattock) > 4. Re: [PATCH 2/2] ASoC: cs4270: add power management support > (Mark Brown) > 5. Re: Master vs. Front/Rear/LFE/... elements (Takashi Iwai) > 6. Re: Beep sound in the end of audio file (Arun KS) > 7. Re: Beep sound in the end of audio file (Mark Brown) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 06 May 2009 08:19:49 -0600 > From: Todd Fischer <todd.fischer(a)ridgerun.com> > Subject: Re: [alsa-devel] Beep sound in the end of audio file > To: "Aggarwal, Anuj" <anuj.aggarwal(a)ti.com> > Cc: "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org>, > "linux-omap(a)vger.kernel.org" <linux-omap(a)vger.kernel.org>, Peter > Ujfalusi <peter.ujfalusi(a)nokia.com> > Message-ID: <1241619589.8928.2038.camel(a)sax-lx.fischerfamily.org> > Content-Type: text/plain; charset="UTF-8" > > Hi, > > FWIW: while developing a GSTreamer presentation using a BegleBoard (rev > C), I noticed the same "tuck" sound. It also occurs a second or two > after the GST application exits. I assumed it was some power management > code that was disabling the audio codec after it had gone idle, but > never looked into it. BeagleBoard also uses TWL4030. > > GST command: gst-launch audiotestsrc freq=1000 num-buffers=100 ! > alsasink > > /proc/version: Linux version 2.6.28-omap1 (ddompe@Aleph) (gcc version > 4.3.1 (GCC) ) #2 Thu Mar 5 08:55:58 CST 2009 > > Code image: http://www.beagleboard.org/~arago/esc/esc2009sj.v12.img.gz > > Todd > > On Wed, 2009-05-06 at 19:30 +0530, Aggarwal, Anuj wrote: > > > > -----Original Message----- > > > From: Peter Ujfalusi [mailto:peter.ujfalusi@nokia.com] > > > Sent: Wednesday, May 06, 2009 11:03 AM > > > To: Aggarwal, Anuj > > > Cc: linux-omap(a)vger.kernel.org; alsa-devel(a)alsa-project.org > > > Subject: Re: Beep sound in the end of audio file > > > > > > Hello, > > > > > > On Tuesday 05 May 2009 20:03:57 ext Aggarwal, Anuj wrote: > > > > Hi, > > > > > > > > After playing out any audio file on OMAP3 EVM, having TWL4030 codec, > > > > I am hearing a beep sound. I have also tried implementing a mute function > > > > in which I disabled all the inputs/outputs but still that didn't help. > > > > > > > > Any idea how this can be avoided? > > > > > > I have not heard from this kind of problem so far, which does not mean, that > > > it does not exist ;) > > > Can you describe the beep sound? > > [Aggarwal, Anuj] It's kind of a "tuck" sound coming in the end, after 1 or 2 seconds. > > > > > > After a quick look I can not pin point the soc board file used with the > > > omap3evm board. Is it in the tree? > > [Aggarwal, Anuj] No, it is not. Reason being we already have beagle & > > sdp3430 there, which are both similar to omap3evm, which I pushed. > > So mine got rejected. We had some discussions on how to handle this > > scenario but nothing got finalized. > > > > > > Does the beep happens in these cases also (after stopping it with Ctrl+C): > > > aplay -f dat /dev/zero > > > aplay -f dat /dev/urandom > > [Aggarwal, Anuj] Yes, it comes in both the cases. > > > > > > > > > CC-ing alsa-devel... > > > > > > > > > > > Thanks and Regards, > > > > Anuj Aggarwal > > > > > > > > Platform Support Products > > > > Texas Instruments Incorporated > > > > > > > > > -- > > > P?ter > > > > -- > > To unsubscribe from this list: send the line "unsubscribe linux-omap" in > > the body of a message to majordomo(a)vger.kernel.org > > More majordomo info at http://vger.kernel.org/majordomo-info.html > > > ------------------------------ > > Message: 2 > Date: Thu, 7 May 2009 00:27:53 +0200 > From: Tino Keitel <tino.keitel(a)gmx.de> > Subject: Re: [alsa-devel] 2.6.29 regression: left audio channel broken > after resume from suspend with Intel HDA > To: linux-kernel(a)vger.kernel.org > Cc: Takashi Iwai <tiwai(a)suse.de>, alsa-devel(a)alsa-project.org > Message-ID: <20090506222753.GA6728(a)mac.home> > Content-Type: text/plain; charset=us-ascii > > On Tue, Apr 28, 2009 at 02:23:09 +0200, Tino Keitel wrote: > > [...] > > > I just saw another, major problem: line in doesn't work anymore with > > 2.6.29 and also 2.6.30-rc3 (no usable input level). It works with > > 2.6.27.20. > > FYI: I just tried 2.6.28 and line-in was still broken. So the last > working kernel is 2.6.27. > > Regards, > Tino > > > ------------------------------ > > Message: 3 > Date: Thu, 07 May 2009 00:40:37 -0700 > From: "Justin P. Mattock" <justinmattock(a)gmail.com> > Subject: Re: [alsa-devel] 2.6.29 regression: left audio channel broken > after resume from suspend with Intel HDA > To: Tino Keitel <tino.keitel(a)gmx.de> > Cc: Takashi Iwai <tiwai(a)suse.de>, alsa-devel(a)alsa-project.org, > linux-kernel(a)vger.kernel.org > Message-ID: <1241682037.1864.4.camel@unix> > Content-Type: text/plain > > On Thu, 2009-05-07 at 09:23 +0200, Tino Keitel wrote: > > On Thu, May 07, 2009 at 00:27:53 +0200, Tino Keitel wrote: > > > On Tue, Apr 28, 2009 at 02:23:09 +0200, Tino Keitel wrote: > > > > > > [...] > > > > > > > I just saw another, major problem: line in doesn't work anymore with > > > > 2.6.29 and also 2.6.30-rc3 (no usable input level). It works with > > > > 2.6.27.20. > > > > > > FYI: I just tried 2.6.28 and line-in was still broken. So the last > > > working kernel is 2.6.27. > > > > I tried this: > > > > git bisect start v2.6.28 v2.6.27 sound/pci/hda/patch_sigmatel.c > > > > The result is commit 4f1e6bc3646ab50b8181555ab7e6eeab68b8632a. > > > > I can't use git revert because of conflicts, so I crafted the attached > > patch against 2.6.30-rc4-00288-g413f81e, and now line-in works. > > > > Regards, > > Tino > > It's funny you mention left audio broken > with hda(testing out the imac91.patch, > works good, but as soon as I boot into osx > the left audio speaker sounds blown out) > nice... ehhh... > > (probably nothing of the sort, but > you never know) > > Justin P. Mattock > > > > > ------------------------------ > > Message: 4 > Date: Thu, 7 May 2009 09:02:37 +0100 > From: Mark Brown <broonie(a)sirena.org.uk> > Subject: Re: [alsa-devel] [PATCH 2/2] ASoC: cs4270: add power > management support > To: Timur Tabi <timur(a)freescale.com> > Cc: alsa-devel(a)alsa-project.org > Message-ID: <20090507080236.GA4098(a)sirena.org.uk> > Content-Type: text/plain; charset=us-ascii > > On Wed, May 06, 2009 at 10:27:19AM -0500, Timur Tabi wrote: > > > Acked-by: Timur Tabi <timur(a)freescale.com> > > Applied, thanks. > > > ------------------------------ > > Message: 5 > Date: Thu, 07 May 2009 10:49:22 +0200 > From: Takashi Iwai <tiwai(a)suse.de> > Subject: Re: [alsa-devel] Master vs. Front/Rear/LFE/... elements > To: Lennart Poettering <mznyfn(a)0pointer.de> > Cc: ALSA Development Mailing List <alsa-devel(a)alsa-project.org> > Message-ID: <s5hbpq5s3fx.wl%tiwai(a)suse.de> > Content-Type: text/plain; charset=US-ASCII > > At Wed, 6 May 2009 19:58:24 +0200, > Lennart Poettering wrote: > > > > Heya! > > > > Some cards expose 'Master' volume sliders. Others expose seperate > > (stereo) sliders for 'Front', 'Rear' and so on. I have trouble dealing > > with them properly in PulseAudio: > > > > First of all, on some cards 'Master' seems not to have any effect on > > the actual analog output, only 'Front' and friends do. Is this a bug > > or intended behaviour? > > It's an old feature. AC97 spec gives the "master" volume control only > for front channels. Thus, old boards with AC97 may inherit this > policy. (The problem of emu10k1 is partly this.) > > Fixing it isn't too difficult with vmaster stuff in the driver side, > but this breaks the compatibility, and hard to find the real test > machines nowadays. In short, "don't touch a working system unless it > gets broken" phase. > > > Can I assume that 'Master' and 'Front' are > > always independant? > > No. > > > Secondly, I have trouble supporting the > > 'Front'/'Rear'/'Side'/... elements properly, since they split up the > > surround channels into seperate elements. Now, this is confusing in > > many ways, even for "amixer" which will then show channels such as > > "Rear Front Left" and so on, which obviously make no > > sense. snd_mixer_selem_has_playback_channel() just returns bogus data > > for these cases. Why are those elements seperate anyway? Why aren't > > they combined into a single multi-channel event? > > That's mainly a historical reason. In old days, there are no mixer > apps supporting really multiple channels because the behavior of OSS. > A stereo pair is easier to handle for apps. > > > Looking at the APIs I > > get the idea that the problem appears to be that elements can only > > control all channels the same are all independantly which doesn't > > really match 1:1 on my multichannel sound cards. However, wouldn't it > > be possible to use the 'index' value of a selem_id for this? I.e. have > > a series of controls by the same name but different indexes which > > would then implement snd_mixer_selem_has_playback_channel() correctly? > > i.e. foo,0 would do front-left/right, foo,1 would do rear, foo,2 would > > do lfe, and so? I have no clue how this implemented internally, so not > > sure how feasible this might be. > > This breaks the existing apps. That's the biggest problem we face > now. We can't change the stuff simply because PA isn't the only app > using that API. > > IMO, the best would be a total rewrite of the current mixer API, as I > mentioned some times. Right now it's more complicated than needed, > but not powerful enough to handle exceptional cases. > > I know designing a generic and fully-working API is pretty difficult, > though... > > > thanks, > > Takashi > > > ------------------------------ > > Message: 6 > Date: Thu, 7 May 2009 15:05:08 +0530 > From: Arun KS <getarunks(a)gmail.com> > Subject: Re: [alsa-devel] Beep sound in the end of audio file > To: Mark Brown <broonie(a)sirena.org.uk> > Cc: "Aggarwal, Anuj" <anuj.aggarwal(a)ti.com>, > "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org>, > "linux-omap(a)vger.kernel.org" <linux-omap(a)vger.kernel.org>, Peter > Ujfalusi <peter.ujfalusi(a)nokia.com> > Message-ID: > <dfeb90390905070235i47f9aed4lbd2298ba07a5b989(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On Wed, May 6, 2009 at 7:37 PM, Mark Brown <broonie(a)sirena.org.uk> wrote: > > On Wed, May 06, 2009 at 07:30:03PM +0530, Aggarwal, Anuj wrote: > > > >> > After a quick look I can not pin point the soc board file used with the > >> > omap3evm board. Is it in the tree? > > > >> [Aggarwal, Anuj] No, it is not. Reason being we already have beagle & > >> sdp3430 there, which are both similar to omap3evm, which I pushed. > >> So mine got rejected. We had some discussions on how to handle this > >> scenario but nothing got finalized. > > > > Not from the ALSA side it didn't... > > Mark, > > Since we are not going with the generic driver, are you going to push > this patch: > > http://www.mail-archive.com/linux-omap@vger.kernel.org/msg07166.html > > > -- > > To unsubscribe from this list: send the line "unsubscribe linux-omap" in > > the body of a message to majordomo(a)vger.kernel.org > > More majordomo info at ?http://vger.kernel.org/majordomo-info.html > > > > > ------------------------------ > > Message: 7 > Date: Thu, 7 May 2009 10:38:02 +0100 > From: Mark Brown <broonie(a)sirena.org.uk> > Subject: Re: [alsa-devel] Beep sound in the end of audio file > To: Arun KS <getarunks(a)gmail.com> > Cc: "Aggarwal, Anuj" <anuj.aggarwal(a)ti.com>, > "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org>, > "linux-omap(a)vger.kernel.org" <linux-omap(a)vger.kernel.org>, Peter > Ujfalusi <peter.ujfalusi(a)nokia.com> > Message-ID: <20090507093801.GA27897(a)sirena.org.uk> > Content-Type: text/plain; charset=us-ascii > > On Thu, May 07, 2009 at 03:05:08PM +0530, Arun KS wrote: > > > Since we are not going with the generic driver, are you going to push > > this patch: > > > http://www.mail-archive.com/linux-omap@vger.kernel.org/msg07166.html > > I can't see anything wrong with it looking in the web archive but somene > should send a copy tested against current ASoC. > > > ------------------------------ > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > End of Alsa-devel Digest, Vol 27, Issue 23 > ******************************************

1 0

[alsa-devel] [PATCH 1/2] ASoC: cs4270: introduce CS4270_I2C_INCR
by Daniel Mack 07 May '09

07 May '09

Replace the magic 0x80 value with a suitable macro definition. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Timur Tabi <timur(a)freescale.com> Cc: Mark Brown <broonie(a)sirena.org.uk> --- sound/soc/codecs/cs4270.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 0f453a0..bc338df 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -56,6 +56,7 @@ #define CS4270_FIRSTREG 0x01 #define CS4270_LASTREG 0x08 #define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1) +#define CS4270_I2C_INCR 0x80 /* Bit masks for the CS4270 registers */ #define CS4270_CHIPID_ID 0xF0 @@ -296,7 +297,7 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec) s32 length; length = i2c_smbus_read_i2c_block_data(i2c_client, - CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache); + CS4270_FIRSTREG | CS4270_I2C_INCR, CS4270_NUMREGS, cache); if (length != CS4270_NUMREGS) { dev_err(codec->dev, "i2c read failure, addr=0x%x\n", -- 1.6.2.1

3 14

Re: [alsa-devel] [Intel] RE: How to play second stream in linux
by Yang, Brandon 07 May '09

07 May '09

Hi Takashi, Thanks for your guidance. I had added some code to play second stream in HD audio driver. Our PM would like us to upstream the source code. He thinks it would be more convenient for customer if they can download the code from ALSA. Our source code is base on kernel 2.6.24.2 and specific to ALC262. As you know, some of our code is hard-coded for our platform. I don't know whether this code can meet your requirement or not. If it can meet your requirement, please help me to upstream it. Thanks very much. The attached file is the patch file. Best Regards, Brandon

1 0

[alsa-devel] 1.0.20 release
by Jaroslav Kysela 06 May '09

06 May '09

Hi all, ALSA 1.0.20 packages (except alsa-oss package) are available for download. Changelog is available at http://www.alsa-project.org/main/index.php/Changes_v1.0.19_v1.0.20 . Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

1 0

[alsa-devel] audio mixing
by Gabriel Johnson 06 May '09

06 May '09

I am using alsa libs to mix audio alerts in my application. I am using the aplay code as an example. I am playing two audio files, one that is 5sec and one that is a beep, from two different threads. The thread that plays the beep loops and plays it every second. I am playing the two audio files through the same snd_pcm_t handle. When the 5sec audio file finishes I then the pcm gets into an xrun state. Instead of using the same handle for all audio should I just associate one handle per audio file? Is there something else I can do to make the pcm handle more stable when playing multiple audio files?

1 0

[alsa-devel] dmix and mmap
by Gabriel Johnson 06 May '09

06 May '09

I am currently working on two issues. First, I am trying to add mmaping support to the pcm driver I am using so that I can use the dmix plugin. My driver is a soc driver for the powerpc mpc5121. The pcm_new function preallocates the dma buffers with the following code: static int mpc512x_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; size_t size = mpc512x_pcm_hardware.buffer_bytes_max; buf->dev.type = SNDRV_DMA_TYPE_DEV; buf->dev.dev = pcm->card->dev; buf->private_data = NULL; buf->area = dma_alloc_coherent(pcm->card->dev, size, &buf->addr, GFP_KERNEL); if (!buf->area) return -ENOMEM; buf->bytes = size; return 0; } Given the way the dma buffers are allocated if I just add SNDRV_PCM_INFO_MMAP and SNDRV_PCM_INFO_MMAP_VALID to my snd_pcm_hardware struct with the alsa framework use a generic mmap function or do I still need to implement one? If I need to implement a mmap callback what is the fastest and most simple way to add that support? Second, I need to test dmix with my driver. My codec only supports S16_BE. When I try to use `aplay -Dplug:dmix -f SE16_BE mywave.raw` I get the error `requested or auto-format is not available`. After looking in pcm_direct.c it appears that when I use -Dplug:dmx it ignores -f S16_BE and is defaulting to some format that my codec doesn't support. I don't have a /etc/alsa.conf or ~/.asoundrc file and /proc/asound/modules doesn't exists. Where is aplay getting the default setting for plug:dmix? How can I override those defaults? I am using alsa-lib-1.0.11rc2. Thanks, Gabe

4 7

[alsa-devel] Call out after period buffer is filled
by Jon Smirl 05 May '09

05 May '09

Is there some way to get a call out after a period DMA buffer is filled so that I can add it to the DMA queue? -- Jon Smirl jonsmirl(a)gmail.com

1 1

[alsa-devel] Issues with HP DV4 laptop sound after Ubuntu Jaunty 9.04 upgrade
by Guy Stalnaker 05 May '09

05 May '09

Hello again. I was able to get my internal speakers working for my HP Pavilion Dv4 laptop by following the advice I found in the alsa-devel archives. That advice seems to have worked for quit a few people who found it by a post I made to the Ubuntu Forums. But now, after I have upgraded my Ubuntu from version 8.10 (when sound worked) to version 9.04, sound no longer works correctly. I have some detailed information about the sequencing and the specific error. Attached is an alsa-info.txt output. Here's what I'm seeing (extracts from dmesg/kernel logs): Apr 29 15:13:17 Harlindon kernel: [ 19.937190] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 Apr 29 15:13:17 Harlindon kernel: [ 19.937270] HDA Intel 0000:00:14.2: setting latency timer to 64 Apr 30 09:37:09 Harlindon kernel: [ 19.941675] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 Apr 30 09:37:09 Harlindon kernel: [ 19.941750] HDA Intel 0000:00:14.2: setting latency timer to 64 May 1 16:06:01 Harlindon kernel: [ 19.852324] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 1 16:06:01 Harlindon kernel: [ 19.852398] HDA Intel 0000:00:14.2: setting latency timer to 64 May 1 16:13:43 Harlindon kernel: [ 20.950985] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 1 16:13:43 Harlindon kernel: [ 20.951060] HDA Intel 0000:00:14.2: setting latency timer to 64 Commit Log for Fri May 1 16:41:58 2009 [Upgrade to Ubuntu Jaunty 9.04] May 1 17:11:05 Harlindon kernel: [ 20.318416] hda-intel: unable to grab IRQ 0, disabling device May 1 17:11:05 Harlindon kernel: [ 20.318517] HDA Intel: probe of 0000:00:14.2 failed with error -16 May 1 17:14:18 Harlindon kernel: [ 22.977762] hda-intel: unable to grab IRQ 0, disabling device May 1 17:14:18 Harlindon kernel: [ 22.977828] HDA Intel: probe of 0000:00:14.2 failed with error -16 May 1 17:18:49 Harlindon kernel: [ 21.105013] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 1 17:18:49 Harlindon kernel: [ 21.105087] HDA Intel 0000:00:14.2: setting latency timer to 64 May 1 21:15:14 Harlindon kernel: [ 19.776991] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 1 21:15:14 Harlindon kernel: [ 19.777893] HDA Intel 0000:00:14.2: setting latency timer to 64 May 2 09:22:01 Harlindon kernel: [ 20.540543] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 2 09:22:01 Harlindon kernel: [ 20.540619] HDA Intel 0000:00:14.2: setting latency timer to 64 May 2 09:26:04 Harlindon kernel: [ 19.819578] hda-intel: unable to grab IRQ 0, disabling device May 2 09:26:04 Harlindon kernel: [ 19.819677] HDA Intel: probe of 0000:00:14.2 failed with error -16 May 2 10:37:28 Harlindon kernel: [ 20.175934] hda-intel: unable to grab IRQ 0, disabling device May 2 10:37:28 Harlindon kernel: [ 20.175999] HDA Intel: probe of 0000:00:14.2 failed with error -16 May 2 13:26:27 Harlindon kernel: [ 20.610591] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 2 13:26:27 Harlindon kernel: [ 20.610666] HDA Intel 0000:00:14.2: setting latency timer to 64 May 2 16:57:23 Harlindon kernel: [ 19.885497] hda-intel: unable to grab IRQ 0, disabling device May 2 16:57:23 Harlindon kernel: [ 19.885575] HDA Intel: probe of 0000:00:14.2 failed with error -16 May 2 18:20:18 Harlindon kernel: [ 22.605977] hda-intel: unable to grab IRQ 0, disabling device May 2 18:20:18 Harlindon kernel: [ 22.606042] HDA Intel: probe of 0000:00:14.2 failed with error -16 What is damned peculiar is that is has sometimes worked (as you can see above). I can find no pattern however. I HAVE downloaded an compiled the latest alsa-driver-snapshot. That last reboot has the same unable to grab irq error. Any suggestions are greatly appreciated. -- Guy Stalnaker, I^2@DOIT, 1210 West Dayton Street, Room 3209 CSS, Madison WI 53719-1220, jstalnak(a)wisc.edu, work 608.263.8035, cell 608.235.4718, fax 608.265.6681, page page-guy(a)watchdog.doit.wisc.edu

2 4

[alsa-devel] looking for advice on buying a card
by Dmitri Seletski 05 May '09

05 May '09

Hello Guys I am thinking to buy pci express sound card. Nothing fancy, just something with dolby surround 7.1 and decent sound quality, rather inexpensive, i am not audiophile. I have noticed that there are not that many sound cards out there which have descent support in Linux(unlike many forums claim). I have had MAJOR problems with pci xfi card from creative labs(fatality gamers edition), my buddy still has a lot of problem with it, like major issues with input. No mic - no skype or other voip, you know yourself... I have found source of asus xonar dx(namely ebay), it is not that expensive compared to many other pci express cards out there, still is expensive enough, from alsa matrix it appears to be quite descent. Question is, is it worth buying? Is it painful to get it running?(probably nothing unusual, matter of loading module), does it work fully(does microphone work, does surround sound 7.1 work?)? Is there something better for similar money?(EUR 73 or £65 or $97) Please do not propose incompletely supported cards(while i wouldn't mind submitting issues to mail lists if i come across bugs, i wouldn't want product that doesn't work) I appreciate any comments. Thank you in advance. Dmitri

3 3

[alsa-devel] Driver code with mpc5200 pointer problem.
by Jon Smirl 05 May '09

05 May '09

Here's the code computing the mpc5200 dma pointer. Could you please take a look at it and let me know what it is doing wrong. /* * Freescale MPC5200 Audio DMA * ALSA SoC Platform driver * * Copyright (C) 2008 Secret Lab Technologies Ltd. */ #define DEBUG #include <linux/init.h> #include <linux/module.h> #include <linux/interrupt.h> #include <linux/device.h> #include <linux/delay.h> #include <linux/of_device.h> #include <linux/of_platform.h> #include <linux/dma-mapping.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/initval.h> #include <sound/soc.h> #include <sound/soc-of-simple.h> #include <sysdev/bestcomm/bestcomm.h> #include <sysdev/bestcomm/gen_bd.h> #include <asm/time.h> #include <asm/mpc52xx.h> #include <asm/mpc52xx_psc.h> #include "mpc5200_dma.h" MODULE_AUTHOR("Grant Likely <grant.likely(a)secretlab.ca>"); MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver"); MODULE_LICENSE("GPL"); /* * Interrupt handlers */ static irqreturn_t psc_dma_status_irq(int irq, void *_psc_dma) { struct psc_dma *psc_dma = _psc_dma; struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; u16 isr; isr = in_be16(&regs->mpc52xx_psc_isr); /* Playback underrun error */ if (psc_dma->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP)) psc_dma->stats.underrun_count++; /* Capture overrun error */ if (psc_dma->capture.active && (isr & MPC52xx_PSC_IMR_ORERR)) psc_dma->stats.overrun_count++; out_8(&regs->command, 4 << 4); /* reset the error status */ return IRQ_HANDLED; } /** * psc_dma_bcom_enqueue_next_buffer - Enqueue another audio buffer * @s: pointer to stream private data structure * * Enqueues another audio period buffer into the bestcomm queue. * * Note: The routine must only be called when there is space available in * the queue. Otherwise the enqueue will fail and the audio ring buffer * will get out of sync */ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) { struct bcom_bd *bd; /* Prepare and enqueue the next buffer descriptor */ bd = bcom_prepare_next_buffer(s->bcom_task); bd->status = s->period_bytes; bd->data[0] = s->period_next_pt; bcom_submit_next_buffer(s->bcom_task, NULL); /* Update for next period */ s->period_next_pt += s->period_bytes; if (s->period_next_pt >= s->period_end) s->period_next_pt = s->period_start; } /* Bestcomm DMA irq handler */ static irqreturn_t psc_dma_bcom_irq(int irq, void *_psc_dma_stream) { struct psc_dma_stream *s = _psc_dma_stream; /* For each finished period, dequeue the completed period buffer * and enqueue a new one in it's place. */ while (bcom_buffer_done(s->bcom_task)) { bcom_retrieve_buffer(s->bcom_task, NULL, NULL); s->period_current_pt += s->period_bytes; if (s->period_current_pt >= s->period_end) s->period_current_pt = s->period_start; psc_dma_bcom_enqueue_next_buffer(s); bcom_enable(s->bcom_task); } /* If the stream is active, then also inform the PCM middle layer * of the period finished event. */ if (s->active) { s->jiffies = jiffies; snd_pcm_period_elapsed(s->stream); } return IRQ_HANDLED; } /** * psc_dma_startup: create a new substream * * This is the first function called when a stream is opened. * * If this is the first stream open, then grab the IRQ and program most of * the PSC registers. */ int mpc5200_dma_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; int rc; dev_dbg(psc_dma->dev, "psc_dma_startup(substream=%p)\n", substream); if (!psc_dma->playback.active && !psc_dma->capture.active) { /* Setup the IRQs */ rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED, "psc-dma-status", psc_dma); rc |= request_irq(psc_dma->capture.irq, &psc_dma_bcom_irq, IRQF_SHARED, "psc-dma-capture", &psc_dma->capture); rc |= request_irq(psc_dma->playback.irq, &psc_dma_bcom_irq, IRQF_SHARED, "psc-dma-playback", &psc_dma->playback); if (rc) { free_irq(psc_dma->irq, psc_dma); free_irq(psc_dma->capture.irq, &psc_dma->capture); free_irq(psc_dma->playback.irq, &psc_dma->playback); return -ENODEV; } } return 0; } EXPORT_SYMBOL_GPL(mpc5200_dma_startup); int mpc5200_dma_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { snd_pcm_set_runtime_buffer(substream, NULL); return 0; } EXPORT_SYMBOL_GPL(mpc5200_dma_hw_free); /** * psc_dma_trigger: start and stop the DMA transfer. * * This function is called by ALSA to start, stop, pause, and resume the DMA * transfer of data. */ int mpc5200_dma_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; struct snd_pcm_runtime *runtime = substream->runtime; struct psc_dma_stream *s; struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; u16 imr; u8 psc_cmd; unsigned long flags; if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_dma->capture; else s = &psc_dma->playback; dev_dbg(psc_dma->dev, "psc_dma_trigger(substream=%p, cmd=%i)" " stream_id=%i\n", substream, cmd, substream->pstr->stream); switch (cmd) { case SNDRV_PCM_TRIGGER_START: s->period_bytes = frames_to_bytes(runtime, runtime->period_size); s->period_start = virt_to_phys(runtime->dma_area); s->period_end = s->period_start + (s->period_bytes * runtime->periods); s->period_next_pt = s->period_start; s->period_current_pt = s->period_start; s->active = 1; /* First; reset everything */ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { out_8(&regs->command, MPC52xx_PSC_RST_RX); out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT); } else { out_8(&regs->command, MPC52xx_PSC_RST_TX); out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT); } /* Next, fill up the bestcomm bd queue and enable DMA. * This will begin filling the PSC's fifo. */ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) bcom_gen_bd_rx_reset(s->bcom_task); else bcom_gen_bd_tx_reset(s->bcom_task); while (!bcom_queue_full(s->bcom_task)) psc_dma_bcom_enqueue_next_buffer(s); bcom_enable(s->bcom_task); /* Due to errata in the dma mode; need to line up enabling * the transmitter with a transition on the frame sync * line */ spin_lock_irqsave(&psc_dma->lock, flags); /* first make sure it is low */ while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0) ; /* then wait for the transition to high */ while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0) ; /* Finally, enable the PSC. * Receiver must always be enabled; even when we only want * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */ psc_cmd = MPC52xx_PSC_RX_ENABLE; if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) psc_cmd |= MPC52xx_PSC_TX_ENABLE; out_8(&regs->command, psc_cmd); spin_unlock_irqrestore(&psc_dma->lock, flags); break; case SNDRV_PCM_TRIGGER_STOP: /* Turn off the PSC */ s->active = 0; if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { if (!psc_dma->playback.active) { out_8(&regs->command, 2 << 4); /* reset rx */ out_8(&regs->command, 3 << 4); /* reset tx */ out_8(&regs->command, 4 << 4); /* reset err */ } } else { out_8(&regs->command, 3 << 4); /* reset tx */ out_8(&regs->command, 4 << 4); /* reset err */ if (!psc_dma->capture.active) out_8(&regs->command, 2 << 4); /* reset rx */ } bcom_disable(s->bcom_task); while (!bcom_queue_empty(s->bcom_task)) bcom_retrieve_buffer(s->bcom_task, NULL, NULL); break; default: dev_dbg(psc_dma->dev, "invalid command\n"); return -EINVAL; } /* Update interrupt enable settings */ imr = 0; if (psc_dma->playback.active) imr |= MPC52xx_PSC_IMR_TXEMP; if (psc_dma->capture.active) imr |= MPC52xx_PSC_IMR_ORERR; out_be16(&regs->isr_imr.imr, imr); return 0; } EXPORT_SYMBOL_GPL(mpc5200_dma_trigger); /** * psc_dma_shutdown: shutdown the data transfer on a stream * * Shutdown the PSC if there are no other substreams open. */ void mpc5200_dma_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; dev_dbg(psc_dma->dev, "psc_dma_shutdown(substream=%p)\n", substream); /* * If this is the last active substream, disable the PSC and release * the IRQ. */ if (!psc_dma->playback.active && !psc_dma->capture.active) { /* Disable all interrupts and reset the PSC */ out_be16(&psc_dma->psc_regs->isr_imr.imr, 0); out_8(&psc_dma->psc_regs->command, 3 << 4); /* reset tx */ out_8(&psc_dma->psc_regs->command, 2 << 4); /* reset rx */ out_8(&psc_dma->psc_regs->command, 1 << 4); /* reset mode */ out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */ /* Release irqs */ free_irq(psc_dma->irq, psc_dma); free_irq(psc_dma->capture.irq, &psc_dma->capture); free_irq(psc_dma->playback.irq, &psc_dma->playback); } } EXPORT_SYMBOL_GPL(mpc5200_dma_shutdown); /* --------------------------------------------------------------------- * The PSC DMA 'ASoC platform' driver * * Can be referenced by an 'ASoC machine' driver * This driver only deals with the audio bus; it doesn't have any * interaction with the attached codec */ static const struct snd_pcm_hardware psc_dma_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, .rate_min = 8000, .rate_max = 48000, .channels_min = 1, .channels_max = 2, .period_bytes_max = 1024 * 1024, .period_bytes_min = 32, .periods_min = 2, .periods_max = 256, .buffer_bytes_max = 2 * 1024 * 1024, .fifo_size = 0, }; static int psc_dma_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; struct psc_dma_stream *s; dev_dbg(psc_dma->dev, "psc_dma_pcm_open(substream=%p)\n", substream); if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_dma->capture; else s = &psc_dma->playback; snd_soc_set_runtime_hwparams(substream, &psc_dma_pcm_hardware); s->stream = substream; return 0; } static int psc_dma_pcm_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; struct psc_dma_stream *s; dev_dbg(psc_dma->dev, "psc_dma_pcm_close(substream=%p)\n", substream); if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_dma->capture; else s = &psc_dma->playback; s->stream = NULL; return 0; } static snd_pcm_uframes_t psc_dma_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; struct psc_dma_stream *s; dma_addr_t count; snd_pcm_uframes_t frames; int delta; if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_dma->capture; else s = &psc_dma->playback; count = s->period_current_pt - s->period_start; delta = jiffies - s->jiffies; delta = delta * runtime->rate / HZ; frames = bytes_to_frames(substream->runtime, count); return frames + delta; } static struct snd_pcm_ops psc_dma_pcm_ops = { .open = psc_dma_pcm_open, .close = psc_dma_pcm_close, .ioctl = snd_pcm_lib_ioctl, .pointer = psc_dma_pcm_pointer, }; static u64 psc_dma_pcm_dmamask = 0xffffffff; static int psc_dma_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; size_t size = psc_dma_pcm_hardware.buffer_bytes_max; int rc = 0; dev_dbg(rtd->socdev->dev, "psc_dma_pcm_new(card=%p, dai=%p, pcm=%p)\n", card, dai, pcm); if (!card->dev->dma_mask) card->dev->dma_mask = &psc_dma_pcm_dmamask; if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; if (pcm->streams[0].substream) { rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, size, &pcm->streams[0].substream->dma_buffer); if (rc) goto playback_alloc_err; } if (pcm->streams[1].substream) { rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, size, &pcm->streams[1].substream->dma_buffer); if (rc) goto capture_alloc_err; } if (rtd->socdev->card->codec->ac97) rtd->socdev->card->codec->ac97->private_data = psc_dma; return 0; capture_alloc_err: if (pcm->streams[0].substream) snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); playback_alloc_err: dev_err(card->dev, "Cannot allocate buffer(s)\n"); return -ENOMEM; } static void psc_dma_pcm_free(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; struct snd_pcm_substream *substream; int stream; dev_dbg(rtd->socdev->dev, "psc_dma_pcm_free(pcm=%p)\n", pcm); for (stream = 0; stream < 2; stream++) { substream = pcm->streams[stream].substream; if (substream) { snd_dma_free_pages(&substream->dma_buffer); substream->dma_buffer.area = NULL; substream->dma_buffer.addr = 0; } } } struct snd_soc_platform mpc5200_dma_platform = { .name = "mpc5200-psc-audio", .pcm_ops = &psc_dma_pcm_ops, .pcm_new = &psc_dma_pcm_new, .pcm_free = &psc_dma_pcm_free, }; EXPORT_SYMBOL_GPL(mpc5200_dma_platform); static int __init mpc5200_soc_platform_init(void) { /* Tell the ASoC OF helpers about it */ of_snd_soc_register_platform(&mpc5200_dma_platform); return snd_soc_register_platform(&mpc5200_dma_platform); } module_init(mpc5200_soc_platform_init); static void __exit mpc5200_soc_platform_exit(void) { snd_soc_unregister_platform(&mpc5200_dma_platform); } module_exit(mpc5200_soc_platform_exit); ------------------------------------------------------------------------------- /* * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. * * Copyright 2008 Jon Smirl, Digispeaker * Author: Jon Smirl <jonsmirl(a)gmail.com> * * 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. */ #define DEBUG #include <linux/init.h> #include <linux/module.h> #include <linux/interrupt.h> #include <linux/device.h> #include <linux/delay.h> #include <linux/of_device.h> #include <linux/of_platform.h> #include <linux/dma-mapping.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/initval.h> #include <sound/soc.h> #include <sound/soc-of-simple.h> #include <sysdev/bestcomm/bestcomm.h> #include <sysdev/bestcomm/gen_bd.h> #include <asm/time.h> #include <asm/mpc52xx.h> #include <asm/mpc52xx_psc.h> #include "mpc5200_dma.h" #include "mpc5200_psc_ac97.h" MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); MODULE_DESCRIPTION("mpc5200 AC97 module"); MODULE_LICENSE("GPL"); #define DRV_NAME "mpc5200-psc-ac97" static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { struct psc_dma *psc_dma = ac97->private_data; int timeout; unsigned int val; spin_lock(&psc_dma->lock); //printk("ac97 read: reg %04x\n", reg); /* Wait for it to be ready */ timeout = 1000; while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_CMDSEND) ) udelay(10); if (!timeout) { printk(KERN_ERR DRV_NAME ": timeout on ac97 bus (rdy)\n"); return 0xffff; } /* Do the read */ out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); /* Wait for the answer */ timeout = 1000; while ((--timeout) && !(in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_DATA_VAL) ) udelay(10); if (!timeout) { printk(KERN_ERR DRV_NAME ": timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); return 0xffff; } /* Get the data */ val = in_be32(&psc_dma->psc_regs->ac97_data); if ( ((val>>24) & 0x7f) != reg ) { printk(KERN_ERR DRV_NAME ": reg echo error on ac97 read\n"); return 0xffff; } val = (val >> 8) & 0xffff; //printk("ac97 read ok: reg %04x val %04x\n", reg, val); spin_unlock(&psc_dma->lock); return (unsigned short) val; } static void psc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct psc_dma *psc_dma = ac97->private_data; int timeout; //printk("ac97 write: reg %04x val %04x\n", reg, val); spin_lock(&psc_dma->lock); /* Wait for it to be ready */ timeout = 1000; while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_CMDSEND) ) udelay(10); if (!timeout) { printk(KERN_ERR DRV_NAME ": timeout on ac97 write\n"); return; } /* Write data */ out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); spin_unlock(&psc_dma->lock); } static void psc_ac97_cold_reset(struct snd_ac97 *ac97) { struct psc_dma *psc_dma = ac97->private_data; printk("psc_ac97_cold_reset %p\n", ac97); /* Do a cold reset */ out_8(&psc_dma->psc_regs->op1, MPC52xx_PSC_OP_RES); udelay(10); out_8(&psc_dma->psc_regs->op0, MPC52xx_PSC_OP_RES); udelay(50); /* PSC recover from cold reset (cfr user manual, not sure if useful) */ out_be32(&psc_dma->psc_regs->sicr, in_be32(&psc_dma->psc_regs->sicr)); } static void psc_ac97_warm_reset(struct snd_ac97 *ac97) { printk("psc_ac97_warm_reset\n"); } struct snd_ac97_bus_ops soc_ac97_ops = { .read = psc_ac97_read, .write = psc_ac97_write, .reset = psc_ac97_cold_reset, .warm_reset = psc_ac97_warm_reset, }; EXPORT_SYMBOL_GPL(soc_ac97_ops); #ifdef CONFIG_PM static int psc_ac97_suspend(struct snd_soc_dai *dai) { return 0; } static int psc_ac97_resume(struct snd_soc_dai *dai) { return 0; } #else #define psc_ac97_suspend NULL #define psc_ac97_resume NULL #endif static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" " periods=%i buffer_size=%i buffer_bytes=%i channels=%i" " rate=%i format=%i\n", __func__, substream, params_period_size(params), params_period_bytes(params), params_periods(params), params_buffer_size(params), params_buffer_bytes(params), params_channels(params), params_rate(params), params_format(params)); // FIXME, need a spinlock to protect access if (params_channels(params) == 1) out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); else out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); return 0; } static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { return 0; } /** * psc_ac97_set_fmt: set the serial format. * * This function is called by the machine driver to tell us what serial * format to use. * * This driver only supports AC97 mode. Return an error if the format is * not SND_SOC_DAIFMT_AC97. * * @format: one of SND_SOC_DAIFMT_xxx */ static int psc_ac97_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) { struct psc_dma *psc_dma = cpu_dai->private_data; dev_dbg(psc_dma->dev, "psc_ac97_set_fmt(cpu_dai=%p, format=%i)\n", cpu_dai, format); return (format == SND_SOC_DAIFMT_AC97) ? 0 : -EINVAL; } /* --------------------------------------------------------------------- * ALSA SoC Bindings * * - Digital Audio Interface (DAI) template * - create/destroy dai hooks */ /** * psc_ac97_dai_template: template CPU Digital Audio Interface */ static struct snd_soc_dai_ops psc_ac97_analog_ops = { .startup = mpc5200_dma_startup, .hw_params = psc_ac97_hw_analog_params, .hw_free = mpc5200_dma_hw_free, .shutdown = mpc5200_dma_shutdown, .trigger = mpc5200_dma_trigger, .set_fmt = psc_ac97_set_fmt, }; static struct snd_soc_dai_ops psc_ac97_digital_ops = { .startup = mpc5200_dma_startup, .hw_params = psc_ac97_hw_digital_params, .hw_free = mpc5200_dma_hw_free, .shutdown = mpc5200_dma_shutdown, .trigger = mpc5200_dma_trigger, .set_fmt = psc_ac97_set_fmt, }; static struct snd_soc_dai psc_ac97_dai_template[] = { { .name = "%s analog", .suspend = psc_ac97_suspend, .resume = psc_ac97_resume, .playback = { .channels_min = 1, .channels_max = 6, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S32_BE, }, .capture = { .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S32_BE, }, .ops = &psc_ac97_analog_ops, }, { .name = "%s digital", .playback = { .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE, }, .ops = &psc_ac97_digital_ops, }}; /* --------------------------------------------------------------------- * Sysfs attributes for debugging */ static ssize_t psc_ac97_status_show(struct device *dev, struct device_attribute *attr, char *buf) { struct psc_dma *psc_dma = dev_get_drvdata(dev); return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x " "tfnum=%i tfstat=0x%.4x\n", in_be16(&psc_dma->psc_regs->sr_csr.status), in_be32(&psc_dma->psc_regs->sicr), in_be16(&psc_dma->fifo_regs->rfnum) & 0x1ff, in_be16(&psc_dma->fifo_regs->rfstat), in_be16(&psc_dma->fifo_regs->tfnum) & 0x1ff, in_be16(&psc_dma->fifo_regs->tfstat)); } static int *psc_ac97_get_stat_attr(struct psc_dma *psc_dma, const char *name) { if (strcmp(name, "playback_underrun") == 0) return &psc_dma->stats.underrun_count; if (strcmp(name, "capture_overrun") == 0) return &psc_dma->stats.overrun_count; return NULL; } static ssize_t psc_ac97_stat_show(struct device *dev, struct device_attribute *attr, char *buf) { struct psc_dma *psc_dma = dev_get_drvdata(dev); int *attrib; attrib = psc_ac97_get_stat_attr(psc_dma, attr->attr.name); if (!attrib) return 0; return sprintf(buf, "%i\n", *attrib); } static ssize_t psc_ac97_stat_store(struct device *dev, struct device_attribute *attr, const char *buf, size_t count) { struct psc_dma *psc_dma = dev_get_drvdata(dev); int *attrib; attrib = psc_ac97_get_stat_attr(psc_dma, attr->attr.name); if (!attrib) return 0; *attrib = simple_strtoul(buf, NULL, 0); return count; } static DEVICE_ATTR(status, 0644, psc_ac97_status_show, NULL); static DEVICE_ATTR(playback_underrun, 0644, psc_ac97_stat_show, psc_ac97_stat_store); static DEVICE_ATTR(capture_overrun, 0644, psc_ac97_stat_show, psc_ac97_stat_store); /* --------------------------------------------------------------------- * OF platform bus binding code: * - Probe/remove operations * - OF device match table */ static int __devinit psc_ac97_of_probe(struct of_device *op, const struct of_device_id *match) { phys_addr_t fifo; struct psc_dma *psc_dma; struct resource res; int i, nDAI, size, psc_id, irq, rc; const __be32 *prop; void __iomem *regs; /* Get the PSC ID */ prop = of_get_property(op->node, "cell-index", &size); if (!prop || size < sizeof *prop) return -ENODEV; psc_id = be32_to_cpu(*prop); /* Fetch the registers and IRQ of the PSC */ irq = irq_of_parse_and_map(op->node, 0); if (of_address_to_resource(op->node, 0, &res)) { dev_err(&op->dev, "Missing reg property\n"); return -ENODEV; } regs = ioremap(res.start, 1 + res.end - res.start); if (!regs) { dev_err(&op->dev, "Could not map registers\n"); return -ENODEV; } /* Allocate and initialize the driver private data */ psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL); if (!psc_dma) { iounmap(regs); return -ENOMEM; } printk("psc_dma %p\n", psc_dma); spin_lock_init(&psc_dma->lock); psc_dma->irq = irq; psc_dma->psc_regs = regs; psc_dma->fifo_regs = regs + sizeof *psc_dma->psc_regs; psc_dma->dev = &op->dev; psc_dma->playback.psc_dma = psc_dma; psc_dma->capture.psc_dma = psc_dma; snprintf(psc_dma->name, sizeof psc_dma->name, "PSC%u AC97", psc_id+1); /* Fill out the CPU DAI structure */ nDAI = ARRAY_SIZE(psc_ac97_dai_template); nDAI = min(nDAI, SOC_OF_SIMPLE_MAX_DAI); for (i = 0; i < nDAI; i++) { memcpy(&psc_dma->dai[i], &psc_ac97_dai_template[i], sizeof(struct snd_soc_dai)); psc_dma->dai[i].private_data = psc_dma; snprintf(psc_dma->stream_name[i], PSC_STREAM_NAME_LEN, psc_ac97_dai_template[i].name, psc_dma->name); psc_dma->dai[i].name = psc_dma->stream_name[i]; psc_dma->dai[i].id = psc_id; } /* Find the address of the fifo data registers and setup the * DMA tasks */ fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32); psc_dma->capture.bcom_task = bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512); psc_dma->playback.bcom_task = bcom_psc_gen_bd_tx_init(psc_id, 10, fifo); if (!psc_dma->capture.bcom_task || !psc_dma->playback.bcom_task) { dev_err(&op->dev, "Could not allocate bestcomm tasks\n"); iounmap(regs); kfree(psc_dma); return -ENODEV; } /* Disable all interrupts and reset the PSC */ out_be16(&psc_dma->psc_regs->isr_imr.imr, 0); out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_RX); /* reset receiver */ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_TX); /* reset transmitter */ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_ERR_STAT); /* reset error */ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_SEL_MODE_REG_1); /* reset mode */ /* Do a cold reset of codec */ out_8(&psc_dma->psc_regs->op1, MPC52xx_PSC_OP_RES); udelay(10); out_8(&psc_dma->psc_regs->op0, MPC52xx_PSC_OP_RES); udelay(50); /* Configure the serial interface mode to AC97 */ psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97; out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr); /* No slots active */ out_be32(&psc_dma->psc_regs->ac97_slots, 0x00000000); /* Set up mode register; * First write: RxRdy (FIFO Alarm) generates rx FIFO irq * Second write: register Normal mode for non loopback */ out_8(&psc_dma->psc_regs->mode, 0); out_8(&psc_dma->psc_regs->mode, 0); /* Set the TX and RX fifo alarm thresholds */ out_be16(&psc_dma->fifo_regs->rfalarm, 0x100); out_8(&psc_dma->fifo_regs->rfcntl, 0x4); out_be16(&psc_dma->fifo_regs->tfalarm, 0x100); out_8(&psc_dma->fifo_regs->tfcntl, 0x7); /* Go */ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_TX_ENABLE); out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RX_ENABLE); /* Lookup the IRQ numbers */ psc_dma->playback.irq = bcom_get_task_irq(psc_dma->playback.bcom_task); psc_dma->capture.irq = bcom_get_task_irq(psc_dma->capture.bcom_task); /* Save what we've done so it can be found again later */ dev_set_drvdata(&op->dev, psc_dma); /* Register the SYSFS files */ rc = device_create_file(psc_dma->dev, &dev_attr_status); rc |= device_create_file(psc_dma->dev, &dev_attr_capture_overrun); rc |= device_create_file(psc_dma->dev, &dev_attr_playback_underrun); if (rc) dev_info(psc_dma->dev, "error creating sysfs files\n"); rc = snd_soc_register_dais(psc_dma->dai, nDAI); if (rc != 0) { printk("Failed to register DAI\n"); return 0; } /* Tell the ASoC OF helpers about it */ of_snd_soc_register_cpu_dai(op->node, psc_dma->dai, nDAI); return 0; } static int __devexit psc_ac97_of_remove(struct of_device *op) { struct psc_dma *psc_dma = dev_get_drvdata(&op->dev); dev_dbg(&op->dev, "psc_ac97_remove()\n"); bcom_gen_bd_rx_release(psc_dma->capture.bcom_task); bcom_gen_bd_tx_release(psc_dma->playback.bcom_task); iounmap(psc_dma->psc_regs); iounmap(psc_dma->fifo_regs); kfree(psc_dma); dev_set_drvdata(&op->dev, NULL); return 0; } /* Match table for of_platform binding */ static struct of_device_id psc_ac97_match[] __devinitdata = { { .compatible = "fsl,mpc5200-psc-ac97", }, {} }; MODULE_DEVICE_TABLE(of, psc_ac97_match); static struct of_platform_driver psc_ac97_driver = { .match_table = psc_ac97_match, .probe = psc_ac97_of_probe, .remove = __devexit_p(psc_ac97_of_remove), .driver = { .name = "mpc5200-psc-ac97", .owner = THIS_MODULE, }, }; /* --------------------------------------------------------------------- * Module setup and teardown; simply register the of_platform driver * for the PSC in AC97 mode. */ static int __init psc_ac97_init(void) { return of_register_platform_driver(&psc_ac97_driver); } module_init(psc_ac97_init); static void __exit psc_ac97_exit(void) { of_unregister_platform_driver(&psc_ac97_driver); } module_exit(psc_ac97_exit); -- Jon Smirl jonsmirl(a)gmail.com

5 44

[alsa-devel] Help-Pause/Resume API
by Sudeept Prusti 05 May '09

05 May '09

Hi all, Can anyone educate me which alsa-lib API used for pause and resume. I have a minimal alsa code for pcm playback and i need the needed API for test pause and resume feature on pcm playback. Best Regards, Sudeept The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it.

2 2

[alsa-devel] module snd-atiixp issue with suspend/hibernate/resume
by Ryan Dunn 05 May '09

05 May '09

I have a Compaq V2000 laptop with the ATI IXP chipset and make use of the snd-atiixp module. The laptop needs to have the ac97_quirk option set to 7 to enable the mute LED. After setting this option in the /etc/modprobe.d/options.conf file, the mute LED works after a reboot. However, after a resume from suspend or hibernate, the mute LED does not work. The alsa-info.sh script reports that the quirk is still set with 7. At this point a reboot will NOT fix the problem. The only way to fix it is to remove and reinsert the module with modprobe. This is on a fresh Ubuntu 9.04 install. Any ideas? I wasn't sure if this should be reported here or on the kernel list. I saw a similar issue with the ac97_quirk on the kernel list and they were referred here. I'm a software developer, so I'd be willing to try ideas/possible solutions if you have them. Thanks, Ryan

3 9

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 05 May '09

05 May '09

The following changes since commit 8560b9321f9050968f393ce1ec67e47c1a0bd5cf: Takashi Iwai (1): Merge branch 'fix/asoc' into topic/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Jinyoung Park (1): ASoC: Fix errors in WM8990 Jonathan Cameron (1): ASoC: IMote2 ASoC Support Lopez Cruz, Misael (2): ASoC: TWL4030: Add VDL analog bypass ASoC: TWL4030: Add voice digital loopback: sidetone Mark Brown (15): ASoC: Fix error message formatting in s3c64xx-i2s driver ASoC: Check for supported CPUs when building s3c-i2s-v2 ASoC: Make S3C64xx clock export function to return struct clk ASoC: Fix data format configuration for S3C64xx IISv2 and add 24 bit ASoC: Fix boot warnings from S3C IISv2 ASoC: Allow use of resource from the platform device for S3C IISv2 ASoC: Display the clock rate used as the basis for rate calculation ASoC: Display S3C IISv2 mode and MS errors by default ASoC: Staticise txctrl and rxctrl for S3C IISv2 ASoC: Use platform device resource for S3C64xx IISv2 ASoC: Use a shared define for AC97 CODEC data formats ASoC: Remove unused DAI format defines Merge commit 'takashi/topic/asoc' into for-2.6.31 Merge branch 'for-2.6.30' into for-2.6.31 ASoC: Remove redundant codec pointer from DAIs Peter Ujfalusi (1): ASoC: TWL4030: Add VIBRA output include/sound/soc-dai.h | 28 ++------- sound/soc/codecs/ac97.c | 4 +- sound/soc/codecs/ad1980.c | 4 +- sound/soc/codecs/twl4030.c | 121 +++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm8990.c | 40 +++++++------- sound/soc/codecs/wm9705.c | 4 +- sound/soc/codecs/wm9712.c | 6 +- sound/soc/codecs/wm9713.c | 6 +- sound/soc/pxa/Kconfig | 9 +++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/imote2.c | 114 ++++++++++++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c-i2s-v2.c | 87 ++++++++++++++++++++++++---- sound/soc/s3c24xx/s3c64xx-i2s.c | 15 ++--- sound/soc/s3c24xx/s3c64xx-i2s.h | 4 +- 14 files changed, 362 insertions(+), 82 deletions(-) create mode 100644 sound/soc/pxa/imote2.c

2 19

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 05 May '09

05 May '09

The following changes since commit 6574612fbb34c63117581e68f2231ddce027e41e: Takashi Iwai (1): ASoC: Remove BROKEN from mpc5200 kconfig are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Jinyoung Park (1): ASoC: Fix errors in WM8990 sound/soc/codecs/wm8990.c | 40 ++++++++++++++++++++-------------------- 1 files changed, 20 insertions(+), 20 deletions(-)

2 1

[alsa-devel] [PATCH 0/1] ASoC: TWL4030: VIBRA output support
by Peter Ujfalusi 05 May '09

05 May '09

Hello, The VIBRA output can be driven with audio data or with local vibrator driver, the following patch adds VIBRA output support (for driving the VIBRA output with audio data) for the TWL4030 codec. --- Peter Ujfalusi (1): ASoC: TWL4030: Add VIBRA output sound/soc/codecs/twl4030.c | 61 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 61 insertions(+), 0 deletions(-)

2 3

[alsa-devel] [ASoC PATCH 1/1] IMote 2 ASoC Support
by Jonathan Cameron 04 May '09

04 May '09

From: Jonathan Cameron <jic23(a)cam.ac.uk> IMote2 ASoC Support This patch adds the ASoC side of the board support for the Crossbow IMB400 daughter board. Thanks to Crossbow for considerable assistance. Signed-off-by: Jonathan Cameron <jic23(a)cam.ac.uk> --- Relatively straight forward configuration with the PXA acting as the clock master. sound/soc/pxa/Kconfig | 9 +++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/imote2.c | 131 +++++++++++++++++++++++++++++++++++++++++++++++ diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index ad8a10f..eb75a1c 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -134,3 +134,12 @@ config SND_PXA2XX_SOC_MIOA701 help Say Y if you want to add support for SoC audio on the MIO A701. + +config SND_PXA2XX_SOC_IMOTE2 + tristate "SoC Audio support for IMote 2" + depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 + select SND_PXA2XX_SOC_I2S + select SND_SOC_WM8940 + help + Say Y if you want to add support for SoC audio on the + IMote 2. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 4b90c3c..6e096b4 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -22,6 +22,7 @@ snd-soc-palm27x-objs := palm27x.o snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o +snd-soc-imote2-objs := imote2.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -35,3 +36,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o +obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c new file mode 100644 index 0000000..8476151 --- /dev/null +++ b/sound/soc/pxa/imote2.c @@ -0,0 +1,131 @@ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/pxa27x.h> +#include <mach/hardware.h> + +#include <mach/audio.h> + +#include "../codecs/wm8940.h" +#include "pxa2xx-i2s.h" +#include "pxa2xx-pcm.h" + +static int imote2_asoc_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int clk = 0; + int ret; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + clk = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* CPU should be clock master */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* Disable the PLL */ + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, 0); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set the I2S system clock as input (unused) */ + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, clk, + SND_SOC_CLOCK_OUT); + + return ret; +} + +static struct snd_soc_ops imote2_asoc_ops = { + .hw_params = imote2_asoc_hw_params, +}; + +static struct snd_soc_dai_link imote2_dai = { + .name = "WM8940", + .stream_name = "WM8940", + .cpu_dai = &pxa_i2s_dai, + .codec_dai = &wm8940_dai, + .ops = &imote2_asoc_ops, +}; + +static struct snd_soc_card snd_soc_imote2 = { + .name = "Imote2", + .platform = &pxa2xx_soc_platform, + .dai_link = &imote2_dai, + .num_links = 1, +}; + +static struct snd_soc_device imote2_snd_devdata = { + .card = &snd_soc_imote2, + .codec_dev = &soc_codec_dev_wm8940, +}; + +static struct platform_device *imote2_snd_device; + +static int __init imote2_asoc_init(void) +{ + int ret; + + if (!machine_is_intelmote2()) + return -ENODEV; + imote2_snd_device = platform_device_alloc("soc-audio", -1); + if (!imote2_snd_device) + return -ENOMEM; + + platform_set_drvdata(imote2_snd_device, &imote2_snd_devdata); + imote2_snd_devdata.dev = &imote2_snd_device->dev; + ret = platform_device_add(imote2_snd_device); + if (ret) + platform_device_put(imote2_snd_device); + + return ret; +} +module_init(imote2_asoc_init); + +static void __exit imote2_asoc_exit(void) +{ + platform_device_unregister(imote2_snd_device); +} +module_exit(imote2_asoc_exit); + +MODULE_AUTHOR("Jonathan Cameron"); +MODULE_DESCRIPTION("ALSA SoC Imote 2"); +MODULE_LICENSE("GPL");

2 4

[alsa-devel] [PATCH 1/2] ASoC: TWL4030: Add VDL analog bypass
by Lopez Cruz, Misael 04 May '09

04 May '09

This patch adds voice downlink analog bypass switch. It follows the same approach as in other analog bypass switches. DAC switch is moved from 'DAC Voice' to 'Analog Voice Playback Mixer', that will also allow voice DAC to be powered in digital voice loopback (sidetone). Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 37 +++++++++++++++++++++++++++++++------ 1 files changed, 31 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index efa1a80..efb371f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -468,6 +468,10 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassr2_control = static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control = SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0); +/* Analog bypass for Voice */ +static const struct snd_kcontrol_new twl4030_dapm_abypassv_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0); + /* Digital bypass gain, 0 mutes the bypass */ static const unsigned int twl4030_dapm_dbypass_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -585,7 +589,7 @@ static int bypass_event(struct snd_soc_dapm_widget *w, struct soc_mixer_control *m = (struct soc_mixer_control *)w->kcontrols->private_value; struct twl4030_priv *twl4030 = w->codec->private_data; - unsigned char reg; + unsigned char reg, misc; reg = twl4030_read_reg_cache(w->codec, m->reg); @@ -597,14 +601,28 @@ static int bypass_event(struct snd_soc_dapm_widget *w, else twl4030->bypass_state &= ~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); + } else if (m->reg == TWL4030_REG_VDL_APGA_CTL) { + /* Analog voice bypass */ + if (reg & (1 << m->shift)) + twl4030->bypass_state |= (1 << 4); + else + twl4030->bypass_state &= ~(1 << 4); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) - twl4030->bypass_state |= (1 << (m->shift ? 5 : 4)); + twl4030->bypass_state |= (1 << (m->shift ? 6 : 5)); else - twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4)); + twl4030->bypass_state &= ~(1 << (m->shift ? 6 : 5)); } + /* Enable master analog loopback mode if any analog switch is enabled*/ + misc = twl4030_read_reg_cache(w->codec, TWL4030_REG_MISC_SET_1); + if (twl4030->bypass_state & 0x1F) + misc |= TWL4030_FMLOOP_EN; + else + misc &= ~TWL4030_FMLOOP_EN; + twl4030_write(w->codec, TWL4030_REG_MISC_SET_1, misc); + if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) { if (twl4030->bypass_state) twl4030_codec_mute(w->codec, 0); @@ -935,7 +953,7 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", - TWL4030_REG_AVDAC_CTL, 4, 0), + SND_SOC_NOPM, 0, 0), /* Analog PGAs */ SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, @@ -962,6 +980,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0, &twl4030_dapm_abypassl2_control, bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Voice Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassv_control, + bypass_event, SND_SOC_DAPM_POST_REG), /* Digital bypasses */ SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0, @@ -979,6 +1000,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { 2, 0, NULL, 0), SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer", TWL4030_REG_AVDAC_CTL, + 4, 0, NULL, 0), /* Output MIXER controls */ /* Earpiece */ @@ -1067,13 +1090,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Analog R1 Playback Mixer", NULL, "DAC Right1"}, {"Analog L2 Playback Mixer", NULL, "DAC Left2"}, {"Analog R2 Playback Mixer", NULL, "DAC Right2"}, + {"Analog Voice Playback Mixer", NULL, "DAC Voice"}, {"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"}, {"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"}, {"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"}, {"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"}, - - {"VDL_APGA", NULL, "DAC Voice"}, + {"VDL_APGA", NULL, "Analog Voice Playback Mixer"}, /* Internal playback routings */ /* Earpiece */ @@ -1169,11 +1192,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left1 Analog Loopback", "Switch", "Analog Left Capture Route"}, {"Right2 Analog Loopback", "Switch", "Analog Right Capture Route"}, {"Left2 Analog Loopback", "Switch", "Analog Left Capture Route"}, + {"Voice Analog Loopback", "Switch", "Analog Left Capture Route"}, {"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"}, {"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"}, {"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"}, {"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"}, + {"Analog Voice Playback Mixer", NULL, "Voice Analog Loopback"}, /* Digital bypass routes */ {"Right Digital Loopback", "Volume", "TX1 Capture Route"}, -- 1.5.4.3

3 3

Re: [alsa-devel] realtek ALC272 support
by Andres Salomon 04 May '09

04 May '09

On Tue, 28 Apr 2009 15:41:08 -0400 Andres Salomon <dilinger(a)queued.net> wrote: > On Mon, 27 Apr 2009 18:08:21 +0200 > Takashi Iwai <tiwai(a)suse.de> wrote: > [...] > > > > > > > > At Fri, 24 Apr 2009 15:24:15 -0400, > > > > > > > > Andres Salomon wrote: > > > > > > > > > > > > > > > > > > Hi Kailang, > > > > > > > > > > > > > > > > > > I noticed that your name was on the ALC272 support > > > > > > > > > patch for ALSA intel-hda. This patch basically sets > > > > > > > > > ALC272 to use (mostly) the same code as ALC662. I > > > > > > > > > have two machines that have ALC272, and both of them > > > > > > > > > need verb tables in order to function. I'm wondering > > > > > > > > > if ALC662 should really be used.. > > > > > > > > > > > > > > > > > > Here's one - > > > > > > > > > > > > > > > > > > http://kernel.ubuntu.com/git?p=ubuntu/ubuntu-hardy-lum.git;a=commitdiff;h=7… > > > > > > > > > > > > > > > > > > This isn't the final version of the patch (there are > > > > > > > > > further commits I made in order to support headphone > > > > > > > > > mic stuff), but it gives you an idea of the codec > > > > > > > > > values. The other is: > > > > > > > > > > > > > > > > > > http://kernel.ubuntu.com/git?p=ubuntu/ubuntu-hardy-lum.git;a=commitdiff;h=7… > > > > > > > > > > > > > > > > > > All of these leave me wonder if there's a specific > > > > > > > > > patch_alc272 function that could be written to rid > > > > > > > > > ourselves of these specific quirks. Are there > > > > > > > > > machines with ALC272 that are functional with the > > > > > > > > > current ALSA code? > > > > > > > > [...] > > > > Could you try sound-unstable tree a bit later again? > > I found a bug in my patch, and fixed and updated GIT tree now. > > At least, the headphone plugging should work now. > > > > The mic auto-detection still doesn't work with model=auto, though. > > So, I'm going to take your patch anyway later. But I just wanted > > to be sure that the current tree could work somehow better... > > > > Hi, > > I just updated and tried the current tree; still no sound/headphone > output. :( Ok, I believe I've made some progress on this. The problem appears to be related to the autoconfig handling of the line out nids. The current code ends up with something like the following: ALSA sound/pci/hda/hda_codec.c:3833: autoconfig: line_outs=1 (0x17/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3837: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3841: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3842: mono: mono_out=0x0 ALSA sound/pci/hda/hda_codec.c:3853: inputs: mic=0x18, fmic=0x19, line=0x0, fline=0x0, cd=0x0, aux=0x0 However, NID 0x17 is actually a speaker_out. The code that checks for line_outs in snd_hda_parse_pin_def_config() unsets the speaker_out and uses that NID for a line_out. For whatever reason, this breaks things (no sound output, no headphone out). According to the ALC272 data sheet (the block diagram on page 5), there are actually 2 line_out NIDs; 0x14 and 0x15. The default pin configs for these two NIDs (0x411111f0 for both) have the Port Connectivity field set to AC_JACK_PORT_NONE, so the auto-config skips those NIDs. If I provide an override to a) set Port Connectivity to AC_JACK_PORT_COMPLEX, and b) set Default Device to AC_JACK_LINE_OUT (from AC_JACK_SPEAKER), I get usable audio (speakers work, headphone works, headphone plugging mutes the speaker properly, and even the mic works). The patch which I've used is below. Clearly, this isn't a very nice solution, so the question is: is there a way to provide an override/quirk for pincfgs (I see one for HWDEP, but that's configuration-specific) within the current hda_codec framework for this specific codec/laptop? Alternatively, I can try and figure out what's wrong with the code in the case where line_out_pins[0] is set to the speaker_out_pins[0] (ie, NID 0x17 in this case). I can check the other ALC272 machine tomorrow (the DELL_ZM1) to see if the line_out hack enables it to work without the verb table. diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b91f6ed..5aa7a0c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3644,6 +3644,15 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, continue; def_conf = snd_hda_codec_get_pincfg(codec, nid); + snd_printd("nid %x: pincfg def_conf=%x\n", nid, def_conf); + if (codec->vendor_id == 0x10ec0272 && + (nid == 0x14 || nid == 0x15)) { + def_conf &= ~(0x3 << AC_DEFCFG_PORT_CONN_SHIFT); + def_conf &= ~(0xf << AC_DEFCFG_DEVICE_SHIFT); + snd_printd("nid %x: overriding def_conf, now %x\n", nid, + def_conf); + } + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; loc = get_defcfg_location(def_conf);

2 12

[alsa-devel] Adding MMAP support to alsa driver
by Gabriel Johnson 04 May '09

04 May '09

What structures need to be filled out in the hardware driver to support MMAP calls?

2 1

[alsa-devel] Too big adjustment messages on hda-intel with ALC883
by Ozan Çağlayan 04 May '09

04 May '09

When I launch pulseaudio, a bunch of the following line is dropped to dmesg: [ 1472.847085] ALSA ../../alsa-kernel/pci/hda/hda_intel.c:1102: Too big adjustment 32 [ 1472.998865] hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. alsa-info output is at: http://www.alsa-project.org/db/?f=fd7037731d908e4287e1d457de6a1483062b2130 I'm using 2.6.30_rc3 with: - 20090421 snapshot of alsa-driver, - 20090410 snapshot of alsa-lib. Pulseaudio version is 0.9.15, timer-based scheduling is enabled. Regards, Ozan Caglayan http://www.pardus.org.tr/eng

2 1

[alsa-devel] [PATCH] sc6000: enable joystick port
by Krzysztof Helt 04 May '09

04 May '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Add module parameter to enable or disable joystick port (gameport) on the SC6600 and later cards. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The second version of the patch with fixed the bug spotted Takashi. Documentation/sound/alsa/ALSA-Configuration.txt | 4 +++- sound/isa/sc6000.c | 9 +++++++-- 2 files changed, 10 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 012858d..11a80a9 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1543,13 +1543,15 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-sc6000 ----------------- - Module for Gallant SC-6000 soundcard. + Module for Gallant SC-6000 soundcard and later models: SC-6600 + and SC-7000. port - Port # (0x220 or 0x240) mss_port - MSS Port # (0x530 or 0xe80) irq - IRQ # (5,7,9,10,11) mpu_irq - MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq dma - DMA # (1,3,0) + joystick - Enable gameport - 0 = disable (default), 1 = enable This module supports multiple cards. diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index c803b2e..9a8bbf6 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -56,6 +56,7 @@ static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x300, 0x310, 0x320, 0x330 */ static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 0 */ static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0, 1, 3 */ +static bool joystick[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = false }; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for sc-6000 based soundcard."); @@ -75,6 +76,8 @@ module_param_array(mpu_irq, int, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for sc-6000 driver."); module_param_array(dma, int, NULL, 0444); MODULE_PARM_DESC(dma, "DMA # for sc-6000 driver."); +module_param_array(joystick, bool, NULL, 0444); +MODULE_PARM_DESC(joystick, "Enable gameport."); /* * Commands of SC6000's DSP (SBPRO+special). @@ -363,7 +366,7 @@ static int __devinit sc6000_init_mss(char __iomem *vport, int config, static void __devinit sc6000_hw_cfg_encode(char __iomem *vport, int *cfg, long xport, long xmpu, - long xmss_port) + long xmss_port, int joystick) { cfg[0] = 0; cfg[1] = 0; @@ -376,6 +379,8 @@ static void __devinit sc6000_hw_cfg_encode(char __iomem *vport, int *cfg, if (xmss_port == 0xe80) cfg[0] |= 0x10; cfg[0] |= 0x40; /* always set */ + if (!joystick) + cfg[0] |= 0x02; cfg[1] |= 0x80; /* enable WSS system */ cfg[1] &= ~0x40; /* disable IDE */ snd_printd("hw cfg %x, %x\n", cfg[0], cfg[1]); @@ -427,7 +432,7 @@ static int __devinit sc6000_init_board(char __iomem *vport, if (!old) { int cfg[2]; sc6000_hw_cfg_encode(vport, &cfg[0], port[dev], mpu_port[dev], - mss_port[dev]); + mss_port[dev], joystick[dev]); if (sc6000_hw_cfg_write(vport, cfg) < 0) { snd_printk(KERN_ERR "sc6000_hw_cfg_write: failed!\n"); return -EIO; -- 1.6.0.3 ---------------------------------------------------------------------- Wygraj 3000 zl, wycieczke lub lot balonem! Sprawdz >> http://link.interia.pl/f2154

2 1

[alsa-devel] [[ALSA] Patch for HP dv4-1222nr laptop 1/1] Addition for HP dv4-1222nr laptop.
by renidragsemaj＠yahoo.com 04 May '09

04 May '09

From: James Gardiner <renidragsemaj(a)yahoo.com> Signed-off-by: James Gardiner <renidragsemaj(a)yahoo.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 917bc5d..27bc568 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -100,6 +100,7 @@ enum { STAC_HP_M4, STAC_HP_DV5, STAC_HP_HDX, + STAC_HP_DV4_1222NR, STAC_92HD71BXX_MODELS }; @@ -1836,6 +1837,7 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_HP_M4] = NULL, [STAC_HP_DV5] = NULL, [STAC_HP_HDX] = NULL, + [STAC_HP_DV4_1222NR] = NULL, }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { @@ -1847,6 +1849,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_HP_M4] = "hp-m4", [STAC_HP_DV5] = "hp-dv5", [STAC_HP_HDX] = "hp-hdx", + [STAC_HP_DV4_1222NR] = "hp-dv4-1222nr", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { @@ -1855,6 +1858,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb, + "HP dv4-1222nr", STAC_HP_DV4_1222NR), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080, "HP", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0, @@ -4520,27 +4525,38 @@ static int stac92xx_resume(struct hda_codec *codec) return 0; } - /* - * using power check for controlling mute led of HP HDX notebooks + * using power check for controlling mute led of HP notebooks * check for mute state only on Speakers (nid = 0x10) * * For this feature CONFIG_SND_HDA_POWER_SAVE is needed, otherwise * the LED is NOT working properly ! + * + * Changed name to reflect that it now works for any designated + * model, not just HP HDX. */ #ifdef CONFIG_SND_HDA_POWER_SAVE -static int stac92xx_hp_hdx_check_power_status(struct hda_codec *codec, +static int stac92xx_hp_check_power_status(struct hda_codec *codec, hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; + unsigned int gpio_bit = 0; /* gets rid of compiler warning */ + + switch (spec->board_config) { + case STAC_HP_DV4_1222NR: + gpio_bit = 0x01; + break; + case STAC_HP_HDX: + gpio_bit = 0x08; + } if (nid == 0x10) { if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) - spec->gpio_data &= ~0x08; /* orange */ + spec->gpio_data &= ~gpio_bit; /* orange */ else - spec->gpio_data |= 0x08; /* white */ + spec->gpio_data |= gpio_bit; /* white */ stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, @@ -5219,6 +5235,22 @@ again: spec->num_smuxes = 0; spec->num_dmuxes = 1; break; + case STAC_HP_DV4_1222NR: + spec->num_dmics = 1; + /* I don't know if it needs 1 or 2 smuxes - will wait for + * bug reports to fix if needed + */ + spec->num_smuxes = 1; + spec->num_dmuxes = 1; +#ifdef CONFIG_SND_HDA_POWER_SAVE + /* This controls MUTE LED */ + spec->gpio_mask |= 0x01; + spec->gpio_dir |= 0x01; + spec->gpio_data |= 0x01; + codec->patch_ops.check_power_status = + stac92xx_hp_check_power_status; +#endif + /* fallthrough */ case STAC_HP_DV5: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); @@ -5239,7 +5271,7 @@ again: /* register check_power_status callback. */ codec->patch_ops.check_power_status = - stac92xx_hp_hdx_check_power_status; + stac92xx_hp_check_power_status; #endif break; }; -- 1.5.6.3

2 1

[alsa-devel] [PATCH 2/2] ASoC: TWL4030: Add voice digital loopback: sidetone
by Lopez Cruz, Misael 04 May '09

04 May '09

This patch add voice digital loopback (sidetone) to the twl4030 driver. It mixes voice uplink attenuated (by sidetone gain) with voice downlink when the codec is working in option2 (voice/audio mode). Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 27 +++++++++++++++++++++++++-- 1 files changed, 25 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index efb371f..23bae74 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -491,6 +491,18 @@ static const struct snd_kcontrol_new twl4030_dapm_dbypassr_control = TWL4030_REG_ATX2ARXPGA, 0, 7, 0, twl4030_dapm_dbypass_tlv); +/* + * Voice Sidetone GAIN volume control: + * from -51 to -10 dB in 1 dB steps (mute instead of -51 dB) + */ +static DECLARE_TLV_DB_SCALE(twl4030_dapm_dbypassv_tlv, -5100, 100, 1); + +/* Digital bypass voice: sidetone (VUL -> VDL)*/ +static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control = + SOC_DAPM_SINGLE_TLV("Volume", + TWL4030_REG_VSTPGA, 0, 0x29, 0, + twl4030_dapm_dbypassv_tlv); + static int micpath_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -607,12 +619,18 @@ static int bypass_event(struct snd_soc_dapm_widget *w, twl4030->bypass_state |= (1 << 4); else twl4030->bypass_state &= ~(1 << 4); + } else if (m->reg == TWL4030_REG_VSTPGA) { + /* Voice digital bypass */ + if (reg) + twl4030->bypass_state |= (1 << 5); + else + twl4030->bypass_state &= ~(1 << 5); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) - twl4030->bypass_state |= (1 << (m->shift ? 6 : 5)); + twl4030->bypass_state |= (1 << (m->shift ? 7 : 6)); else - twl4030->bypass_state &= ~(1 << (m->shift ? 6 : 5)); + twl4030->bypass_state &= ~(1 << (m->shift ? 7 : 6)); } /* Enable master analog loopback mode if any analog switch is enabled*/ @@ -991,6 +1009,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0, &twl4030_dapm_dbypassr_control, bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Voice Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassv_control, bypass_event, + SND_SOC_DAPM_POST_REG), SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0), @@ -1203,9 +1224,11 @@ static const struct snd_soc_dapm_route intercon[] = { /* Digital bypass routes */ {"Right Digital Loopback", "Volume", "TX1 Capture Route"}, {"Left Digital Loopback", "Volume", "TX1 Capture Route"}, + {"Voice Digital Loopback", "Volume", "TX2 Capture Route"}, {"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"}, {"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"}, + {"Analog Voice Playback Mixer", NULL, "Voice Digital Loopback"}, }; -- 1.5.4.3

2 1

[alsa-devel] Issues with HP DV4 laptop sound after Ubuntu Jaunty 9.04 upgrade
by Guy Stalnaker 03 May '09

03 May '09

Hello again. I was able to get my internal speakers working for my HP Pavilion Dv4 laptop by following the advice I found in the alsa-devel archives. That advice seems to have worked for quit a few people who found it by a post I made to the Ubuntu Forums. But now, after I have upgraded my Ubuntu from version 8.10 (when sound worked) to version 9.04, sound no longer works correctly. I have some detailed information about the sequencing and the specific error. Attached is an alsa-info.txt output. Here's what I'm seeing (extracts from dmesg/kernel logs): Apr 29 15:13:17 Harlindon kernel: [ 19.937190] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 Apr 29 15:13:17 Harlindon kernel: [ 19.937270] HDA Intel 0000:00:14.2: setting latency timer to 64 Apr 30 09:37:09 Harlindon kernel: [ 19.941675] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 Apr 30 09:37:09 Harlindon kernel: [ 19.941750] HDA Intel 0000:00:14.2: setting latency timer to 64 May 1 16:06:01 Harlindon kernel: [ 19.852324] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 1 16:06:01 Harlindon kernel: [ 19.852398] HDA Intel 0000:00:14.2: setting latency timer to 64 May 1 16:13:43 Harlindon kernel: [ 20.950985] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 1 16:13:43 Harlindon kernel: [ 20.951060] HDA Intel 0000:00:14.2: setting latency timer to 64 Commit Log for Fri May 1 16:41:58 2009 [Upgrade to Ubuntu Jaunty 9.04] May 1 17:11:05 Harlindon kernel: [ 20.318416] hda-intel: unable to grab IRQ 0, disabling device May 1 17:11:05 Harlindon kernel: [ 20.318517] HDA Intel: probe of 0000:00:14.2 failed with error -16 May 1 17:14:18 Harlindon kernel: [ 22.977762] hda-intel: unable to grab IRQ 0, disabling device May 1 17:14:18 Harlindon kernel: [ 22.977828] HDA Intel: probe of 0000:00:14.2 failed with error -16 May 1 17:18:49 Harlindon kernel: [ 21.105013] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 1 17:18:49 Harlindon kernel: [ 21.105087] HDA Intel 0000:00:14.2: setting latency timer to 64 May 1 21:15:14 Harlindon kernel: [ 19.776991] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 1 21:15:14 Harlindon kernel: [ 19.777893] HDA Intel 0000:00:14.2: setting latency timer to 64 May 2 09:22:01 Harlindon kernel: [ 20.540543] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 2 09:22:01 Harlindon kernel: [ 20.540619] HDA Intel 0000:00:14.2: setting latency timer to 64 May 2 09:26:04 Harlindon kernel: [ 19.819578] hda-intel: unable to grab IRQ 0, disabling device May 2 09:26:04 Harlindon kernel: [ 19.819677] HDA Intel: probe of 0000:00:14.2 failed with error -16 May 2 10:37:28 Harlindon kernel: [ 20.175934] hda-intel: unable to grab IRQ 0, disabling device May 2 10:37:28 Harlindon kernel: [ 20.175999] HDA Intel: probe of 0000:00:14.2 failed with error -16 May 2 13:26:27 Harlindon kernel: [ 20.610591] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 May 2 13:26:27 Harlindon kernel: [ 20.610666] HDA Intel 0000:00:14.2: setting latency timer to 64 May 2 16:57:23 Harlindon kernel: [ 19.885497] hda-intel: unable to grab IRQ 0, disabling device May 2 16:57:23 Harlindon kernel: [ 19.885575] HDA Intel: probe of 0000:00:14.2 failed with error -16 May 2 18:20:18 Harlindon kernel: [ 22.605977] hda-intel: unable to grab IRQ 0, disabling device May 2 18:20:18 Harlindon kernel: [ 22.606042] HDA Intel: probe of 0000:00:14.2 failed with error -16 What is damned peculiar is that is has sometimes worked (as you can see above). I can find no pattern however. I HAVE downloaded an compiled the latest alsa-driver-snapshot. That last reboot has the same unable to grab irq error. Any suggestions are greatly appreciated. -- Guy Stalnaker, I^2@DOIT, 1210 West Dayton Street, Room 3209 CSS, Madison WI 53719-1220, jstalnak(a)wisc.edu, work 608.263.8035, cell 608.235.4718, fax 608.265.6681, page page-guy(a)watchdog.doit.wisc.edu upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.56 !!################################ !!Script ran on: Sun May 3 04:38:21 UTC 2009 !!Linux Distribution !!------------------ Ubuntu 9.04 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 9.04" !!Kernel Information !!------------------ Kernel release: 2.6.28-6-386 Operating System: GNU/Linux Architecture: i686 Processor: unknown SMP Enabled: No !!ALSA Version !!------------ Driver version: 1.0.19 Library version: 1.0.18 Utilities version: 1.0.18 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No aRts: Installed - Yes (/usr/bin/artsd) Running - No Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xd2410000 irq 19 !!PCI Soundcards installed in the system !!-------------------------------------- 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) 01:05.1 Audio device: ATI Technologies Inc RS780 Azalia controller !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:14.2 0403: 1002:4383 Subsystem: 103c:30fb -- 01:05.1 0403: 1002:960f Subsystem: 1002:960f !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd_hda_intel: model=hp-dv5 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 32,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 model : hp-dv5,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: ATI RS690/780 HDMI Address: 0 Function Id: 0x1 Vendor Id: 0x1002791a Subsystem Id: 0x00791a00 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x40]: 48000 bits [0x2]: 16 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x201: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 Node 0x03 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000094: OUT Detect HDMI Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x02 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 6 May 2 18:25 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 5 May 2 18:25 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 4 May 2 18:25 /dev/snd/pcmC0D3p crw-rw----+ 1 root audio 116, 3 May 2 18:25 /dev/snd/seq crw-rw----+ 1 root audio 116, 2 May 2 18:25 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** !!Amixer output !!------------- !!-------Mixer controls for card 0 [HDMI] Card hw:0 'HDMI'/'HDA ATI HDMI at 0xd2410000 irq 19' Mixer name : 'ATI RS690/780 HDMI' Components : 'HDA:1002791a,00791a00,00100000' Controls : 4 Simple ctrls : 1 Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] !!Alsactl output !!------------- --startcollapse-- state.HDMI { control.1 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.2 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.3 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value false } } --endcollapse-- !!All Loaded Modules !!------------------ Module ppp_deflate zlib_deflate bsd_comp ppp_async crc_ccitt ppp_generic slhc binfmt_misc ppdev rfcomm bridge stp bnep sco l2cap bluetooth ipv6 container sbs pci_slot sbshc rfkill input_polldev cpufreq_powersave cpufreq_stats cpufreq_userspace cpufreq_ondemand cpufreq_conservative freq_table cbc dm_crypt sha256_generic sha1_generic sha512_generic aes_i586 aes_generic lp parport joydev snd_hda_codec_atihdmi snd_hda_intel snd_seq_dummy snd_hda_codec snd_seq_oss snd_seq_midi snd_hwdep snd_rawmidi snd_pcm_oss snd_mixer_oss snd_seq_midi_event psmouse i2c_piix4 snd_pcm uvcvideo compat_ioctl32 serio_raw snd_seq i2c_core sierra snd_timer snd_seq_device videodev v4l1_compat usblp evdev usbserial fglrx snd soundcore snd_page_alloc agpgart battery leds_hp_disk led_class shpchp pci_hotplug video output lis3lv02d sdhci_pci sdhci mmc_core ac button reiserfs ata_generic pata_acpi pata_atiixp sg sr_mod cdrom sd_mod crc_t10dif usbhid hid atiixp ahci libata r8169 mii usb_storage scsi_mod dm_mirror dm_region_hash dm_log dm_snapshot dm_mod thermal processor fan fbcon tileblit font bitblit softcursor !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x03 0x18560010 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs:

1 0

Re: [alsa-devel] [Alsa-user] looking for advice on buying a card
by Dmitri Seletski 02 May '09

02 May '09

Hi Gary. Well, thank you for the idea, but i somehow not comfortable with it. I have to buy this thing, even if it's inexpensive, i will have to pay good amount of money for delivery. I am looking around 20 euro, not counting price for device itself. Than again, i have to buy new card and pay for delivery there. To make things worse, none xfi sound cards are no longer produced by creative labs and i really don't like idea of second hand sound cards(even though you have to be very skillful to break one of those). So it simply makes sense to pay a bit extra(i was loosing those money with that pcie to pci extentsion and its delivery anyway) but to get that card I was referring before, Asus Xonar DX I know it's supported by alsa, because of this: http://www.alsa-project.org/main/index.php/Matrix:Vendor-Asus I am just double checking, since this specific card is really promising one and i really feel like buying one. But again, i am opened for other options. Thanks a lot for your thoughts. I have kinda got used to USB cards by now myself and all those troubles they cause :) gary wrote: > Having spent an ungodly obscene amount of time trying to get two > different USB audio cards to work in 64 bit opensuse, I suggest not > trying to be a pioneer. I solved my need for multiple sound cards > (exceeding the number of PCI slots I have) by using a PCI bridge. > > For one soundcard, you can try thing: > http://www.siig.com/ViewProduct.aspx?pn=JJ-000111-S1 > It doesn't list linux in support unfortunately. However, the latest > linux kernal might recognize the card. Then pick a soundcard from the > list of supported devices. > > For my particular situation, I bought a Magma PCI expander on ebay. I > plugged it in and it worked with the soundcards I had on hand (Diamond > 5.1 using C-Media chipset). With the shipping, it worked out to be > about $39 a PCI slot. [My time, about 10 minutes, not counting > searching ebay.] Thus the price for the SIIG seems reasonable if you > only need to run one card. However, I can't vouch it will work. The > Magma boxes do work, but are obscenely expensive new and bulky. > > > > Dmitri Seletski wrote: >> Hello Guys >> I am thinking to buy pci express sound card. Nothing fancy, just >> something with dolby surround 7.1 and decent sound quality, rather >> inexpensive, i am not audiophile. >> >> I have noticed that there are not that many sound cards out there which >> have descent support in Linux(unlike many forums claim). I have had >> MAJOR problems with pci xfi card from creative labs(fatality gamers >> edition), my buddy still has a lot of problem with it, like major issues >> with input. No mic - no skype or other voip, you know yourself... >> >> I have found source of asus xonar dx(namely ebay), it is not that >> expensive compared to many other pci express cards out there, still is >> expensive enough, from alsa matrix it appears to be quite descent. >> Question is, is it worth buying? Is it painful to get it >> running?(probably nothing unusual, matter of loading module), does it >> work fully(does microphone work, does surround sound 7.1 work?)? >> Is there something better for similar money?(EUR 73 or £65 or $97) >> Please do not propose incompletely supported cards(while i wouldn't mind >> submitting issues to mail lists if i come across bugs, i wouldn't want >> product that doesn't work) >> >> I appreciate any comments. >> >> Thank you in advance. >> Dmitri >> >> ------------------------------------------------------------------------------ >> >> Register Now & Save for Velocity, the Web Performance & Operations >> Conference from O'Reilly Media. Velocity features a full day of >> expert-led, hands-on workshops and two days of sessions from industry >> leaders in dedicated Performance & Operations tracks. Use code >> vel09scf and Save an extra 15% before 5/3. >> http://p.sf.net/sfu/velocityconf >> _______________________________________________ >> Alsa-user mailing list >> Alsa-user(a)lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/alsa-user >> >> >> ------------------------------------------------------------------------ >> >> >> No virus found in this incoming message. >> Checked by AVG - www.avg.com Version: 8.0.238 / Virus Database: >> 270.12.13/2091 - Release Date: 05/01/09 17:52:00 >> > ------------------------------------------------------------------------ > > > No virus found in this outgoing message. > Checked by AVG - www.avg.com > Version: 8.0.238 / Virus Database: 270.12.15/2093 - Release Date: 05/02/09 14:23:00 > >

1 0

[alsa-devel] New codec for STAC9766 used on Efika
by Jon Smirl 02 May '09

02 May '09

Could you please review this new STAC9766 codec implementation. Some of the controls aren't where they need to be. For example the mutes on the mixer inputs. I'm not sure where they are supposed to go. How should SPDIF support work? The hardware supports simultaneous use of analog and SPDIF. Do I want one DAI or two? If I have two DAI then ALSA needs to interleave the samples into a single DMA stream. So if one stream isn't active it needs to be filled with silence. Other options... A single DAI with a switch A single DAI with the same output stream on both analog and SPDIF. I've tried to add the DAPM routes. I don't know of anyone using this chip on a low power design. Datasheet is here: http://www.idt.com/products/getDoc.cfm?docID=13134007 I get a few errors from the current config: Failed to add route All Analog Mixer->Pop Bypass Mux dapm: STAC9766: configuring unknown pin OUT3 dapm: STAC9766: configuring unknown pin MONOOUT Failed to add route HPOUTL->Headphone Jack dapm: STAC9766: configuring unknown pin OUT3 dapm: STAC9766: configuring unknown pin MONOOUT Failed to add route HPOUTL->Headphone Jack /* * stac9766.c -- ALSA Soc STAC9766 codec support * * Copyright 2008 Jon Smirl, Digispeaker * Author: Jon Smirl <jonsmirl(a)gmail.com> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * * Features:- * * o Support for AC97 Codec, S/PDIF * o Support for DAPM */ #include <linux/init.h> #include <linux/module.h> #include <linux/device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> #include <sound/initval.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/soc-of-simple.h> #include "stac9766.h" #define STAC9766_VERSION "0.10" /* * STAC9766 register cache */ static const u16 stac9766_reg[] = { 0x6A90, 0x8000, 0x8000, 0x8000, // 6 0x0000, 0x0000, 0x8008, 0x8008, // e 0x8808, 0x8808, 0x8808, 0x8808, // 16 0x8808, 0x0000, 0x8000, 0x0000, // 1e 0x0000, 0x0000, 0x0000, 0x000f, // 26 0x0a05, 0x0400, 0xbb80, 0x0000, // 2e 0x0000, 0xbb80, 0x0000, 0x0000, // 36 0x0000, 0x2000, 0x0000, 0x0100, // 3e 0x0000, 0x0000, 0x0080, 0x0000, // 46 0x0000, 0x0000, 0x0003, 0xffff, // 4e 0x0000, 0x0000, 0x0000, 0x0000, // 56 0x4000, 0x0000, 0x0000, 0x0000, // 5e 0x1201, 0xFFFF, 0xFFFF, 0x0000, // 66 0x0000, 0x0000, 0x0000, 0x0000, // 6e 0x0000, 0x0000, 0x0000, 0x0006, // 76 0x0000, 0x0000, 0x0000, 0x0000, // 7e }; static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; static const char *stac9766_3D_separation[] = {"Off", "Low", "Medium", "High"}; static const struct soc_enum stac9766_enum[] = { SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux), /* Record Mux 0 */ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux), /* Mono Mux 1 */ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux), /* Mic1/2 Mux 2 */ SOC_ENUM_SINGLE(AC97_SENSE_INFO, 1, 2, stac9766_SPDIF_mux), /* SPDIF Mux 3 */ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux), /* Pop Bypass Mux 4 */ SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux), /* Record All Mux 5 */ SOC_ENUM_SINGLE(AC97_3D_CONTROL, 2, 4, stac9766_3D_separation), /* 3D Separation 7 */ }; static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1), SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1), SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1), SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), SOC_SINGLE("Mixer PC Beep Volume", AC97_PC_BEEP, 1, 15, 1), SOC_SINGLE("PC Beep Frequency", AC97_PC_BEEP, 5, 127, 1), SOC_SINGLE("Mixer Phone Volume", AC97_PHONE, 0, 31, 1), SOC_SINGLE("Mixer Mic Volume", AC97_MIC, 0, 31, 1), SOC_SINGLE("Mic Boost", AC97_MIC, 6, 1, 1), SOC_SINGLE("Mic Gain", AC97_STAC_ANALOG_SPECIAL, 2, 1, 1), SOC_SINGLE("Stereo Mic", AC97_STAC_STEREO_MIC, 0, 1, 1), SOC_DOUBLE("Mixer Line Volume", AC97_LINE, 8, 0, 31, 1), SOC_DOUBLE("Mixer CD Volume", AC97_CD, 8, 0, 31, 1), SOC_DOUBLE("Mixer Video Volume", AC97_VIDEO, 8, 0, 31, 1), SOC_DOUBLE("Mixer AUX Volume", AC97_AUX, 8, 0, 31, 1), SOC_DOUBLE("All Analog PCM Volume", AC97_PCM, 8, 0, 31, 1), SOC_DOUBLE("Record Gain", AC97_REC_GAIN, 8, 0, 31, 1), SOC_SINGLE("Record Gain Switch", AC97_REC_GAIN, 15, 1, 1), SOC_SINGLE("3D Effect Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0), SOC_ENUM("3D Separation", stac9766_enum[6]), }; /* Mixer */ static const struct snd_kcontrol_new stac9766_main_mixer_controls[] = { SOC_DAPM_SINGLE("PC Beep Switch", AC97_PC_BEEP, 15, 1, 1), SOC_DAPM_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1), SOC_DAPM_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), SOC_DAPM_SINGLE("Line Switch", AC97_LINE, 15, 1, 1), SOC_DAPM_SINGLE("CD Switch", AC97_CD, 15, 1, 1), SOC_DAPM_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1), SOC_DAPM_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1), }; /* Record All Mixer */ static const struct snd_kcontrol_new stac9766_record_all_mixer_controls[] = { SOC_DAPM_SINGLE("PCM Switch", AC97_PCM, 15, 1, 1), SOC_DAPM_SINGLE("Mixer", 0, 0, 0, 0), }; /* Record Mux 0 */ static const struct snd_kcontrol_new stac9766_record_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[0]); /* Mono Mux 1 */ static const struct snd_kcontrol_new stac9766_mono_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[1]); /* Mic1/2 Mux 2 */ static const struct snd_kcontrol_new stac9766_mic_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[2]); /* SPDIF Mux 3 */ static const struct snd_kcontrol_new stac9766_spdif_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[3]); /* Pop Bypass 4 */ static const struct snd_kcontrol_new stac9766_popbypass_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[4]); /* Record All 5 */ static const struct snd_kcontrol_new stac9766_record_all_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[5]); static const struct snd_soc_dapm_widget stac9766_dapm_widgets[] = { SND_SOC_DAPM_INPUT("PCMOut"), SND_SOC_DAPM_INPUT("PCBEEP"), SND_SOC_DAPM_INPUT("Phone"), SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_INPUT("Line"), SND_SOC_DAPM_INPUT("CD"), SND_SOC_DAPM_INPUT("AUX"), SND_SOC_DAPM_INPUT("Video"), SND_SOC_DAPM_DAC("DAC", "Analog Playback", AC97_POWERDOWN, 9, 1), SND_SOC_DAPM_MUX("SPDIF Mux", SND_SOC_NOPM, 0, 0, &stac9766_spdif_mux_controls), SND_SOC_DAPM_MUX("Mic1/2 Mux", SND_SOC_NOPM, 0, 0, &stac9766_mic_mux_controls), SND_SOC_DAPM_MIXER("Mixer", SND_SOC_NOPM, 0, 1, &stac9766_main_mixer_controls[0], ARRAY_SIZE(stac9766_main_mixer_controls)), SND_SOC_DAPM_MUX("Record All Mux", SND_SOC_NOPM, 0, 0, &stac9766_record_all_mux_controls), SND_SOC_DAPM_MUX("Record Mux", SND_SOC_NOPM, 0, 0, &stac9766_record_mux_controls), SND_SOC_DAPM_MUX("Mono Mux", SND_SOC_NOPM, 0, 0, &stac9766_mono_mux_controls), SND_SOC_DAPM_MUX("Pop Bypass Mux", SND_SOC_NOPM, 0, 0, &stac9766_popbypass_mux_controls), SND_SOC_DAPM_MIXER("All Analog", SND_SOC_NOPM, 0, 1, &stac9766_record_all_mixer_controls[0], ARRAY_SIZE(stac9766_record_all_mixer_controls)), SND_SOC_DAPM_ADC("ADC", "Analog Capture", AC97_POWERDOWN, 8, 1), SND_SOC_DAPM_OUTPUT("HP"), SND_SOC_DAPM_OUTPUT("LINE"), SND_SOC_DAPM_OUTPUT("MONO"), SND_SOC_DAPM_OUTPUT("PCMIn"), SND_SOC_DAPM_VMID("VMID"), }; static const struct snd_soc_dapm_route audio_map[] = { {"HP", NULL, "Pop Bypass Mux"}, {"LINE", NULL, "Pop Bypass Mux"}, /* Mono Mux */ {"MONO", NULL, "Pop Bypass Mux"}, {"MONO", NULL, "Mic1/2 Mux"}, /* Pop Bypass Mux */ {"Pop Bypass Mux", NULL, "DAC"}, {"Pop Bypass Mux", NULL, "All Analog Mixer"}, /* Record Mux */ {"ADC", NULL, "Mic1/2 Mux"}, {"ADC", NULL, "CD"}, {"ADC", NULL, "Video"}, {"ADC", NULL, "AUX"}, {"ADC", NULL, "Line"}, {"ADC", NULL, "Record All Mux"}, {"ADC", NULL, "Phone"}, {"PCMIn", NULL, "ADC"}, /* Record All Mux */ {"Record All Mux", NULL, "Mixer"}, {"Record All Mux", NULL, "All Analog Mixer"}, /* All Analog Mixer */ {"All Analog", NULL, "Mixer"}, {"All Analog", "PCM Switch", "DAC"}, /* Mixer */ {"Mixer", "PC Beep Switch", "PCBEEP"}, {"Mixer", "Phone Switch", "Phone"}, {"Mixer", "Mic Switch", "Mic1/2 Mux"}, {"Mixer", "Line Switch", "Line"}, {"Mixer", "CD Switch", "CD"}, {"Mixer", "AUX Switch", "AUX"}, {"Mixer", "Video Switch", "Video"}, {"DAC", NULL, "PCMOut"}, /* Mic1/2 Mux */ {"Mic1/2 Mux", NULL, "MIC1"}, {"Mic1/2 Mux", NULL, "MIC2"}, /* SPDIF Mux */ {"SPDIF Mux", NULL, "PCMOut"}, {"SPDIF Mux", NULL, "ADC"}, {NULL, NULL, NULL}, }; static int stac9766_add_widgets(struct snd_soc_codec *codec) { snd_soc_dapm_new_controls(codec, stac9766_dapm_widgets, ARRAY_SIZE(stac9766_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); return 0; } unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) { u16 val = 0, *cache = codec->reg_cache; if (reg / 2 > ARRAY_SIZE(stac9766_reg)) return -EIO; if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2) { val = soc_ac97_ops.read(codec->ac97, reg); return val; } return cache[reg / 2]; } int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { u16 *cache = codec->reg_cache; if (reg / 2 > ARRAY_SIZE(stac9766_reg)) return -EIO; soc_ac97_ops.write(codec->ac97, reg, val); cache[reg / 2] = val; return 0; } static int ac97_analog_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; unsigned short reg, vra; vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); //vra |= 0x4; stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); printk("AC97_EXTENDED_STATUS %x\n", vra); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = AC97_PCM_FRONT_DAC_RATE; else reg = AC97_PCM_LR_ADC_RATE; return stac9766_ac97_write(codec, reg, runtime->rate); } static int ac97_digital_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { printk("stac9766: ac97_digital_prepare\n"); return 0; } static int stac9766_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { switch (level) { case SND_SOC_BIAS_ON: /* full On */ stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); break; case SND_SOC_BIAS_PREPARE: /* partial On */ stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); break; case SND_SOC_BIAS_STANDBY: /* Off, with power */ stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); break; case SND_SOC_BIAS_OFF: /* Off, without power */ /* disable everything including AC link */ stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); break; } codec->bias_level = level; return 0; } static int stac9766_codec_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static int stac9766_codec_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; u16 id; /* give the codec an AC97 warm reset to start the link */ codec->ac97->bus->ops->warm_reset(codec->ac97); id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); if (id != 0x4c13) { printk(KERN_ERR "stac9766 failed to resume"); return -EIO; } stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (codec->suspend_bias_level == SND_SOC_BIAS_ON) stac9766_set_bias_level(codec, SND_SOC_BIAS_ON); return 0; } static struct snd_soc_dai_ops stac9766_dai_ops_analog = { .prepare = ac97_analog_prepare, }; static struct snd_soc_dai_ops stac9766_dai_ops_digital = { .prepare = ac97_digital_prepare, }; struct snd_soc_dai stac9766_dai[] = { { .name = "stac9766 analog", .id = 0, .ac97_control = 1, /* stream cababilities */ .playback = { .stream_name = "stac9766 analog", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S32_BE, }, .capture = { .stream_name = "stac9766 analog", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S32_BE, }, /* alsa ops */ .ops = &stac9766_dai_ops_analog, }, { .name = "stac9766 digital", .id = 1, .ac97_control = 1, /* stream cababilities */ .playback = { .stream_name = "stac9766 digital", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME, }, /* alsa ops */ .ops = &stac9766_dai_ops_digital, }}; EXPORT_SYMBOL_GPL(stac9766_dai); int stac9766_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) return 1; } soc_ac97_ops.reset(codec->ac97); if (soc_ac97_ops.warm_reset) soc_ac97_ops.warm_reset(codec->ac97); if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) return -EIO; return 0; } static int stac9766_codec_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec; int ret = 0; printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (socdev->card->codec == NULL) return -ENOMEM; codec = socdev->card->codec; mutex_init(&codec->mutex); codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); if (codec->reg_cache == NULL) { ret = -ENOMEM; goto cache_err; } codec->reg_cache_size = sizeof(stac9766_reg); codec->reg_cache_step = 2; codec->name = "STAC9766"; codec->owner = THIS_MODULE; codec->dai = stac9766_dai; codec->num_dai = ARRAY_SIZE(stac9766_dai); codec->write = stac9766_ac97_write; codec->read = stac9766_ac97_read; codec->set_bias_level = stac9766_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) goto codec_err; /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) goto pcm_err; /* do a cold reset for the controller and then try * a warm reset followed by an optional cold reset for codec */ stac9766_reset(codec, 0); ret = stac9766_reset(codec, 1); if (ret < 0) { printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n"); goto reset_err; } stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE(stac9766_snd_ac97_controls)); stac9766_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) goto reset_err; return 0; reset_err: snd_soc_free_pcms(socdev); pcm_err: snd_soc_free_ac97_codec(codec); codec_err: kfree(codec->private_data); cache_err: kfree(socdev->card->codec); socdev->card->codec = NULL; return ret; } static int stac9766_codec_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; if (codec == NULL) return 0; snd_soc_dapm_free(socdev); snd_soc_free_pcms(socdev); snd_soc_free_ac97_codec(codec); kfree(codec->reg_cache); kfree(codec); return 0; } struct snd_soc_codec_device soc_codec_dev_stac9766 = { .probe = stac9766_codec_probe, .remove = stac9766_codec_remove, .suspend = stac9766_codec_suspend, .resume = stac9766_codec_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); static int __init stac9766_probe(struct platform_device *pdev) { #if defined(CONFIG_SND_SOC_OF_SIMPLE) /* Tell the of_soc helper about this codec */ of_snd_soc_register_codec(&soc_codec_dev_stac9766, pdev->dev.archdata.of_node, stac9766_dai, ARRAY_SIZE(stac9766_dai), pdev->dev.archdata.of_node); #endif return 0; } static struct platform_driver stac9766_driver = { .probe = stac9766_probe, .driver = { .name = "stac9766", }, }; static __init int stac9766_driver_init(void) { snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); return platform_driver_register(&stac9766_driver); } static __exit void stac9766_driver_exit(void) { } module_init(stac9766_driver_init); module_exit(stac9766_driver_exit); MODULE_DESCRIPTION("ASoC stac9766 driver"); MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); MODULE_LICENSE("GPL"); -- Jon Smirl jonsmirl(a)gmail.com

2 8

Re: [alsa-devel] errors in wm8990 codec driver
by Park Jinyoung 01 May '09

01 May '09

Hello, I made wm8990 patch file on your request. The patch file was attached in this. I use my company email, because I have no other method for file attach in my company. Thanks. 2009/4/30 Mark Brown <broonie(a)sirena.org.uk>: > On Thu, Apr 30, 2009 at 02:03:03PM +0900, Jinyoung Park wrote: >> I found a number of typing errors in wm8990 codec driver. >> The mis-typing exist in dapm controller definitions and dapm route definitions, >> so happen mis-matched error when snd_soc_dapm_add_routes(). >> Belows are my modified code. Please check... > > This all looks reasonable - could you please supply a Signed-off-by for > your patch? It may also help to use git send-email to send your patch > or attach it, gmail tends to mangle patches so that they can't be > applied and it looks like some of that may have happened here, though > I've not actually tried applying it. >

2 1

[alsa-devel] [PATCH] ASoC: TWL4030: Disable DACs in analog loopback
by Lopez Cruz, Misael 30 Apr '09

30 Apr '09

In analog loopback mode, DACs should not be enabled. For that reason, DAC widgets (DAC Right1, DAC Left1, DAC Right2, DAC Left2) will power DACs only during playback. Analog loopback requires to set a master enable bit when any of the analog bypass switches are enabled. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 34 +++++++++++++++++++++------------- 1 files changed, 21 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index efa1a80..d3184a0 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -64,7 +64,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_VRXPGA (0x14) */ 0x00, /* REG_VSTPGA (0x15) */ 0x00, /* REG_VRX2ARXPGA (0x16) */ - 0x0c, /* REG_AVDAC_CTL (0x17) */ + 0x00, /* REG_AVDAC_CTL (0x17) */ 0x00, /* REG_ARX2VTXPGA (0x18) */ 0x00, /* REG_ARXL1_APGA_CTL (0x19) */ 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */ @@ -585,7 +585,7 @@ static int bypass_event(struct snd_soc_dapm_widget *w, struct soc_mixer_control *m = (struct soc_mixer_control *)w->kcontrols->private_value; struct twl4030_priv *twl4030 = w->codec->private_data; - unsigned char reg; + unsigned char reg, misc; reg = twl4030_read_reg_cache(w->codec, m->reg); @@ -605,6 +605,14 @@ static int bypass_event(struct snd_soc_dapm_widget *w, twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4)); } + /* Enable master analog loopback mode if any analog switch is enabled*/ + misc = twl4030_read_reg_cache(w->codec, TWL4030_REG_MISC_SET_1); + if (twl4030->bypass_state & 0xF) + misc |= TWL4030_FMLOOP_EN; + else + misc &= ~TWL4030_FMLOOP_EN; + twl4030_write(w->codec, TWL4030_REG_MISC_SET_1, misc); + if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) { if (twl4030->bypass_state) twl4030_codec_mute(w->codec, 0); @@ -927,13 +935,13 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* DACs */ SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback", - SND_SOC_NOPM, 0, 0), + TWL4030_REG_AVDAC_CTL, 0, 0), SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback", - SND_SOC_NOPM, 0, 0), + TWL4030_REG_AVDAC_CTL, 1, 0), SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback", - SND_SOC_NOPM, 0, 0), + TWL4030_REG_AVDAC_CTL, 2, 0), SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback", - SND_SOC_NOPM, 0, 0), + TWL4030_REG_AVDAC_CTL, 3, 0), SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", TWL4030_REG_AVDAC_CTL, 4, 0), @@ -971,14 +979,14 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { &twl4030_dapm_dbypassr_control, bypass_event, SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL, + SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", TWL4030_REG_AVDAC_CTL, - 1, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", TWL4030_REG_AVDAC_CTL, - 2, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, - 3, 0, NULL, 0), /* Output MIXER controls */ /* Earpiece */ -- 1.5.4.3

2 5

[alsa-devel] errors in wm8990 codec driver
by Jinyoung Park 30 Apr '09

30 Apr '09

Hello, I found a number of typing errors in wm8990 codec driver. The mis-typing exist in dapm controller definitions and dapm route definitions, so happen mis-matched error when snd_soc_dapm_add_routes(). Belows are my modified code. Please check... Thanks. diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index cedf3f9..ddc328e 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -729,7 +729,7 @@ SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_P inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* AINLMUX */ -SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0, +SND_SOC_DAPM_MUX_E("AINLMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0, &wm8990_dapm_ainlmux_controls, inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), @@ -740,7 +740,7 @@ SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_P inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* AINRMUX */ -SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0, +SND_SOC_DAPM_MUX_E("AINRMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0, &wm8990_dapm_ainrmux_controls, inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), @@ -848,40 +848,40 @@ static const struct snd_soc_dapm_route audio_map[] = { {"LIN12 PGA", "LIN2 Switch", "LIN2"}, /* LIN34 PGA */ {"LIN34 PGA", "LIN3 Switch", "LIN3"}, - {"LIN34 PGA", "LIN4 Switch", "LIN4"}, + {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"}, /* INMIXL */ {"INMIXL", "Record Left Volume", "LOMIX"}, {"INMIXL", "LIN2 Volume", "LIN2"}, {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, - /* AILNMUX */ - {"AILNMUX", "INMIXL Mix", "INMIXL"}, - {"AILNMUX", "DIFFINL Mix", "LIN12PGA"}, - {"AILNMUX", "DIFFINL Mix", "LIN34PGA"}, - {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"}, - {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* AINLMUX */ + {"AINLMUX", "INMIXL Mix", "INMIXL"}, + {"AINLMUX", "DIFFINL Mix", "LIN12 PGA"}, + {"AINLMUX", "DIFFINL Mix", "LIN34 PGA"}, + {"AINLMUX", "RXVOICE Mix", "LIN4/RXN"}, + {"AINLMUX", "RXVOICE Mix", "RIN4/RXP"}, /* ADC */ - {"Left ADC", NULL, "AILNMUX"}, + {"Left ADC", NULL, "AINLMUX"}, /* RIN12 PGA */ {"RIN12 PGA", "RIN1 Switch", "RIN1"}, {"RIN12 PGA", "RIN2 Switch", "RIN2"}, /* RIN34 PGA */ {"RIN34 PGA", "RIN3 Switch", "RIN3"}, - {"RIN34 PGA", "RIN4 Switch", "RIN4"}, + {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"}, /* INMIXL */ {"INMIXR", "Record Right Volume", "ROMIX"}, {"INMIXR", "RIN2 Volume", "RIN2"}, {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, - /* AIRNMUX */ - {"AIRNMUX", "INMIXR Mix", "INMIXR"}, - {"AIRNMUX", "DIFFINR Mix", "RIN12PGA"}, - {"AIRNMUX", "DIFFINR Mix", "RIN34PGA"}, - {"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"}, - {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* AINRMUX */ + {"AINRMUX", "INMIXR Mix", "INMIXR"}, + {"AINRMUX", "DIFFINR Mix", "RIN12 PGA"}, + {"AINRMUX", "DIFFINR Mix", "RIN34 PGA"}, + {"AINRMUX", "RXVOICE Mix", "LIN4/RXN"}, + {"AINRMUX", "RXVOICE Mix", "RIN4/RXP"}, /* ADC */ - {"Right ADC", NULL, "AIRNMUX"}, + {"Right ADC", NULL, "AINRMUX"}, /* LOMIX */ {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"}, @@ -922,7 +922,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"}, /* OUT3MIX */ - {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"}, + {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"}, {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"}, /* OUT4MIX */ @@ -949,7 +949,7 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Output Pins */ {"LON", NULL, "LONMIX"}, {"LOP", NULL, "LOPMIX"}, - {"OUT", NULL, "OUT3MIX"}, + {"OUT3", NULL, "OUT3MIX"}, {"LOUT", NULL, "LOUT PGA"}, {"SPKN", NULL, "SPKMIX"}, {"ROUT", NULL, "ROUT PGA"},

2 1

[alsa-devel] [RFC PATCH 3/5] pxa: clean up the legacy SSP API
by Eric Miao 30 Apr '09

30 Apr '09

Actually, due to the number of possible SSP configurations, it's quite a difficult job to maintain an API that can be well re-used. The guideline here is to keep only a list of SSP devices and their basic information, and it's up to the user driver (like SPI or Audio SSP) to configure the SSP in a desired way. Now that the legacy 'struct ssp_dev' based APIs are removed, and only the functions below are kept, modified a little bit to make them aware of the new 'struct ssp_device': ssp_enable() ssp_disable() ssp_save_state() ssp_restore_state() And finally, ssp_{save,restore}_state() can be merged into the suspend() and resume() interface of the SSP's platform_driver, thus saving the user driver from doing some similar things in common. Signed-off-by: Eric Miao <eric.miao(a)marvell.com> --- arch/arm/mach-pxa/include/mach/ssp.h | 32 +---- arch/arm/mach-pxa/ssp.c | 240 +--------------------------------- sound/soc/pxa/pxa-ssp.c | 65 +++++----- 3 files changed, 45 insertions(+), 292 deletions(-) diff --git a/arch/arm/mach-pxa/include/mach/ssp.h b/arch/arm/mach-pxa/include/mach/ssp.h index cb5cb76..e431685 100644 --- a/arch/arm/mach-pxa/include/mach/ssp.h +++ b/arch/arm/mach-pxa/include/mach/ssp.h @@ -58,26 +58,10 @@ struct ssp_state { u32 psp; }; -struct ssp_dev { - struct ssp_device *ssp; - u32 port; - u32 mode; - u32 flags; - u32 psp_flags; - u32 speed; - int irq; -}; - -int ssp_write_word(struct ssp_dev *dev, u32 data); -int ssp_read_word(struct ssp_dev *dev, u32 *data); -int ssp_flush(struct ssp_dev *dev); -void ssp_enable(struct ssp_dev *dev); -void ssp_disable(struct ssp_dev *dev); -void ssp_save_state(struct ssp_dev *dev, struct ssp_state *ssp); -void ssp_restore_state(struct ssp_dev *dev, struct ssp_state *ssp); -int ssp_init(struct ssp_dev *dev, u32 port, u32 init_flags); -int ssp_config(struct ssp_dev *dev, u32 mode, u32 flags, u32 psp_flags, u32 speed); -void ssp_exit(struct ssp_dev *dev); +void ssp_enable(struct ssp_device *ssp); +void ssp_disable(struct ssp_device *ssp); +void ssp_save_state(struct ssp_device *ssp, struct ssp_state *); +void ssp_restore_state(struct ssp_device *ssp, struct ssp_state *); /** * ssp_write_reg - Write to a SSP register @@ -86,9 +70,9 @@ void ssp_exit(struct ssp_dev *dev); * @reg: Register to write to * @val: Value to be written. */ -static inline void ssp_write_reg(struct ssp_device *dev, u32 reg, u32 val) +static inline void ssp_write_reg(struct ssp_device *ssp, u32 reg, u32 val) { - __raw_writel(val, dev->mmio_base + reg); + __raw_writel(val, ssp->mmio_base + reg); } /** @@ -97,9 +81,9 @@ static inline void ssp_write_reg(struct ssp_device *dev, u32 reg, u32 val) * @dev: SSP device to access * @reg: Register to read from */ -static inline u32 ssp_read_reg(struct ssp_device *dev, u32 reg) +static inline u32 ssp_read_reg(struct ssp_device *ssp, u32 reg) { - return __raw_readl(dev->mmio_base + reg); + return __raw_readl(ssp->mmio_base + reg); } struct ssp_device *ssp_request(int port, const char *label); diff --git a/arch/arm/mach-pxa/ssp.c b/arch/arm/mach-pxa/ssp.c index 965e38c..f48807a 100644 --- a/arch/arm/mach-pxa/ssp.c +++ b/arch/arm/mach-pxa/ssp.c @@ -35,153 +35,25 @@ #include <mach/ssp.h> #include <mach/regs-ssp.h> -#define TIMEOUT 100000 - -static irqreturn_t ssp_interrupt(int irq, void *dev_id) +void ssp_enable(struct ssp_device *ssp) { - struct ssp_dev *dev = dev_id; - struct ssp_device *ssp = dev->ssp; - unsigned int status; - - status = __raw_readl(ssp->mmio_base + SSSR); - __raw_writel(status, ssp->mmio_base + SSSR); - - if (status & SSSR_ROR) - printk(KERN_WARNING "SSP(%d): receiver overrun\n", dev->port); - - if (status & SSSR_TUR) - printk(KERN_WARNING "SSP(%d): transmitter underrun\n", dev->port); - - if (status & SSSR_BCE) - printk(KERN_WARNING "SSP(%d): bit count error\n", dev->port); - - return IRQ_HANDLED; -} - -/** - * ssp_write_word - write a word to the SSP port - * @data: 32-bit, MSB justified data to write. - * - * Wait for a free entry in the SSP transmit FIFO, and write a data - * word to the SSP port. - * - * The caller is expected to perform the necessary locking. - * - * Returns: - * %-ETIMEDOUT timeout occurred - * 0 success - */ -int ssp_write_word(struct ssp_dev *dev, u32 data) -{ - struct ssp_device *ssp = dev->ssp; - int timeout = TIMEOUT; - - while (!(__raw_readl(ssp->mmio_base + SSSR) & SSSR_TNF)) { - if (!--timeout) - return -ETIMEDOUT; - cpu_relax(); - } - - __raw_writel(data, ssp->mmio_base + SSDR); - - return 0; -} - -/** - * ssp_read_word - read a word from the SSP port - * - * Wait for a data word in the SSP receive FIFO, and return the - * received data. Data is LSB justified. - * - * Note: Currently, if data is not expected to be received, this - * function will wait for ever. - * - * The caller is expected to perform the necessary locking. - * - * Returns: - * %-ETIMEDOUT timeout occurred - * 32-bit data success - */ -int ssp_read_word(struct ssp_dev *dev, u32 *data) -{ - struct ssp_device *ssp = dev->ssp; - int timeout = TIMEOUT; - - while (!(__raw_readl(ssp->mmio_base + SSSR) & SSSR_RNE)) { - if (!--timeout) - return -ETIMEDOUT; - cpu_relax(); - } - - *data = __raw_readl(ssp->mmio_base + SSDR); - return 0; -} - -/** - * ssp_flush - flush the transmit and receive FIFOs - * - * Wait for the SSP to idle, and ensure that the receive FIFO - * is empty. - * - * The caller is expected to perform the necessary locking. - */ -int ssp_flush(struct ssp_dev *dev) -{ - struct ssp_device *ssp = dev->ssp; - int timeout = TIMEOUT * 2; - - /* ensure TX FIFO is empty instead of not full */ - if (cpu_is_pxa3xx()) { - while (__raw_readl(ssp->mmio_base + SSSR) & 0xf00) { - if (!--timeout) - return -ETIMEDOUT; - cpu_relax(); - } - timeout = TIMEOUT * 2; - } - - do { - while (__raw_readl(ssp->mmio_base + SSSR) & SSSR_RNE) { - if (!--timeout) - return -ETIMEDOUT; - (void)__raw_readl(ssp->mmio_base + SSDR); - } - if (!--timeout) - return -ETIMEDOUT; - } while (__raw_readl(ssp->mmio_base + SSSR) & SSSR_BSY); - - return 0; -} - -/** - * ssp_enable - enable the SSP port - * - * Turn on the SSP port. - */ -void ssp_enable(struct ssp_dev *dev) -{ - struct ssp_device *ssp = dev->ssp; uint32_t sscr0; sscr0 = __raw_readl(ssp->mmio_base + SSCR0); sscr0 |= SSCR0_SSE; __raw_writel(sscr0, ssp->mmio_base + SSCR0); } +EXPORT_SYMBOL(ssp_enable); -/** - * ssp_disable - shut down the SSP port - * - * Turn off the SSP port, optionally powering it down. - */ -void ssp_disable(struct ssp_dev *dev) +void ssp_disable(struct ssp_device *ssp) { - struct ssp_device *ssp = dev->ssp; uint32_t sscr0; sscr0 = __raw_readl(ssp->mmio_base + SSCR0); sscr0 &= ~SSCR0_SSE; __raw_writel(sscr0, ssp->mmio_base + SSCR0); } +EXPORT_SYMBOL(ssp_disable); /** * ssp_save_state - save the SSP configuration @@ -189,16 +61,12 @@ void ssp_disable(struct ssp_dev *dev) * * Save the configured SSP state for suspend. */ -void ssp_save_state(struct ssp_dev *dev, struct ssp_state *state) +void ssp_save_state(struct ssp_device *ssp, struct ssp_state *state) { - struct ssp_device *ssp = dev->ssp; - state->cr0 = __raw_readl(ssp->mmio_base + SSCR0); state->cr1 = __raw_readl(ssp->mmio_base + SSCR1); state->to = __raw_readl(ssp->mmio_base + SSTO); state->psp = __raw_readl(ssp->mmio_base + SSPSP); - - ssp_disable(dev); } /** @@ -207,9 +75,8 @@ void ssp_save_state(struct ssp_dev *dev, struct ssp_state *state) * * Restore the SSP configuration saved previously by ssp_save_state. */ -void ssp_restore_state(struct ssp_dev *dev, struct ssp_state *state) +void ssp_restore_state(struct ssp_device *ssp, struct ssp_state *state) { - struct ssp_device *ssp = dev->ssp; uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE; __raw_writel(sssr, ssp->mmio_base + SSSR); @@ -221,89 +88,6 @@ void ssp_restore_state(struct ssp_dev *dev, struct ssp_state *state) __raw_writel(state->cr0, ssp->mmio_base + SSCR0); } -/** - * ssp_config - configure SSP port settings - * @mode: port operating mode - * @flags: port config flags - * @psp_flags: port PSP config flags - * @speed: port speed - * - * Port MUST be disabled by ssp_disable before making any config changes. - */ -int ssp_config(struct ssp_dev *dev, u32 mode, u32 flags, u32 psp_flags, u32 speed) -{ - struct ssp_device *ssp = dev->ssp; - - dev->mode = mode; - dev->flags = flags; - dev->psp_flags = psp_flags; - dev->speed = speed; - - /* set up port type, speed, port settings */ - __raw_writel((dev->speed | dev->mode), ssp->mmio_base + SSCR0); - __raw_writel(dev->flags, ssp->mmio_base + SSCR1); - __raw_writel(dev->psp_flags, ssp->mmio_base + SSPSP); - - return 0; -} - -/** - * ssp_init - setup the SSP port - * - * initialise and claim resources for the SSP port. - * - * Returns: - * %-ENODEV if the SSP port is unavailable - * %-EBUSY if the resources are already in use - * %0 on success - */ -int ssp_init(struct ssp_dev *dev, u32 port, u32 init_flags) -{ - struct ssp_device *ssp; - int ret; - - ssp = ssp_request(port, "SSP"); - if (ssp == NULL) - return -ENODEV; - - dev->ssp = ssp; - dev->port = port; - - /* do we need to get irq */ - if (!(init_flags & SSP_NO_IRQ)) { - ret = request_irq(ssp->irq, ssp_interrupt, - 0, "SSP", dev); - if (ret) - goto out_region; - dev->irq = ssp->irq; - } else - dev->irq = NO_IRQ; - - /* turn on SSP port clock */ - clk_enable(ssp->clk); - return 0; - -out_region: - ssp_free(ssp); - return ret; -} - -/** - * ssp_exit - undo the effects of ssp_init - * - * release and free resources for the SSP port. - */ -void ssp_exit(struct ssp_dev *dev) -{ - struct ssp_device *ssp = dev->ssp; - - ssp_disable(dev); - if (dev->irq != NO_IRQ) - free_irq(dev->irq, dev); - clk_disable(ssp->clk); - ssp_free(ssp); -} - static DEFINE_MUTEX(ssp_lock); static LIST_HEAD(ssp_list); @@ -531,18 +315,6 @@ static void __exit pxa_ssp_exit(void) arch_initcall(pxa_ssp_init); module_exit(pxa_ssp_exit); -EXPORT_SYMBOL(ssp_write_word); -EXPORT_SYMBOL(ssp_read_word); -EXPORT_SYMBOL(ssp_flush); -EXPORT_SYMBOL(ssp_enable); -EXPORT_SYMBOL(ssp_disable); -EXPORT_SYMBOL(ssp_save_state); -EXPORT_SYMBOL(ssp_restore_state); -EXPORT_SYMBOL(ssp_init); -EXPORT_SYMBOL(ssp_exit); -EXPORT_SYMBOL(ssp_config); - MODULE_DESCRIPTION("PXA SSP driver"); MODULE_AUTHOR("Liam Girdwood"); MODULE_LICENSE("GPL"); - diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 308a657..a65993c 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -42,7 +42,7 @@ * SSP audio private data */ struct ssp_priv { - struct ssp_dev dev; + struct ssp_device *ssp; unsigned int sysclk; int dai_fmt; #ifdef CONFIG_PM @@ -222,10 +222,8 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, int ret = 0; if (!cpu_dai->active) { - priv->dev.port = cpu_dai->id + 1; - priv->dev.irq = NO_IRQ; - clk_enable(priv->dev.ssp->clk); - ssp_disable(&priv->dev); + clk_enable(priv->ssp->clk); + ssp_disable(priv->ssp); } return ret; } @@ -238,8 +236,8 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, struct ssp_priv *priv = cpu_dai->private_data; if (!cpu_dai->active) { - ssp_disable(&priv->dev); - clk_disable(priv->dev.ssp->clk); + ssp_disable(priv->ssp); + clk_disable(priv->ssp->clk); } } @@ -252,8 +250,9 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) if (!cpu_dai->active) return 0; - ssp_save_state(&priv->dev, &priv->state); - clk_disable(priv->dev.ssp->clk); + ssp_save_state(priv->ssp, &priv->state); + ssp_disable(priv->ssp); + clk_disable(priv->ssp->clk); return 0; } @@ -264,10 +263,9 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) if (!cpu_dai->active) return 0; - clk_enable(priv->dev.ssp->clk); - ssp_restore_state(&priv->dev, &priv->state); - ssp_enable(&priv->dev); - + clk_enable(priv->ssp->clk); + ssp_restore_state(priv->ssp, &priv->state); + ssp_enable(priv->ssp); return 0; } @@ -280,10 +278,9 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) * ssp_set_clkdiv - set SSP clock divider * @div: serial clock rate divider */ -static void ssp_set_scr(struct ssp_dev *dev, u32 div) +static void ssp_set_scr(struct ssp_device *ssp, u32 div) { - struct ssp_device *ssp = dev->ssp; - u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR; + u32 sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SCR; ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div))); } @@ -295,7 +292,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; int val; u32 sscr0 = ssp_read_reg(ssp, SSCR0) & @@ -326,7 +323,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, break; case PXA_SSP_CLK_AUDIO: priv->sysclk = 0; - ssp_set_scr(&priv->dev, 1); + ssp_set_scr(priv->ssp, 1); sscr0 |= SSCR0_ACS; break; default: @@ -336,11 +333,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, /* The SSP clock must be disabled when changing SSP clock mode * on PXA2xx. On PXA3xx it must be enabled when doing so. */ if (!cpu_is_pxa3xx()) - clk_disable(priv->dev.ssp->clk); + clk_disable(priv->ssp->clk); val = ssp_read_reg(ssp, SSCR0) | sscr0; ssp_write_reg(ssp, SSCR0, val); if (!cpu_is_pxa3xx()) - clk_enable(priv->dev.ssp->clk); + clk_enable(priv->ssp->clk); return 0; } @@ -352,7 +349,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; int val; switch (div_id) { @@ -387,7 +384,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, ssp_write_reg(ssp, SSACD, val); break; case PXA_SSP_DIV_SCR: - ssp_set_scr(&priv->dev, div); + ssp_set_scr(ssp, div); break; default: return -ENODEV; @@ -403,7 +400,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70; #if defined(CONFIG_PXA3xx) @@ -472,7 +469,7 @@ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, unsigned int mask, int slots) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; u32 sscr0; sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); @@ -494,7 +491,7 @@ static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, int tristate) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; u32 sscr1; sscr1 = ssp_read_reg(ssp, SSCR1); @@ -516,7 +513,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; u32 sscr0; u32 sscr1; u32 sspsp; @@ -622,7 +619,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; int dma = 0, chn = params_channels(params); u32 sscr0; u32 sspsp; @@ -735,12 +732,12 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret = 0; struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; int val; switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - ssp_enable(&priv->dev); + ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: val = ssp_read_reg(ssp, SSCR1); @@ -759,7 +756,7 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, else val |= SSCR1_RSRE; ssp_write_reg(ssp, SSCR1, val); - ssp_enable(&priv->dev); + ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_STOP: val = ssp_read_reg(ssp, SSCR1); @@ -770,7 +767,7 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, ssp_write_reg(ssp, SSCR1, val); break; case SNDRV_PCM_TRIGGER_SUSPEND: - ssp_disable(&priv->dev); + ssp_disable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: val = ssp_read_reg(ssp, SSCR1); @@ -800,8 +797,8 @@ static int pxa_ssp_probe(struct platform_device *pdev, if (!priv) return -ENOMEM; - priv->dev.ssp = ssp_request(dai->id + 1, "SoC audio"); - if (priv->dev.ssp == NULL) { + priv->ssp = ssp_request(dai->id + 1, "SoC audio"); + if (priv->ssp == NULL) { ret = -ENODEV; goto err_priv; } @@ -819,7 +816,7 @@ static void pxa_ssp_remove(struct platform_device *pdev, struct snd_soc_dai *dai) { struct ssp_priv *priv = dai->private_data; - ssp_free(priv->dev.ssp); + ssp_free(priv->ssp); } #define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ -- 1.6.0.4

4 11

[alsa-devel] 6-channel output and i2s
by Jon Smirl 29 Apr '09

29 Apr '09

My hardware was has three stereo i2s lines. These three lines are all tied to the same external clock so they are synchronized. Because there are three i2s lines, each with it's own DMA queue and FIFO, I need three independent output buffers. What is the correct way to set this up in ALSA? -- Jon Smirl jonsmirl(a)gmail.com

2 4

[alsa-devel] Detect when sound card is in use
by Claus Ilginnis 29 Apr '09

29 Apr '09

Hi everyone, I have a problem and I dont't know where to ask my questions anymore, so I hope this mailing list can help me make progress with my problem.... I have a speaker set connected to an USB-Power-Plug (by which I can turn the speakers on and off ) Now I want to detect, when the soundcard is in use and when not. (to turn the speakers on when needed...) (I am a C++ programmer so I can mess some things up) So my question is: Which way is best to detect wheather my sound card is in use or not ? Is there a general overview/documentation of the sound kernel modules which describes the general relationship between them ? What would you do ? PLEASE help ! - Mit freundlichen Grüßen / Best regards Claus Ilginnis

3 3

[alsa-devel] Getting the correct capture mixer
by Ludovico Cavedon 29 Apr '09

29 Apr '09

Hi, I am working on fixing some audio related bugs of a VoIP application (qutecom) which, like most VoIP applications, allows the user to change the playback and recording volume. For playback volume I am opening the "Master" mixer element. However I am not able to figure out how the application should get the correct mixer for playback. Is there any rule? For example, my sound card has two capture volumes, one "Int Mic" and one "Ext Mic". I guess I should choose on the basis of what is the user-selected mic. In fact there are also two mutually exclusive switches "IntMic" and "ExtMic". But how should the application understand that? Here part of the output of amixer, my sound card is "Intel [HDA Intel], device 0: CONEXANT Analog [CONEXANT Analog]" ---------------------------- Simple mixer control 'Ext Mic',0 Capabilities: pvolume cvolume pswitch cswitch Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 20 Capture 0 - 23 Front Left: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Front Right: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Simple mixer control 'ExtMic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [off] Simple mixer control 'Int Mic',0 Capabilities: pvolume cvolume pswitch cswitch Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 20 Capture 0 - 23 Front Left: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Front Right: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Simple mixer control 'IntMic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [on] ------------------------ As an aside, it looks like that the capture volume has no effect, as well all as the cswitches and pswitches of "Int Mic" and "Ext Mic". Thanks, Ludovico

1 0

[alsa-devel] No sound on Acer Aspire 6935G
by Karthik Ramgopal 29 Apr '09

29 Apr '09

Hi, I am using Ubuntu 9.04 Jaunty Jackalope with the latest alsa-driver snapshot provided by Takashi. The sound system does not work without a hda-verb command manually executed every time the system starts up. The command I execute is: hda-verb /dev/snd/hwC0D0 0x15 SET_EAPD_BTLENABLE 2 If I do this I get audio out. But my internal mic does not work and nor does jack sensing. When I insert a device into the headphone jack the speakers do not get muted and there is also no sound from the headphones themselves. Even an external mic/line-in device does not work. I have attached the output of alsa-info.sh here.. Please go through it and try to solve the problem. Thanks and Regards, Karthik

3 17

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 29 Apr '09

29 Apr '09

The following changes since commit 8a1f936acdfd53cb0a981f3f80483863dcd84fa9: Peter Ujfalusi (1): ASoC: TWL4030: Add 4 channel TDM support are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Daniel Mack (3): ASoC: add SOC_DOUBLE_EXT macro ASoC: cs4270: fix Master Capture Switch polarity ASoC: cs4270: add Master Playback Switch Jon Smirl (3): ASoC: Set the MPC5200 i2s driver to BROKEN status. ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s ASoC: Rename the PSC functions to DMA Jonathan Cameron (1): ASoC WM8940 Driver Joonyoung Shim (1): ASoC: TWL4030: Fix gain control for earpiece amplifier Mark Brown (10): ASoC: Fix S3C64xx IIS device registration and support both ports ASoC: S3C2412: Failing to get the I2S clock is an error ASoC: Enforce symmetric rates for S3C64xx I2S interface ASoC: Include WM8350 register definitions in CODEC header ASoC: s3c-i2s-v2 diagnostic improvements ASoC: Use our registration function for S3C64xx ASoC: Fix logic in WM8350 master clocking check Merge branch 'for-2.6.30' into for-2.6.31 ASoC: Staticise TLV values in WM8940 Merge branch 'for-2.6.30' into for-2.6.31 Peter Ujfalusi (1): ASoC: Beagle: Add support for 4 channel include/sound/soc.h | 8 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4270.c | 44 ++- sound/soc/codecs/twl4030.c | 8 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8350.h | 1 + sound/soc/codecs/wm8940.c | 955 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8940.h | 104 +++++ sound/soc/fsl/Kconfig | 6 +- sound/soc/fsl/Makefile | 2 + sound/soc/fsl/mpc5200_dma.c | 457 +++++++++++++++++++ sound/soc/fsl/mpc5200_dma.h | 81 ++++ sound/soc/fsl/mpc5200_psc_i2s.c | 644 ++++----------------------- sound/soc/omap/omap3beagle.c | 26 +- sound/soc/s3c24xx/s3c2412-i2s.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 148 ++++-- sound/soc/s3c24xx/s3c64xx-i2s.h | 2 +- 18 files changed, 1867 insertions(+), 629 deletions(-) create mode 100644 sound/soc/codecs/wm8940.c create mode 100644 sound/soc/codecs/wm8940.h create mode 100644 sound/soc/fsl/mpc5200_dma.c create mode 100644 sound/soc/fsl/mpc5200_dma.h

4 35

[alsa-devel] Setting the AC97 private data
by Jon Smirl 29 Apr '09

29 Apr '09

My AC97 CPU read/write routines need private data set so that they know which port to use. The mpc5200 has multiple PSCs that can function in AC97 mode. static int psc_dma_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { ... if (rtd->socdev->card->codec->ac97) rtd->socdev->card->codec->ac97->private_data = psc_dma; .. } Is there some easier way to set this data? -- Jon Smirl jonsmirl(a)gmail.com

2 3

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 28 Apr '09

28 Apr '09

The following changes since commit 0c95de73a711d376dc17afe484f919bd5b66c016: Jon Smirl (1): ASoC: Set the MPC5200 i2s driver to BROKEN status. are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Joonyoung Shim (1): ASoC: TWL4030: Fix gain control for earpiece amplifier sound/soc/codecs/twl4030.c | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-)

1 0

[alsa-devel] [PATCH] ALSA: au1x00: convert to platform device
by omelinux.net＠alsa-project.org 28 Apr '09

28 Apr '09

Make sound/mips/au1x00.c a proper platform_driver. Cc: Charles Eidsness <charles(a)cooper-street.com> Signed-off-by: Manuel Lauss <mano(a)roarinelk.homelinux.net> --- Compile-tested only; I don't have the right hardware to test this. Charles: If you still have access to the hardware, could you please test this patch? Just add something like the following to your board's platform device registration: #include <asm/mach-au1x00/au1000_dma.h> static struct resource au1xxx_ac97c_res[] = { [0] = { .start = AC97_PHYS_ADDR, .end = AC97_PHYS_ADDR + 0xfff, .flags = IORESOURCE_MEM, }, [1] = { .start = AU1000_AC97C_INT, .end = AU1000_AC97C_INT, .flags = IORESOURCE_IRQ, }, [2] = { .start = DMA_ID_AC97C_TX, .end = DMA_ID_AC97C_TX, .flags = IORESOURCE_DMA, }, [3] = { .start = DMA_ID_AC97C_RX, .end = DMA_ID_AC97C_RX, .flags = IORESOURCE_DMA, }, }; static struct platform_device au1xxx_ac97c_device = { .name = "au1000-ac97c", .id = -1, .resource = au1xxx_ac97c_res, .num_resources = ARRAY_SIZE(au1xxx_ac97c_res), }; and register the au1xxx_ac97c_device. Thank you! sound/mips/au1x00.c | 244 +++++++++++++++++++++++++++++++-------------------- 1 files changed, 147 insertions(+), 97 deletions(-) diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index 3e763d6..11b5865 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -37,6 +37,7 @@ #include <linux/ioport.h> #include <linux/interrupt.h> #include <linux/init.h> +#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/initval.h> @@ -97,6 +98,7 @@ struct snd_au1000 { struct snd_pcm *pcm; struct audio_stream *stream[2]; /* playback & capture */ + int dmaid[2]; /* tx(0)/rx(1) DMA ids */ }; /*--------------------------- Local Functions --------------------------------*/ @@ -464,13 +466,13 @@ snd_au1000_pcm_new(struct snd_au1000 *au1000) spin_lock_init(&au1000->stream[CAPTURE]->dma_lock); flags = claim_dma_lock(); - if ((au1000->stream[PLAYBACK]->dma = request_au1000_dma(DMA_ID_AC97C_TX, + if ((au1000->stream[PLAYBACK]->dma = request_au1000_dma(au1000->dmaid[0], "AC97 TX", au1000_dma_interrupt, IRQF_DISABLED, au1000->stream[PLAYBACK])) < 0) { release_dma_lock(flags); return -EBUSY; } - if ((au1000->stream[CAPTURE]->dma = request_au1000_dma(DMA_ID_AC97C_RX, + if ((au1000->stream[CAPTURE]->dma = request_au1000_dma(au1000->dmaid[1], "AC97 RX", au1000_dma_interrupt, IRQF_DISABLED, au1000->stream[CAPTURE])) < 0){ release_dma_lock(flags); @@ -550,69 +552,12 @@ get the interrupt driven case to work efficiently */ spin_unlock(&au1000->ac97_lock); } -static int __devinit -snd_au1000_ac97_new(struct snd_au1000 *au1000) -{ - int err; - struct snd_ac97_bus *pbus; - struct snd_ac97_template ac97; - static struct snd_ac97_bus_ops ops = { - .write = snd_au1000_ac97_write, - .read = snd_au1000_ac97_read, - }; - - if ((au1000->ac97_res_port = request_mem_region(CPHYSADDR(AC97C_CONFIG), - 0x100000, "Au1x00 AC97")) == NULL) { - snd_printk(KERN_ERR "ALSA AC97: can't grap AC97 port\n"); - return -EBUSY; - } - au1000->ac97_ioport = (struct au1000_ac97_reg *) - KSEG1ADDR(au1000->ac97_res_port->start); - - spin_lock_init(&au1000->ac97_lock); - - /* configure pins for AC'97 - TODO: move to board_setup.c */ - au_writel(au_readl(SYS_PINFUNC) & ~0x02, SYS_PINFUNC); - - /* Initialise Au1000's AC'97 Control Block */ - au1000->ac97_ioport->cntrl = AC97C_RS | AC97C_CE; - udelay(10); - au1000->ac97_ioport->cntrl = AC97C_CE; - udelay(10); - - /* Initialise External CODEC -- cold reset */ - au1000->ac97_ioport->config = AC97C_RESET; - udelay(10); - au1000->ac97_ioport->config = 0x0; - mdelay(5); - - /* Initialise AC97 middle-layer */ - if ((err = snd_ac97_bus(au1000->card, 0, &ops, au1000, &pbus)) < 0) - return err; - - memset(&ac97, 0, sizeof(ac97)); - ac97.private_data = au1000; - if ((err = snd_ac97_mixer(pbus, &ac97, &au1000->ac97)) < 0) - return err; - - return 0; -} - /*------------------------------ Setup / Destroy ----------------------------*/ -void -snd_au1000_free(struct snd_card *card) +static void snd_au1000_free(struct snd_card *card) { struct snd_au1000 *au1000 = card->private_data; - if (au1000->ac97_res_port) { - /* put internal AC97 block into reset */ - au1000->ac97_ioport->cntrl = AC97C_RS; - au1000->ac97_ioport = NULL; - release_and_free_resource(au1000->ac97_res_port); - } - if (au1000->stream[PLAYBACK]) { if (au1000->stream[PLAYBACK]->dma >= 0) free_au1000_dma(au1000->stream[PLAYBACK]->dma); @@ -624,71 +569,176 @@ snd_au1000_free(struct snd_card *card) free_au1000_dma(au1000->stream[CAPTURE]->dma); kfree(au1000->stream[CAPTURE]); } -} + if (au1000->ac97_res_port) { + /* put internal AC97 block into reset */ + au1000->ac97_ioport->cntrl = AC97C_RS; + iounmap(au1000->ac97_ioport); + au1000->ac97_ioport = NULL; + release_and_free_resource(au1000->ac97_res_port); + au1000->ac97_res_port = NULL; + } +} -static struct snd_card *au1000_card; +static struct snd_ac97_bus_ops ops = { + .write = snd_au1000_ac97_write, + .read = snd_au1000_ac97_read, +}; -static int __init -au1000_init(void) +static int __devinit au1000_ac97_probe(struct platform_device *pdev) { int err; + void __iomem *io; + struct resource *r; struct snd_card *card; struct snd_au1000 *au1000; + struct snd_ac97_bus *pbus; + struct snd_ac97_template ac97; err = snd_card_create(-1, "AC97", THIS_MODULE, - sizeof(struct snd_au1000), &card); + sizeof(struct snd_au1000), &card); if (err < 0) - return err; + goto out0; - card->private_free = snd_au1000_free; au1000 = card->private_data; au1000->card = card; - au1000->stream[PLAYBACK] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL); - au1000->stream[CAPTURE ] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL); - /* so that snd_au1000_free will work as intended */ - au1000->ac97_res_port = NULL; - if (au1000->stream[PLAYBACK]) - au1000->stream[PLAYBACK]->dma = -1; - if (au1000->stream[CAPTURE ]) - au1000->stream[CAPTURE ]->dma = -1; - - if (au1000->stream[PLAYBACK] == NULL || - au1000->stream[CAPTURE ] == NULL) { - snd_card_free(card); - return -ENOMEM; + /* TX DMA ID */ + r = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r) { + err = -ENODEV; + snd_printk(KERN_INFO "no TX DMA platform resource!\n"); + goto out1; } - - if ((err = snd_au1000_ac97_new(au1000)) < 0 ) { - snd_card_free(card); - return err; + au1000->dmaid[0] = r->start; + + /* RX DMA ID */ + r = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r) { + err = -ENODEV; + snd_printk(KERN_INFO "no RX DMA platform resource!\n"); + goto out1; } + au1000->dmaid[1] = r->start; - if ((err = snd_au1000_pcm_new(au1000)) < 0) { - snd_card_free(card); - return err; + au1000->stream[PLAYBACK] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL); + if (!au1000->stream[PLAYBACK]) + goto out1; + au1000->stream[PLAYBACK]->dma = -1; + + au1000->stream[CAPTURE] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL); + if (!au1000->stream[CAPTURE]) + goto out2; + au1000->stream[CAPTURE]->dma = -1; + + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r) + goto out3; + + err = -EBUSY; + au1000->ac97_res_port = request_mem_region(r->start, + r->end - r->start + 1, pdev->name); + if (!au1000->ac97_res_port) { + snd_printk(KERN_ERR "ALSA AC97: can't grab AC97 port\n"); + goto out3; } + io = ioremap(r->start, r->end - r->start + 1); + if (!io) + goto out4; + + au1000->ac97_ioport = (struct au1000_ac97_reg *)io; + + spin_lock_init(&au1000->ac97_lock); + + /* configure pins for AC'97 + TODO: move to board_setup.c */ + au_writel(au_readl(SYS_PINFUNC) & ~0x02, SYS_PINFUNC); + + /* Initialise Au1000's AC'97 Control Block */ + au1000->ac97_ioport->cntrl = AC97C_RS | AC97C_CE; + udelay(10); + au1000->ac97_ioport->cntrl = AC97C_CE; + udelay(10); + + /* Initialise External CODEC -- cold reset */ + au1000->ac97_ioport->config = AC97C_RESET; + udelay(10); + au1000->ac97_ioport->config = 0x0; + mdelay(5); + + /* Initialise AC97 middle-layer */ + err = snd_ac97_bus(au1000->card, 0, &ops, au1000, &pbus); + if (err < 0) + goto out5; + + /* from here on let ALSA call the special freeing function */ + card->private_free = snd_au1000_free; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = au1000; + err = snd_ac97_mixer(pbus, &ac97, &au1000->ac97); + if (err < 0) + goto out1; + + err = snd_au1000_pcm_new(au1000); + if (err < 0) + goto out1; + strcpy(card->driver, "Au1000-AC97"); strcpy(card->shortname, "AMD Au1000-AC97"); sprintf(card->longname, "AMD Au1000--AC97 ALSA Driver"); - if ((err = snd_card_register(card)) < 0) { - snd_card_free(card); - return err; - } + err = snd_card_register(card); + if (err < 0) + goto out1; + + printk( KERN_INFO "ALSA AC97: Driver Initialized\n" ); + + platform_set_drvdata(pdev, card); - printk(KERN_INFO "ALSA AC97: Driver Initialized\n"); - au1000_card = card; return 0; + +out5: + au1000->ac97_ioport->cntrl = AC97C_RS; + iounmap(io); +out4: + release_and_free_resource(au1000->ac97_res_port); +out3: + kfree(au1000->stream[PLAYBACK]); +out2: + kfree(au1000->stream[PLAYBACK]); +out1: + snd_card_free(card); +out0: + return err; } -static void __exit au1000_exit(void) +static int __devexit au1000_ac97_remove(struct platform_device *pdev) { - snd_card_free(au1000_card); + struct snd_card *card = platform_get_drvdata(pdev); + snd_card_free(card); + return 0; } -module_init(au1000_init); -module_exit(au1000_exit); +struct platform_driver au1000_ac97c_driver = { + .driver = { + .name = "au1000-ac97c", + .owner = THIS_MODULE, + }, + .probe = au1000_ac97_probe, + .remove = __devexit_p(au1000_ac97_remove), +}; + +static int __init au1000_ac97_load(void) +{ + return platform_driver_register(&au1000_ac97c_driver); +} + +static void __exit au1000_ac97_unload(void) +{ + platform_driver_unregister(&au1000_ac97c_driver); +} +module_init(au1000_ac97_load); +module_exit(au1000_ac97_unload); -- 1.6.2

4 5

[alsa-devel] File SUPPORTED_KERNELS should be updated
by Susan Cragin 28 Apr '09

28 Apr '09

The following is the entire text of that file. ------------------------------------------------------------------------ The alsa-drivers in this package are designed for the following kernels: - Vanilla 2.6.25 or earlier - Vanilla 2.4.31 or earlier - Vanilla 2.2.26 or earlier It's not guaranteed that they work with any newer version than above or modified kernels by distributors. ----------------------------------------------------------------------- Isn't the information obsolete? Aren't the new alsa-drivers in fact destined for 2.6.28 and 2.6.29? Or am I missing something? Susan

2 1

Re: [alsa-devel] No sound with Creative Labs CA0106 Soundblaster
by Takashi Iwai 28 Apr '09

28 Apr '09

At Thu, 16 Apr 2009 23:23:55 +0200, Bernhard Posselt wrote: > > Am Donnerstag 16 April 2009 16:10:29 schrieben Sie: > > At Wed, 15 Apr 2009 04:00:26 +0200, > > > > Bernhard Posselt wrote: > > > Hi, some months ago i bought a new soundcard for my PC which costed 20€. > > > This card is not working. > > > > > > Im using Kubuntu Jaunty Beta, its a fresh install from the beta iso. I > > > also upgraded the alsadrivers with this package > > > http://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver- > > >snapshot.tar.bz2 > > > > > > Still no sound > > > > Please show your alsa-info.sh output (run with --no-upload option). > > > > If it's an unknown model, try to pass some serial number via subsystem > > module option, such as, > > > > # modprobe snd-ca0106 subsystem=0x10131102 > > > > The available numbers are found in ca0106_chip_details[] list in > > ca0106/ca0106_main.c. > > > > > > Takashi > > It's an Audigy Soundblaster LE, or better was. I guess you thought that i > would be using the SE model. No, I just thought you are using a model that isn't listed in the ca0106 driver's white list, no matter which one. Createive tends to ship quite lots of variants with different SSIDs although many are compatible. Takashi

2 1

[alsa-devel] [PATCH] ASoC: TWL4030: Fix gain control for earpiece amplifier
by Joonyoung Shim 28 Apr '09

28 Apr '09

The gain control for earpiece amplifier uses 0dB ~ 12dB according to the TRM, but the present code is implemented to -6dB ~ 6dB. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index a7875c2..bf522a9 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -785,6 +785,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0); static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1); /* + * Gain control for earpiece amplifier + * 0 dB to 12 dB in 6 dB steps (mute instead of -6) + */ +static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1); + +/* * Capture gain after the ADCs * from 0 dB to 31 dB in 1 dB steps */ @@ -859,7 +865,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { 4, 3, 0, output_tvl), SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume", - TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl), + TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl), /* Common capture gain controls */ SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume", -- 1.6.0.4

2 1

[alsa-devel] [PATCH 1/1] indigo-express: add missing 64KHz flags
by Giuliano Pochini 28 Apr '09

28 Apr '09

Indigo-express cards also support 64KHz sampling rate: this patch adds missing SNDRV_PCM_RATE_64000 flags. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> --- alsa-kernel/pci/echoaudio/indigodjx.c__orig 2009-04-27 22:57:20.000000000 +0200 +++ alsa-kernel/pci/echoaudio/indigodjx.c 2009-04-27 22:58:04.000000000 +0200 @@ -88,6 +88,7 @@ static struct snd_pcm_hardware pcm_hardw .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 32000, --- alsa-kernel/pci/echoaudio/indigoiox.c__orig 2009-04-27 22:56:46.000000000 +0200 +++ alsa-kernel/pci/echoaudio/indigoiox.c 2009-04-27 22:57:54.000000000 +0200 @@ -89,6 +89,7 @@ static struct snd_pcm_hardware pcm_hardw .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 32000, -- Giuliano.

2 1

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 28 Apr '09

28 Apr '09

The following changes since commit a396e32ef0f3b98700abb9a6da3530c945e55908: Mark Brown (1): ASoC: s3c-i2s-v2 needs to declare a license for modular builds are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Jon Smirl (1): ASoC: Set the MPC5200 i2s driver to BROKEN status. Mark Brown (1): ASoC: Fix logic in WM8350 master clocking check sound/soc/codecs/wm8350.c | 2 +- sound/soc/fsl/Kconfig | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-)

2 1

[alsa-devel] Benq S32 headphones detection
by Aleh 28 Apr '09

28 Apr '09

Hi, I had a problem with muting speakers when headphones are plugged-in on my Benq S32B. So I added proper board configuration to sound/pci/hda/patch_realtek.c and now it works. I wonder if anyone has similar hardware (S-series Benq laptop) to test the patch (attached) if you find it useful.

2 12

[alsa-devel] [PATCH v2 0/4] Mark mpc5200 i2s driver BROKEN
by Jon Smirl 28 Apr '09

28 Apr '09

Mark the mpc5200 i2s driver BROKEN and lay out a foundation for fixing it. Separate out the mpc5200 core DMA routines from mpc5200_psc_i2s.c into a standalone mpc5200_dma.c. No intention to change how anything works, this is just a code reorg needed for later patches to build on. --- Jon Smirl (4): Rename exports in mpc5200_dma.c Rename the PSC functions to DMA Basic split of mpc5200 DMA code out from mpc5200_psc_i2s Set the MPC5200 i2s driver to BROKEN status. sound/soc/fsl/Kconfig | 6 sound/soc/fsl/Makefile | 2 sound/soc/fsl/mpc5200_dma.c | 463 ++++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_dma.h | 81 +++++ sound/soc/fsl/mpc5200_psc_i2s.c | 644 +++++---------------------------------- 5 files changed, 633 insertions(+), 563 deletions(-) create mode 100644 sound/soc/fsl/mpc5200_dma.c create mode 100644 sound/soc/fsl/mpc5200_dma.h -- Jon Smirl jonsmirl(a)gmail.com

4 11

[alsa-devel] [RFC] SoC WM8940 Driver
by Jonathan Cameron 27 Apr '09

27 Apr '09

From: Jonathan Cameron <jic23(a)cam.ac.uk> Initial support for the WM8940 Mono Codec with speaker driver --- This is my first foray into the world of alsa and aSoC, so I thought I'd make an initial posting of this driver before getting bogged down in testing every last feature works. The board I have has it wired up to i2c limiting options for interface testing and I've currently only tested it as a slave to a pxa271 using i2s. Board is an xbow Imote 2. I will hopefully be able to test a few other interface combinations. All comments welcomed! My apologies if parts of this are complete rubbish! diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c new file mode 100644 index 0000000..3491b9c --- /dev/null +++ b/sound/soc/codecs/wm8940.c @@ -0,0 +1,921 @@ +/* + * wm8940.c -- WM8940 ALSA Soc Audio driver + * + * Author: Jonathan Cameron <jic23(a)cam.ac.uk> + * + * Based on wm8510.c + * Copyright 2006 Wolfson Microelectronics PLC. + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> + * + * 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. + * + * Not currently handled: + * VROI (resistance control for unused outputs. + * AUXMode (inverting vs mixer) + * No means to obtain current gain if alc enabled. + * No use made of gpio + * Auto increment writes (can't think why you'd want to disable it! + * Fast VMID discharge for power down + * Soft Start. + * LoutR control + * DLR and ALR Swaps not enabled + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "wm8940.h" + +struct wm8940_priv { + unsigned int sysclk; + unsigned int mode; + unsigned int clk_inversion; +}; + +static u16 wm8940_reg_defaults[] = { + [WM8940_SOFTRESET] = 0x8940, + [WM8940_POWER1] = 0x0000, + [WM8940_POWER2] = 0x0000, + [WM8940_POWER3] = 0x0000, + [WM8940_IFACE] = 0x0010, + [WM8940_COMPANDINGCTL] = 0x0000, + [WM8940_CLOCK] = 0x0140, + [WM8940_ADDCNTRL] = 0x0000, + [WM8940_GPIO] = 0x0000, + [WM8940_CTLINT] = 0x0002, + [WM8940_DAC] = 0x0000, + [WM8940_DACVOL] = 0x00FF, + + [WM8940_ADC] = 0x0100, + [WM8940_ADCVOL] = 0x00FF, + [WM8940_NOTCH1 ... WM8940_NOTCH8] = 0x0000, + [WM8940_DACLIM1] = 0x0032, + [WM8940_DACLIM2] = 0x0000, + + [WM8940_ALC1] = 0x0038, + [WM8940_ALC2] = 0x000B, + [WM8940_ALC3] = 0x0032, + [WM8940_NOISEGATE] = 0x0000, + [WM8940_PLLN] = 0x0041, + [WM8940_PLLK1] = 0x000C, + [WM8940_PLLK2] = 0x0093, + [WM8940_PLLK3] = 0x00E9, + + [WM8940_ALC4] = 0x0030, + + [WM8940_INPUTCTL] = 0x0002, + [WM8940_PGAGAIN] = 0x0050, + + [WM8940_ADCBOOST] = 0x0002, + + [WM8940_OUTPUTCTL] = 0x0002, + [WM8940_SPKMIX] = 0x0000, + + [WM8940_SPKVOL] = 0x0079, + + [WM8940_MONOMIX] = 0x0000, +}; + +struct snd_soc_codec_device soc_codec_dev_wm8940; + +#define WM8940_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) + +#define WM8940_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + + +static inline unsigned int wm8940_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + if (reg > ARRAY_SIZE(wm8940_reg_defaults)) + return -1; + + return cache[reg]; +} + +static inline int wm8940_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + + if (reg > ARRAY_SIZE(wm8940_reg_defaults)) + return -1; + + cache[reg] = value; + + return 0; +} + +static int wm8940_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[3] = { reg, + (value & 0xff00) >> 8, + (value & 0x00ff) + }; + + wm8940_write_reg_cache(codec, reg, value); + + return (codec->hw_write(codec->control_data, data, 3) == 3) ? 0 : -EIO; + +} + +static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" }; +static const char *wm8940_alc_mode_text[] = {"ALC", "Limiter"}; +static const char *wm8940_mic_bias_levels_text[] = {"0.9", "0.65"}; +static const char *wm8940_filter_mode_text[] = {"Audio", "Application"}; + +static const struct soc_enum wm8940_enum[] = { + SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 1, 4, wm8940_companding), + SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 3, 4, wm8940_companding), + SOC_ENUM_SINGLE(WM8940_ALC3, 8, 2, wm8940_alc_mode_text), + SOC_ENUM_SINGLE(WM8940_INPUTCTL, 8, 2, wm8940_mic_bias_levels_text), + SOC_ENUM_SINGLE(WM8940_ADC, 7, 2, wm8940_filter_mode_text), +}; + +DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1); +DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0); +DECLARE_TLV_DB_SCALE(wm8940_pga_vol_tlv, -1200, 75, 0); +DECLARE_TLV_DB_SCALE(wm8940_alc_min_tlv, -1200, 600, 0); +DECLARE_TLV_DB_SCALE(wm8940_alc_max_tlv, 675, 600, 0); +DECLARE_TLV_DB_SCALE(wm8940_alc_tar_tlv, -2250, 50, 0); +DECLARE_TLV_DB_SCALE(wm8940_lim_boost_tlv, 0, 100, 0); +DECLARE_TLV_DB_SCALE(wm8940_lim_thresh_tlv, -600, 100, 0); +DECLARE_TLV_DB_SCALE(wm8940_adc_tlv, -127500, 50, 1); + +static const struct snd_kcontrol_new wm8940_snd_controls[] = { + SOC_SINGLE("Digital Loopback Switch dac to adc", WM8940_COMPANDINGCTL, + 6, 1, 0), + SOC_ENUM("DAC Companding", wm8940_enum[1]), + SOC_ENUM("ADC Companding", wm8940_enum[0]), + SOC_SINGLE("Companding 8 bit", WM8940_COMPANDINGCTL, 5, 1, 0), + SOC_SINGLE("Digital Loopback Switch adc to dac", WM8940_COMPANDINGCTL, + 0, 1, 0), + + /* Auto level control */ + SOC_ENUM("ALC Mode", wm8940_enum[2]), + SOC_SINGLE("ALC Enable Switch", WM8940_ALC1, 8, 1, 0), + SOC_SINGLE_TLV("ALC Capture Max Gain", WM8940_ALC1, + 3, 7, 1, wm8940_alc_max_tlv), + SOC_SINGLE_TLV("ALC Capture Min Gain", WM8940_ALC1, + 0, 7, 0, wm8940_alc_min_tlv), + SOC_SINGLE_TLV("ALC Capture Target", WM8940_ALC2, + 0, 14, 0, wm8940_alc_tar_tlv), + SOC_SINGLE("ALC Capture Hold", WM8940_ALC2, 4, 10, 0), + SOC_SINGLE("ALC Capture Decay", WM8940_ALC3, 4, 10, 0), + SOC_SINGLE("ALC Capture Attach", WM8940_ALC3, 0, 10, 0), + SOC_SINGLE("ALC ZC Switch", WM8940_ALC4, 1, 1, 0), + SOC_SINGLE("ALC Capture Noise Gate Switch", WM8940_NOISEGATE, + 3, 1, 0), + SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8940_NOISEGATE, + 0, 7, 0), + + SOC_SINGLE("DAC Playback Limiter Switch", WM8940_DACLIM1, 8, 1, 0), + SOC_SINGLE("DAC Playback Limiter Attack", WM8940_DACLIM1, 0, 9, 0), + SOC_SINGLE("DAC Playback Limiter Decay", WM8940_DACLIM1, 4, 11, 0), + SOC_SINGLE_TLV("DAC Playback Limiter Threshold", WM8940_DACLIM2, + 4, 9, 1, wm8940_lim_thresh_tlv), + SOC_SINGLE_TLV("DAC Playback Limiter Boost", WM8940_DACLIM2, + 0, 12, 0, wm8940_lim_boost_tlv), + + SOC_SINGLE("Capture PGA ZC Switch", WM8940_PGAGAIN, 7, 1, 0), + SOC_SINGLE_TLV("Capture PGA Volume", WM8940_PGAGAIN, + 0, 63, 0, wm8940_pga_vol_tlv), + SOC_SINGLE_TLV("Digital Playback Volume", WM8940_DACVOL, + 0, 255, 0, wm8940_adc_tlv), + SOC_SINGLE_TLV("Digital Capture Volume", WM8940_ADCVOL, + 0, 255, 0, wm8940_adc_tlv), + SOC_ENUM("Mic Bias Level", wm8940_enum[3]), + SOC_SINGLE("Capture Boost(+20dB)", WM8940_ADCBOOST, 8, 1, 0), + + SOC_SINGLE_TLV("Speaker Playback Volume", WM8940_SPKVOL, + 0, 63, 0, wm8940_spk_vol_tlv), + SOC_SINGLE("Speaker Playback Mute", WM8940_SPKVOL, 6, 1, 0), + SOC_SINGLE_TLV("Speaker Playback Attenuation", WM8940_SPKVOL, + 8, 1, 1, wm8940_att_tlv), + SOC_SINGLE("Speaker Playback ZC", WM8940_SPKVOL, 7, 1, 0), + + SOC_SINGLE("Mono Out Mute", WM8940_MONOMIX, 6, 1, 0), + SOC_SINGLE_TLV("Mono Out Attenuation", WM8940_MONOMIX, + 7, 1, 1, wm8940_att_tlv), + + SOC_SINGLE("High Pass Filter Switch", WM8940_ADC, 8, 1, 0), + SOC_ENUM("High Pass Filter Mode", wm8940_enum[4]), + SOC_SINGLE("High Pass Filter Cut Off", WM8940_ADC, 4, 7, 0), + SOC_SINGLE("ADC Inversion Switch", WM8940_ADC, 0, 1, 0), + SOC_SINGLE("DAC Inversion Switch", WM8940_DAC, 0, 1, 0), + SOC_SINGLE("DAC Auto Mute Switch", WM8940_DAC, 2, 1, 0), + SOC_SINGLE("ZC Timeout Clock Enable Switch", WM8940_ADDCNTRL, 0, 1, 0), + + /* Notch filters */ + SOC_SINGLE("Notch Filter 1 Switch", WM8940_NOTCH1, 14, 1, 0), + SOC_SINGLE("Notch Filter 2 Switch", WM8940_NOTCH3, 14, 1, 0), + SOC_SINGLE("Notch Filter 3 Switch", WM8940_NOTCH5, 14, 1, 0), + SOC_SINGLE("Notch Filter 4 Switch", WM8940_NOTCH7, 14, 1, 0), + + SOC_SINGLE("Notch Filter 1 Update Switch", WM8940_NOTCH1, 15, 1, 0), + SOC_SINGLE("Notch Filter 2 Update Switch", WM8940_NOTCH3, 15, 1, 0), + SOC_SINGLE("Notch Filter 3 Update Switch", WM8940_NOTCH5, 15, 1, 0), + SOC_SINGLE("Notch Filter 4 Update Switch", WM8940_NOTCH7, 15, 1, 0), + + SOC_SINGLE("Notch Filter 1 Coefficient a0", WM8940_NOTCH1, 0, 16383, 1), + SOC_SINGLE("Notch Filter 2 Coefficient a0", WM8940_NOTCH3, 0, 16383, 1), + SOC_SINGLE("Notch Filter 3 Coefficient a0", WM8940_NOTCH5, 0, 16383, 1), + SOC_SINGLE("Notch Filter 4 Coefficient a0", WM8940_NOTCH7, 0, 16383, 1), + + SOC_SINGLE("Notch Filter 1 Coefficient a1", WM8940_NOTCH2, 0, 16383, 1), + SOC_SINGLE("Notch Filter 2 Coefficient a1", WM8940_NOTCH4, 0, 16383, 1), + SOC_SINGLE("Notch Filter 3 Coefficient a1", WM8940_NOTCH6, 0, 16383, 1), + SOC_SINGLE("Notch Filter 4 Coefficient a1", WM8940_NOTCH8, 0, 16383, 1), +}; + +static const struct snd_kcontrol_new wm8940_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", WM8940_SPKMIX, 1, 1, 0), + SOC_DAPM_SINGLE("Aux Playback Switch", WM8940_SPKMIX, 5, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_SPKMIX, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8940_mono_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", WM8940_MONOMIX, 1, 1, 0), + SOC_DAPM_SINGLE("Aux Playback Switch", WM8940_MONOMIX, 2, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_MONOMIX, 0, 1, 0), +}; + +DECLARE_TLV_DB_SCALE(wm8940_boost_vol_tlv, -1500, 300, 1); +static const struct snd_kcontrol_new wm8940_input_boost_controls[] = { + SOC_DAPM_SINGLE("Mic PGA Switch", WM8940_PGAGAIN, 6, 1, 1), + SOC_DAPM_SINGLE_TLV("Aux Volume", WM8940_ADCBOOST, + 0, 7, 0, wm8940_boost_vol_tlv), + SOC_DAPM_SINGLE_TLV("Mic Volume", WM8940_ADCBOOST, + 4, 7, 0, wm8940_boost_vol_tlv), +}; + +static const struct snd_kcontrol_new wm8940_micpga_controls[] = { + SOC_DAPM_SINGLE("AUX Switch", WM8940_INPUTCTL, 2, 1, 0), + SOC_DAPM_SINGLE("MICP Switch", WM8940_INPUTCTL, 0, 1, 0), + SOC_DAPM_SINGLE("MICN Switch", WM8940_INPUTCTL, 1, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8940_dapm_widgets[] = { + + SND_SOC_DAPM_MIXER("Speaker Mixer", WM8940_POWER3, 2, 0, + &wm8940_speaker_mixer_controls[0], + ARRAY_SIZE(wm8940_speaker_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", WM8940_POWER3, 3, 0, + &wm8940_mono_mixer_controls[0], + ARRAY_SIZE(wm8940_mono_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8940_POWER3, 0, 0), + + SND_SOC_DAPM_PGA("SpkN Out", WM8940_POWER3, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("SpkP Out", WM8940_POWER3, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono Out", WM8940_POWER3, 7, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("MONOOUT"), + SND_SOC_DAPM_OUTPUT("SPKOUTP"), + SND_SOC_DAPM_OUTPUT("SPKOUTN"), + + SND_SOC_DAPM_PGA("Aux Input", WM8940_POWER1, 6, 0, NULL, 0), + SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8940_POWER2, 0, 0), + SND_SOC_DAPM_MIXER("Mic PGA", WM8940_POWER2, 2, 0, + &wm8940_micpga_controls[0], + ARRAY_SIZE(wm8940_micpga_controls)), + SND_SOC_DAPM_MIXER("Boost Mixer", WM8940_POWER2, 4, 0, + &wm8940_input_boost_controls[0], + ARRAY_SIZE(wm8940_input_boost_controls)), + SND_SOC_DAPM_MICBIAS("Mic Bias", WM8940_POWER1, 4, 0), + + SND_SOC_DAPM_INPUT("MICN"), + SND_SOC_DAPM_INPUT("MICP"), + SND_SOC_DAPM_INPUT("AUX"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Mono output mixer */ + {"Mono Mixer", "PCM Playback Switch", "DAC"}, + {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, + {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Speaker output mixer */ + {"Speaker Mixer", "PCM Playback Switch", "DAC"}, + {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, + {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Outputs */ + {"Mono Out", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono Out"}, + {"SpkN Out", NULL, "Speaker Mixer"}, + {"SpkP Out", NULL, "Speaker Mixer"}, + {"SPKOUTN", NULL, "SpkN Out"}, + {"SPKOUTP", NULL, "SpkP Out"}, + + /* Microphone PGA */ + {"Mic PGA", "MICN Switch", "MICN"}, + {"Mic PGA", "MICP Switch", "MICP"}, + {"Mic PGA", "AUX Switch", "AUX"}, + + /* Boost Mixer */ + {"Boost Mixer", "Mic PGA Switch", "Mic PGA"}, + {"Boost Mixer", "Mic Volume", "MICP"}, + {"Boost Mixer", "Aux Volume", "Aux Input"}, + + {"ADC", NULL, "Boost Mixer"}, +}; + +static int wm8940_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8940_dapm_widgets, + ARRAY_SIZE(wm8940_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_widgets(codec); +} +static int wm8940_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8940_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8940_snd_controls[i], codec, + NULL)); + if (err < 0) + return err; + } + + return 0; +} + +static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct wm8940_priv *wm8940 = codec->private_data; + + /* clear fmt and wl */ + u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFF07; + u16 addcntrl = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFF1; + + switch (params_rate(params)) { + case SNDRV_PCM_RATE_8000: + addcntrl |= (0x5 << 1); + break; + case SNDRV_PCM_RATE_11025: + addcntrl |= (0x4 << 1); + break; + case SNDRV_PCM_RATE_16000: + addcntrl |= (0x3 << 1); + break; + case SNDRV_PCM_RATE_22050: + addcntrl |= (0x2 << 1); + break; + case SNDRV_PCM_RATE_32000: + addcntrl |= (0x1 << 1); + break; + case SNDRV_PCM_RATE_44100: + case SNDRV_PCM_RATE_48000: + break; + } + wm8940_write(codec, WM8940_ADDCNTRL, addcntrl); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= (1 << 5); + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= (2 << 5); + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= (3 << 5); + break; + } + + switch (wm8940->mode) { + case SND_SOC_DAIFMT_I2S: + iface |= (2 << 3); + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= (1 << 3); + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= (3 << 3); + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= (3 << 3) | (1 << 7); + break; + } + + switch (wm8940->clk_inversion) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= (1 << 7); + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= (1 << 8); + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= (1 << 8) | (1 << 7); + break; + } + wm8940_write(codec, WM8940_IFACE, iface); + + return 0; +} + +static int wm8940_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8940_read_reg_cache(codec, WM8940_DAC) & 0xffbf; + + return wm8940_write(codec, + WM8940_DAC, + mute ? mute_reg | 0x40 : mute_reg); + +} + +static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8940_priv *wm8940 = codec->private_data; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + wm8940->mode = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + break; + } + wm8940->clk_inversion = fmt & SND_SOC_DAIFMT_INV_MASK; + + return 0; +} + +static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8940_BCLKDIV: + reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFFEF3; + wm8940_write(codec, WM8940_CLOCK, reg | (div << 2)); + break; + case WM8940_MCLKDIV: + reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFF1F; + wm8940_write(codec, WM8940_CLOCK, reg | (div << 5)); + break; + case WM8940_OPCLKDIV: + reg = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFCF; + wm8940_write(codec, WM8940_ADDCNTRL, reg | (div << 4)); + break; + } + return 0; +} + +struct pll_ { + unsigned int pre_scale:2; + unsigned int n:4; + unsigned int k; +}; + +static struct pll_ pll_div; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 24) * 10) +/* This unfortunately is rather different from the wm8510 */ +static void pll_factors(unsigned int target, unsigned int source) +{ + unsigned long long Kpart; + unsigned int K, Ndiv, Nmod; + + Ndiv = target / source; + /* Ndiv would be greater than 12, so add a pre multiply by 2*/ + if (Ndiv > 12) { + /* FIXME: This will loose accuracy, how to deal? */ + printk(KERN_WARNING "Incorrectly handled case\n"); + source <<= 1; + pll_div.pre_scale = 0; + Ndiv = target / source; + } else if (Ndiv < 3) { + source >>= 2; + pll_div.pre_scale = 3; + Ndiv = target / source; + } else if (Ndiv < 6) { + source >>= 1; + pll_div.pre_scale = 2; + } else + pll_div.pre_scale = 1; + + if ((Ndiv < 6) || (Ndiv > 12)) + printk(KERN_WARNING + "WM8510 N value %d outwith recommended range!d\n", + Ndiv); + + pll_div.n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div.k = K; +} + +/* Untested at the moment */ +static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + if (freq_in == 0 || freq_out == 0) { + /* Clock CODEC directly from MCLK */ + reg = wm8940_read_reg_cache(codec, WM8940_CLOCK); + wm8940_write(codec, WM8940_CLOCK, reg & 0x0ff); + + /* Turn off PLL */ + reg = wm8940_read_reg_cache(codec, WM8940_POWER1); + wm8940_write(codec, WM8940_POWER1, reg & 0x1df); + + /* It is also possible to turn it off completely - TODO */ + return 0; + } + + pll_factors(freq_out*8, freq_in); + + wm8940_write(codec, WM8940_PLLN, (pll_div.pre_scale << 4) | pll_div.n); + wm8940_write(codec, WM8940_PLLK1, pll_div.k >> 18); + wm8940_write(codec, WM8940_PLLK2, (pll_div.k >> 9) & 0x1ff); + wm8940_write(codec, WM8940_PLLK3, pll_div.k & 0x1ff); + /* Enable the PLL */ + reg = wm8940_read_reg_cache(codec, WM8940_POWER1); + wm8940_write(codec, WM8940_POWER1, reg | 0x020); + + /* Run CODEC from PLL instead of MCLK */ + reg = wm8940_read_reg_cache(codec, WM8940_CLOCK); + wm8940_write(codec, WM8940_CLOCK, reg | 0x100); + + return 0; +} + +static int wm8940_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 val; + u16 pwr_reg = wm8940_read_reg_cache(codec, WM8940_POWER1) & 0x1F0; + + switch (level) { + case SND_SOC_BIAS_ON: + /* ensure bufioen and biasen */ + pwr_reg |= (1 << 2) | (1 << 3); + /* Enable thermal shutdown */ + val = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL); + wm8940_write(codec, WM8940_OUTPUTCTL, val | 0x2); + /* set vmid to 75k and unmute dac */ + wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1); + break; + case SND_SOC_BIAS_PREPARE: + /* ensure bufioen and biasen */ + pwr_reg |= (1 << 2) | (1 << 3); + /* set vmid to 2.5k for fast start */ + wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x3); + break; + case SND_SOC_BIAS_STANDBY: + /* ensure bufioen and biasen */ + pwr_reg |= (1 << 2) | (1 << 3); + /* set vmid to 300k for standby */ + wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x2); + break; + case SND_SOC_BIAS_OFF: + wm8940_write(codec, WM8940_POWER1, pwr_reg); + break; + } + + return 0; +} + +static int wm8940_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8940_priv *wm8940 = codec->private_data; + + wm8940_write(codec, WM8940_CLOCK, 0); + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 16934400: + case 18432000: + wm8940->sysclk = freq; + return 0; + } + return -EINVAL; +} + +struct snd_soc_dai wm8940_dai = { + .name = "WM8940", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, /* lie - pxa-i2s has min of 2*/ + .rates = WM8940_RATES, + .formats = WM8940_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, /* lie */ + .rates = WM8940_RATES, + .formats = WM8940_FORMATS, + }, + .ops = { + .hw_params = wm8940_i2s_hw_params, + .set_sysclk = wm8940_set_dai_sysclk, + .digital_mute = wm8940_mute, + .set_fmt = wm8940_set_dai_fmt, + .set_clkdiv = wm8940_set_dai_clkdiv, + .set_pll = wm8940_set_dai_pll, + }, +}; +EXPORT_SYMBOL_GPL(wm8940_dai); + +static int wm8940_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8940_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8940_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware + * Could use auto incremented writes to speed this up + */ + for (i = 0; i < ARRAY_SIZE(wm8940_reg_defaults); i++) + wm8940_write(codec, i, cache[i]); + wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8940_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +#define wm8940_reset(c) wm8940_write(c, WM8940_SOFTRESET, 0); + +static int wm8940_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "WM8940"; + codec->owner = THIS_MODULE; + codec->read = wm8940_read_reg_cache; + codec->write = wm8940_write; + codec->set_bias_level = wm8940_set_bias_level; + codec->dai = &wm8940_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults); + codec->reg_cache = kmemdup(wm8940_reg_defaults, + sizeof wm8940_reg_defaults, + GFP_KERNEL); + + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto error_free_cache; + } + ret = wm8940_reset(codec); + if (ret) + goto error_free_cache; + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto error_free_cache; + + codec->bias_level = SND_SOC_BIAS_OFF; + wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8940_add_controls(codec); + wm8940_add_widgets(codec); + + /* Enables the level shifters and non-vmid derived bias + * current generator + */ + ret = wm8940_write(codec, WM8940_POWER1, 0x180); + if (ret < 0) + goto card_err; + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8940: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +error_free_cache: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *wm8940_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int wm8940_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct snd_soc_device *socdev = wm8940_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + ret = wm8940_init(socdev); + if (ret < 0) + pr_err("Failed to initialise WM8940\n"); + + return ret; +} +static int wm8940_i2c_remove(struct i2c_client *client) +{ + return 0; +} + +static const struct i2c_device_id wm8940_i2c_id[] = { + { "wm8940", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id); + +static struct i2c_driver wm8940_i2c_driver = { + .driver = { + .name = "WM8940 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = wm8940_i2c_probe, + .remove = wm8940_i2c_remove, + .id_table = wm8940_i2c_id, +}; + +static int wm8940_add_i2c_device(struct platform_device *pdev, + const struct wm8940_setup_data *setup) +{ + struct i2c_board_info info = { + .type = "wm8940", + .addr = setup->i2c_address, + }; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8940_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&wm8940_i2c_driver); + return -ENODEV; +} + +#endif + +static int wm8940_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8940_setup_data *setup; + struct snd_soc_codec *codec; + struct wm8940_priv *wm8940; + int ret = 0; + + setup = socdev->codec_data; + codec = kzalloc(sizeof *codec, GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + wm8940 = kzalloc(sizeof(*wm8940), GFP_KERNEL); + if (wm8940 == NULL) + return -ENOMEM; + codec->private_data = wm8940; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + wm8940_socdev = socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + codec->hw_write = (hw_write_t)i2c_master_send; + ret = wm8940_add_i2c_device(pdev, setup); + } +#endif + if (ret != 0) + kfree(codec); + return ret; +} + +static int wm8940_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + wm8940_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); + i2c_del_driver(&wm8940_i2c_driver); +#endif + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8940 = { + .probe = wm8940_probe, + .remove = wm8940_remove, + .suspend = wm8940_suspend, + .resume = wm8940_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8940); + +static int __init wm8940_modinit(void) +{ + return snd_soc_register_dai(&wm8940_dai); +} +module_init(wm8940_modinit); + +static void __exit wm8940_exit(void) +{ + snd_soc_unregister_dai(&wm8940_dai); +} +module_exit(wm8940_exit); + +MODULE_DESCRIPTION("ASoC WM8940 driver"); +MODULE_AUTHOR("Jonathan Cameron"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8940.h b/sound/soc/codecs/wm8940.h new file mode 100644 index 0000000..ab1221c --- /dev/null +++ b/sound/soc/codecs/wm8940.h @@ -0,0 +1,102 @@ +/* + * wm8940.h -- WM8940 Soc Audio driver + * + * 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. + */ + +#ifndef _WM8940_H +#define _WM8940_H + +struct wm8940_setup_data { + /* The one I have isn't spi wired, so can't test */ + int i2c_bus; + unsigned short i2c_address; +}; +extern struct snd_soc_dai wm8940_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8940; + +/* WM8940 register space */ +#define WM8940_SOFTRESET 0x00 +#define WM8940_POWER1 0x01 +#define WM8940_POWER2 0x02 +#define WM8940_POWER3 0x03 +#define WM8940_IFACE 0x04 +#define WM8940_COMPANDINGCTL 0x05 +#define WM8940_CLOCK 0x06 +#define WM8940_ADDCNTRL 0x07 +#define WM8940_GPIO 0x08 +#define WM8940_CTLINT 0x09 +#define WM8940_DAC 0x0A +#define WM8940_DACVOL 0x0B + +#define WM8940_ADC 0x0E +#define WM8940_ADCVOL 0x0F +#define WM8940_NOTCH1 0x10 +#define WM8940_NOTCH2 0x11 +#define WM8940_NOTCH3 0x12 +#define WM8940_NOTCH4 0x13 +#define WM8940_NOTCH5 0x14 +#define WM8940_NOTCH6 0x15 +#define WM8940_NOTCH7 0x16 +#define WM8940_NOTCH8 0x17 +#define WM8940_DACLIM1 0x18 +#define WM8940_DACLIM2 0x19 + +#define WM8940_ALC1 0x20 +#define WM8940_ALC2 0x21 +#define WM8940_ALC3 0x22 +#define WM8940_NOISEGATE 0x23 +#define WM8940_PLLN 0x24 +#define WM8940_PLLK1 0x25 +#define WM8940_PLLK2 0x26 +#define WM8940_PLLK3 0x27 + +#define WM8940_ALC4 0x2A + +#define WM8940_INPUTCTL 0x2C +#define WM8940_PGAGAIN 0x2D + +#define WM8940_ADCBOOST 0x2F + +#define WM8940_OUTPUTCTL 0x31 +#define WM8940_SPKMIX 0x32 + +#define WM8940_SPKVOL 0x36 + +#define WM8940_MONOMIX 0x38 + + + +/* Clock divider Id's */ +#define WM8940_BCLKDIV 0 +#define WM8940_MCLKDIV 1 +#define WM8940_OPCLKDIV 2 + +/* MCLK clock dividers */ +#define WM8940_MCLKDIV_1 0 +#define WM8940_MCLKDIV_1_5 1 +#define WM8940_MCLKDIV_2 2 +#define WM8940_MCLKDIV_3 3 +#define WM8940_MCLKDIV_4 4 +#define WM8940_MCLKDIV_6 5 +#define WM8940_MCLKDIV_8 6 +#define WM8940_MCLKDIV_12 7 + +/* BCLK clock dividers */ +#define WM8940_BCLKDIV_1 0 +#define WM8940_BCLKDIV_2 1 +#define WM8940_BCLKDIV_4 2 +#define WM8940_BCLKDIV_8 3 +#define WM8940_BCLKDIV_16 4 +#define WM8940_BCLKDIV_32 5 + +/* PLL Out Dividers */ +#define WM8940_OPCLKDIV_1 0 +#define WM8940_OPCLKDIV_2 1 +#define WM8940_OPCLKDIV_3 2 +#define WM8940_OPCLKDIV_4 3 + +#endif /* _WM8940_H */ +

2 9

[alsa-devel] Creative E-MU 0404 PCI
by A. K. 27 Apr '09

27 Apr '09

Hi all, I'm using EMU 0404 PCI card with Linux (emu10k1), and, AFAIK, not all of this card's features are currently supported. I'd appreciate if you could give me some information about this. Also, I've got troubles with sound recording (playback works quite well). Amixer shows lots of parameters - does anybody know how to configure them right for recording sound ? Thanks for your help. -Jack

1 0

[alsa-devel] device underruns
by Daniel Mack 27 Apr '09

27 Apr '09

Hello, I'm currently stress-testing my snd-usb-caiaq driver under several environments and happen to see a number of such messages echoed once in a while at the start of aplay (1.0.16): underrun!!! (at least 1558380812.470 ms long) underrun!!! (at least 1558380818.673 ms long) underrun!!! (at least 1558380812.465 ms long) Also, pulseaudio (0.9.14) keeps complaining with the following message all over the place: E: module-alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Most likely this is an ALSA driver bug. Please report this issue to the PulseAudio developers. What puzzles me a bit is that I don't really know where and how the driver could confuse the core so badly. All it reports is a valid pointer to the current head in each substream. Is there anything else it could provide for tighter syncing? Any hint how to debug this? Or maybe it's not a bug in this specific driver but in the core somewhere? Thanks, Daniel

2 8

[alsa-devel] [PATCH 1/3] ASoC: cs4270: fix Master Capture Switch polarity
by Daniel Mack 27 Apr '09

27 Apr '09

The control modifies the MUTE register, hence the polarity must be inverted. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)sirena.org.uk> Cc: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 7fa09a3..3c34fe6 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -486,7 +486,7 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0), SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1), SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0), - SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0) + SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1) }; /* -- 1.6.2.1

4 23

[PATCH] ALSA: snd-usb-caiaq: fix reported elapsed periods
by Daniel Mack 27 Apr '09

27 Apr '09

Reset the internal period position counter upon stream startup. This fixes initial aplay underruns and problems related to latency picky applications such as pulseaudio. Bumped the version number to 1.3.14. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/audio.c | 12 +++++++----- sound/usb/caiaq/device.c | 2 +- 2 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 3f45c0f..b13ce76 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -195,11 +195,14 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) debug("%s(%p)\n", __func__, substream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dev->period_out_count[index] = BYTES_PER_SAMPLE + 1; dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; - else + } else { + dev->period_in_count[index] = BYTES_PER_SAMPLE; dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE; - + } + if (dev->streaming) return 0; @@ -300,8 +303,7 @@ static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev, if (!sub) continue; - pb = frames_to_bytes(sub->runtime, - sub->runtime->period_size); + pb = snd_pcm_lib_period_bytes(sub); cnt = (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) ? &dev->period_out_count[stream] : &dev->period_in_count[stream]; diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 6d51770..515de1c 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.13"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.14"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," -- 1.6.2.1

1 0

[alsa-devel] [PATCH] sc6000: enable joystick port
by Krzysztof Helt 27 Apr '09

27 Apr '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Add module parameter to enable or disable joystick port (gameport) on the SC6600 and later cards. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/sc6000.c | 9 +++++++-- 1 files changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index c803b2e..7e2526a 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -56,6 +56,7 @@ static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x300, 0x310, 0x320, 0x330 */ static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 0 */ static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0, 1, 3 */ +static int joystick[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = 0 }; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for sc-6000 based soundcard."); @@ -75,6 +76,8 @@ module_param_array(mpu_irq, int, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for sc-6000 driver."); module_param_array(dma, int, NULL, 0444); MODULE_PARM_DESC(dma, "DMA # for sc-6000 driver."); +module_param_array(joystick, int, NULL, 0444); +MODULE_PARM_DESC(dma, "Enable gameport."); /* * Commands of SC6000's DSP (SBPRO+special). @@ -363,7 +366,7 @@ static int __devinit sc6000_init_mss(char __iomem *vport, int config, static void __devinit sc6000_hw_cfg_encode(char __iomem *vport, int *cfg, long xport, long xmpu, - long xmss_port) + long xmss_port, int joystick) { cfg[0] = 0; cfg[1] = 0; @@ -376,6 +379,8 @@ static void __devinit sc6000_hw_cfg_encode(char __iomem *vport, int *cfg, if (xmss_port == 0xe80) cfg[0] |= 0x10; cfg[0] |= 0x40; /* always set */ + if (!joystick) + cfg[0] |= 0x02; cfg[1] |= 0x80; /* enable WSS system */ cfg[1] &= ~0x40; /* disable IDE */ snd_printd("hw cfg %x, %x\n", cfg[0], cfg[1]); @@ -427,7 +432,7 @@ static int __devinit sc6000_init_board(char __iomem *vport, if (!old) { int cfg[2]; sc6000_hw_cfg_encode(vport, &cfg[0], port[dev], mpu_port[dev], - mss_port[dev]); + mss_port[dev], joystick[dev]); if (sc6000_hw_cfg_write(vport, cfg) < 0) { snd_printk(KERN_ERR "sc6000_hw_cfg_write: failed!\n"); return -EIO; -- 1.6.0.3 ---------------------------------------------------------------------- Kawalerka od 69 tys. zl. Sprawdz atrakcyjne oferty >>> http://link.interia.pl/f2132

2 1

[alsa-devel] Problem bringing up new alsa driver
by Jon Smirl 26 Apr '09

26 Apr '09

My new ALSA driver is playing the first three samples in a loop. It's not progressing forward in the wav file. It's behaving like I'm not calling snd_pcm_period_elapsed() but I have a printk() in it so I'm sure it is being called. Later samples are not getting copied into the DMA buffer, Can anyone give me a clue as to what to look for? root@phyCORE-MPC5200B-tiny:~ aplay phone.wav snd_pcm_playback_open snd_pcm_open snd_pcm_open_file snd_pcm_open_substream mpc5200-psc-ac97 f0002200.sound: psc_dma_startup(substream=c78bd800) mpc5200-psc-ac97 f0002200.sound: psc_dma_pcm_open(substream=c78bd800) Playing WAVE 'phone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo mpc5200-psc-ac97 f0002200.sound: psc_ac97_hw_analog_params(substream=c78bd800) p_size=5513 p_bytes=22052 periods=3 buffer_size=22050 buffer_bytes=88200 channels=2 snd_pcm_prepare stac9766: ac97_analog_prepare rate 44100 mpc5200-psc-ac97 f0002200.sound: psc_dma_trigger(substream=c78bd800, cmd=1) stream_id=0 snd_pcm_hwsync snd_pcm_update_hw_ptr snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 5513 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 11026 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 0 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 5513 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 11026 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 0 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 5513 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 11026 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 0 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 5513 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 11026 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 0 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 5513 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 11026 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 0 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 5513 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 11026 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 0 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 5513 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 11026 snd_pcm_period_elapsed snd_pcm_update_hw_ptr_interrupt 0 Aborted by signal Interrupt... mpc5200-psc-ac97 f0002200.sound: psc_dma_trigger(substream=c78bd800, cmd=0) stream_id=0 mpc5200-psc-ac97 f0002200.sound: psc_dma_shutdown(substream=c78bd800) mpc5200-psc-ac97 f0002200.sound: psc_dma_pcm_close(substream=c78bd800) root@phyCORE-MPC5200B-tiny: -- Jon Smirl jonsmirl(a)gmail.com

4 9

[alsa-devel] Breakage in pcm_lib.c
by Jon Smirl 26 Apr '09

26 Apr '09

These recent commits to pcm_lib.c have fundamentally changed the way snd_pcm_update_hw_ptr_interrupt() works. They have introduced a requirement that snd_pcm_ops.pointer function be able to report the current position of the DMA transfer while the transfer is in progress. The chip I'm working, PowerPC mpc5200, can't do that. There is no access to the internal registers of the DMA engine. On the mpc5200 snd_pcm_ops.pointer always returns the beginning address of the DMA buffer. These checks break audio on the mpc5200 by trying to fix up the pointers when they are actually correct. These checks can probably be modified to take this hardware limitation into account. To simulate our problem on your desktop machine, modify snd_pcm_ops.pointer in your HDA driver to return the front of the DMA buffer instead of the current position inside the buffer. commit bbf6ad1399e9516b0a95de3ad58ffbaed670e4cc Author: Jaroslav Kysela <perex(a)perex.cz> Date: Fri Apr 10 12:28:58 2009 +0200 [ALSA] pcm-midlevel: Add more strict buffer position checks based on jiffies Some drivers like Intel8x0 or Intel HDA are broken for some hardware variants. This patch adds more strict buffer position checks based on jiffies when internal hw_ptr is updated. Enable xrun_debug to see mangling of wrong positions. As a side effect, the hw_ptr interrupt update routine might do slightly better job when many interrupts are lost. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> commit 8b22d943c34b616eefbd6d2f8f197a53b1f29fd0 Author: Takashi Iwai <tiwai(a)suse.de> Date: Fri Mar 20 16:26:15 2009 +0100 ALSA: pcm - Safer boundary checks Make the boundary checks a bit safer. These caese are rare or theoretically won't happen, but nothing bad to keep the checks safer... Signed-off-by: Takashi Iwai <tiwai(a)suse.de> commit ded652f7024bc2d7b6118b561a44187af30841b0 Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Mar 19 10:08:49 2009 +0100 ALSA: pcm - Fix delta calculation at boundary overlap When the hw_ptr_interrupt reaches the boundary, it must check whether the hw_base was already lapped and corret the delta value appropriately. Also, rebasing the hw_ptr needs a correction because buffer_size isn't always aligned to period_size. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> root@phyCORE:~ aplay phone.wav mpc5200-psc-ac97 f0002200.sound: psc_dma_startup(substream=c792d700) mpc5200-psc-ac97 f0002200.sound: psc_dma_pcm_open(substream=c792d700) Playing WAVE 'phone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo mpc5200-psc-ac97 f0002200.sound: psc_ac97_hw_analog_params(substream=c792d700) p_size=5513 p_bytes=44104 periods=3 buffer_size=22050 buffer_bytes=176400 channels=2 rate=44100 format=11 boundary 22050 boundary 1 1445068800 stac9766: ac97_analog_prepare rate 44100 rate is bb80 5622 mpc5200-psc-ac97 f0002200.sound: psc_dma_trigger(substream=c792d700, cmd=1) stream_id=0 psc_dma_pcm_pointer pos 5513 hw_ptr_interrupt 5513 new_hw_ptr 5513 delta 0 buffer_size 22050 old_hw_ptr 0 new_hw_ptr 5513 rate 44100 HZ 300 jdelta 50 jiffies -79179 runtime->hw_ptr_jiffies -79229 Saved jiffies a -79179 psc_dma_pcm_pointer pos 11026 hw_ptr_interrupt 11026 new_hw_ptr 11026 delta 0 buffer_size 22050 old_hw_ptr 5513 new_hw_ptr 11026 rate 44100 HZ 300 jdelta 38 jiffies -79141 runtime->hw_ptr_jiffies -79179 Saved jiffies a -79141 psc_dma_pcm_pointer pos 16539 hw_ptr_interrupt 16539 new_hw_ptr 16539 delta 0 buffer_size 22050 old_hw_ptr 11026 new_hw_ptr 16539 rate 44100 HZ 300 jdelta 37 jiffies -79104 runtime->hw_ptr_jiffies -79141 Saved jiffies a -79104 psc_dma_pcm_pointer pos 0 hw_ptr_interrupt 22052 new_hw_ptr 0 delta -22052 buffer_size 22050 ALSA sound/core/pcm_lib.c:241: PCM: Unexpected hw_pointer value (stream=0, pos=0, intr_ptr=22052) old_hw_ptr 16539 new_hw_ptr 22052 rate 44100 HZ 300 jdelta 38 jiffies -79066 runtime->hw_ptr_jiffies -79104 Saved jiffies a -79066 psc_dma_pcm_pointer pos 5513 hw_ptr_interrupt 27565 new_hw_ptr 27563 delta -2 buffer_size 22050 old_hw_ptr 22052 new_hw_ptr 49613 rate 44100 HZ 300 jdelta 37 jiffies -79029 runtime->hw_ptr_jiffies -79066 ALSA sound/core/pcm_lib.c:269: PCM: hw_ptr skipping! [Q] (pos=5513, delta=27561, period=5513, jdelta=37/187/0) Saved jiffies a -79029 psc_dma_pcm_pointer pos 11026 hw_ptr_interrupt 27565 new_hw_ptr 33076 delta 5511 buffer_size 22050 old_hw_ptr 22052 new_hw_ptr 33076 rate 44100 HZ 300 jdelta 38 jiffies -78991 runtime->hw_ptr_jiffies -79029 ALSA sound/core/pcm_lib.c:269: PCM: hw_ptr skipping! [Q] (pos=11026, delta=11024, period=5513, jdelta=38/74/0) Saved jiffies a -78991 psc_dma_pcm_pointer pos 16539 hw_ptr_interrupt 27565 new_hw_ptr 38589 delta 11024 buffer_size 22050 old_hw_ptr 22052 new_hw_ptr 38589 rate 44100 HZ 300 jdelta 37 jiffies -78954 runtime->hw_ptr_jiffies -78991 ALSA sound/core/pcm_lib.c:269: PCM: hw_ptr skipping! [Q] (pos=16539, delta=16537, period=5513, jdelta=37/112/0) Saved jiffies a -78954 psc_dma_pcm_pointer pos 0 hw_ptr_interrupt 27565 new_hw_ptr 22050 delta -5515 buffer_size 22050 old_hw_ptr 22052 new_hw_ptr 44100 rate 44100 HZ 300 jdelta 38 jiffies -78916 runtime->hw_ptr_jiffies -78954 ALSA sound/core/pcm_lib.c:269: PCM: hw_ptr skipping! [Q] (pos=0, delta=22048, period=5513, jdelta=38/149/0) Saved jiffies a -78916 psc_dma_pcm_pointer pos 5513 hw_ptr_interrupt 27565 new_hw_ptr 27563 delta -2 buffer_size 22050 old_hw_ptr 22052 new_hw_ptr 49613 rate 44100 HZ 300 jdelta 37 jiffies -78879 runtime->hw_ptr_jiffies -78916 ALSA sound/core/pcm_lib.c:269: PCM: hw_ptr skipping! [Q] (pos=5513, delta=27561, period=5513, jdelta=37/187/0) Saved jiffies a -78879 psc_dma_pcm_pointer pos 11026 hw_ptr_interrupt 27565 new_hw_ptr 33076 delta 5511 buffer_size 22050 old_hw_ptr 22052 new_hw_ptr 33076 rate 44100 HZ 300 jdelta 38 jiffies -78841 runtime->hw_ptr_jiffies -78879 ALSA sound/core/pcm_lib.c:269: PCM: hw_ptr skipping! [Q] (pos=11026, delta=11024, period=5513, jdelta=38/74/0) Saved jiffies a -78841 psc_dma_pcm_pointer pos 16539 hw_ptr_interrupt 27565 new_hw_ptr 38589 delta 11024 buffer_size 22050 old_hw_ptr 22052 new_hw_ptr 38589 rate 44100 HZ 300 jdelta 37 jiffies -78804 runtime->hw_ptr_jiffies -78841 ALSA sound/core/pcm_lib.c:269: PCM: hw_ptr skipping! [Q] (pos=16539, delta=16537, period=5513, jdelta=37/112/0) Saved jiffies a -78804 Aborted by signal Interrupt... mpc5200-psc-ac97 f0002200.sound: psc_dma_trigger(substream=c792d700, cmd=0) stream_id=0 mpc5200-psc-ac97 f0002200.sound: psc_dma_shutdown(substream=c792d700) mpc5200-psc-ac97 f0002200.sound: psc_dma_pcm_close(substream=c792d700) root@phyCORE:~ mpc5200-psc-ac97: timeout on ac97 write -- Jon Smirl jonsmirl(a)gmail.com

3 6

[alsa-devel] salsa with noninterleaved needs irqpoll?
by maarten van es 26 Apr '09

26 Apr '09

>>> >>> And what about the sample format? I looked at the drivers; the two >>> sound cards do not have a common sample format. >>> >>> >>> Best regards, >>> Clemens >>> >> > > Thanks, the problem was indeed that I was setting hwparams that are > not supported by the card. I do think salsa lib could return a more > relevant error than "operation not permitted" and I'll see if I can > come up with something for that. > Right now though I am trying to figure out what snd_pcm_writen needs > for its second parameter, void** bufs. I get the error "bad address" > when passing in an array of pointers to a malloc-ed space big enough > to hold a channels' worth of data. Sorry if I am missing obvious > documentation about this, but what should I pass for this parameter? > > Regards > Maarten > Hi again, Ok, now I finally got everything working and am asking the following mostly out of curiosity. The problem I had with snd_pcm_writen by the way has to do somehow with using a chroot environment; everything works when directly booting into the uclibc / salsa system. For it to work I need to add "irqpoll" to the kernel boot options. Without this, the card does not start playing (though no error messages from salsa) and after a few seconds the kernel tells me to add irqpoll to the boot options. On a glibc / alsa based system everything works fine without this option as it does with a c-media card with salsa. What does it do and why is it needed with the hdsp 9632 on a system with salsa? Thanks Maarten

2 1

[alsa-devel] Getting the correct capture mixer
by Ludovico Cavedon 26 Apr '09

26 Apr '09

Hi, I am working on fixing some audio related bugs of a VoIP application (qutecom) which, like most VoIP applications, allows the user to change the playback and recording volume. For playback volume I am opening the "Master" mixer element. However I am not able to figure out how the application should get the correct mixer for playback. Is there any rule? For example, my sound card has two capture volumes, one "Int Mic" and one "Ext Mic". I guess I should choose on the basis of what is the user-selected mic. In fact there are also two mutually exclusive switches "IntMic" and "ExtMic". But how should the application understand that? Here part of the output of amixer, my sound card is "Intel [HDA Intel], device 0: CONEXANT Analog [CONEXANT Analog]" ---------------------------- Simple mixer control 'Ext Mic',0 Capabilities: pvolume cvolume pswitch cswitch Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 20 Capture 0 - 23 Front Left: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Front Right: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Simple mixer control 'ExtMic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [off] Simple mixer control 'Int Mic',0 Capabilities: pvolume cvolume pswitch cswitch Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 20 Capture 0 - 23 Front Left: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Front Right: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Simple mixer control 'IntMic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [on] ------------------------ As an aside, it looks like that the capture volume has no effect, as well all as the cswitches and pswitches of "Int Mic" and "Ext Mic". Thanks, Ludovico

1 0

[alsa-devel] .pointer return value
by Harsha, Priya 25 Apr '09

25 Apr '09

Hi, Can anyone educate me on what value should be returned to .pointer function? Should I be passing the number of frames played till that point from the time stream started or thnumber of frames played such that the bytes is within the buf_size? And if I am using, snd_pcm_indirect_playback_pointer, what should be my input to it? Should I be passing the number of bytes played till that point from the time stream started or the number of bytes within buf_size? Thanks, Harsha

2 2

Re: [alsa-devel] salsa error message setting hwparams
by maarten van es 24 Apr '09

24 Apr '09

On Fri, Apr 24, 2009 at 10:30 AM, maarten van es <maarten.h.van.es(a)gmail.com> wrote: > Hi > > Right now I don't have the machine at hand as I'm at work, but I don't > think the sample format is the problem. Like I said, my program works > fine with the hdsp on glibc / alsa. On uclibc / salsa I get the error > setting any hw param, starting with setting the access method to > rw_interleaved, and including setting the sample rate while I switched > the card already to the correct rate with amixer. > Furthermore, if the problem is an unsupported format I wouldn't expect > the error to be "operation not permitted". > I'll try that program anyway when I'm back home to see what it tells me. > > Thanks > Maarten > > On Fri, Apr 24, 2009 at 10:03 AM, Clemens Ladisch <clemens(a)ladisch.de> wrote: >> (please don't top-post) >> >> maarten van es wrote: >>> Thanks, but that's not it. I make sure I set the correct sample rate >>> with amixer first. >> >> And what about the sample format? I looked at the drivers; the two >> sound cards do not have a common sample format. >> >> >> Best regards, >> Clemens >> > Thanks, the problem was indeed that I was setting hwparams that are not supported by the card. I do think salsa lib could return a more relevant error than "operation not permitted" and I'll see if I can come up with something for that. Right now though I am trying to figure out what snd_pcm_writen needs for its second parameter, void** bufs. I get the error "bad address" when passing in an array of pointers to a malloc-ed space big enough to hold a channels' worth of data. Sorry if I am missing obvious documentation about this, but what should I pass for this parameter? Regards Maarten

1 0

[alsa-devel] Help
by Sudeept Prusti 24 Apr '09

24 Apr '09

Hi all, As i am working on alsa for minimal PCM playback (8khz,mono,16 bit pcm file). When i tried to allocate 1k buffer and send through snd_pcm_writei() call it doesn't play perfect and skips are there in between. If we keep on increasing in allocating the buffer for 31 kb then it could play the buffers randomly. Code is attached for ur reference below. ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- #include <alsa/asoundlib.h> #include <stdio.h> #include <stdlib.h> #define k (31*1024) static void playback(FILE *fd); int main(int argc, char *argv[]) { FILE *fd; fd = fopen(argv[1],"r"); playback(fd); return 0; } static void playback(FILE *fd) { int rc,err; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; snd_pcm_uframes_t frames; unsigned char *buffer; int seek_size=0; int nonblock = 0; printf("Entered Playback Module \n"); /* Open PCM device for playback. */ rc = snd_pcm_open(&handle, "default",SND_PCM_STREAM_PLAYBACK, 0); if(rc >= 0) { printf("Open PCM Success: %d \n",rc); } if (rc < 0) { fprintf(stderr,"unable to open pcm device: %s\n",snd_strerror(rc)); exit(1); } /* PCM Hardware Parameters Allocation. */ err = snd_pcm_hw_params_malloc (&params); if(err>=0) { printf("Hardware Params Allocation Success: %d \n",err); } if (err < 0) { fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err)); exit (1); } /* Intialize PCM Hardware Parameters */ err = snd_pcm_hw_params_any(handle, params); if(err >=0) { printf("PCM HW params Initialization Success : %d \n",err); } if (err < 0) { printf("Cannot initialize hardware parameter structure \n"); exit(EXIT_FAILURE); } /* Set the PCM Hardware parameters. */ /* Interleaved mode */ err = snd_pcm_hw_params_set_access(handle, params,SND_PCM_ACCESS_RW_INTERLEAVED); if (err >= 0) { printf("Access Type Available: %d \n",err); } if (err < 0) { printf("Access Type Not-available: %d \n",err); exit(EXIT_FAILURE); } /* Signed 16-bit little-endian format */ err = snd_pcm_hw_params_set_format(handle, params,SND_PCM_FORMAT_S16_LE); if (err >= 0) { printf("Sample format Available: %d \n",err); } if (err < 0) { printf("Sample format Non-Available \n"); exit(EXIT_FAILURE); } /* One Channel (Mono) */ err = snd_pcm_hw_params_set_channels(handle, params, 1); if (err >= 0) { printf("Channels Available: %d \n",err); } if (err < 0) { printf("Channels Non-Available \n"); exit(EXIT_FAILURE); } /* 8khz sampling rate */ err = snd_pcm_hw_params_set_rate (handle, params, 8000, 0); if (err >= 0) { printf ("Sample Rate Set Success: %d \n",err); } if (err < 0) { fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err)); exit (EXIT_FAILURE); } /* Write the parameters to the alsa-driver */ rc = snd_pcm_hw_params(handle, params); if(rc >=0) { printf("Set HW Parameters Success: %d \n",rc); } if (rc < 0) { fprintf(stderr,"Unable to Set HW Parameters: %s\n",snd_strerror(rc)); exit(1); } /* Allocate buffer*/ buffer = (unsigned char *) malloc(k); if (buffer == NULL) { printf("Not Enough Memory: %c \n",buffer); exit(1); } while (!feof(fd)) { if(fd != NULL) { printf("File pointer is not NULL \n"); } printf("Current Position of File Pointer is: %d\n",ftell(fd)); rc = fread(buffer,k,1,fd); printf("Data Read from File is: %d \n",rc); if(rc > 0) { printf("File Read Success \n"); } seek_size = seek_size + k; fseek(fd,seek_size,SEEK_SET); if (rc == 0) { printf("End of File Reached: %d \n",rc); break; } rc = snd_pcm_writei(handle, buffer, k); printf("Write: %d bytes \n",rc); if (rc == -EPIPE) { printf("underrun occurred \n"); snd_pcm_prepare(handle); } else if (rc < 0) { fprintf(stderr,"Error From Writei Call: %s\n",snd_strerror(rc)); exit(1); } }//End of While fclose(fd); snd_pcm_drain(handle); snd_pcm_close(handle); free(buffer); }//End of Playback -------------------------------------------------------------------------------------------------------------- Can anyone help me out where i am doing wrong.Ur's valuable suggestions are welcomed. Best Regards, Sudeept

3 3

[alsa-devel] [PATCH 0/1] ASoC: 4 channel mode for beagle board
by Peter Ujfalusi 24 Apr '09

24 Apr '09

Hello, Resending the last part of the previous series, which adds 4 channel mode support for the Beagle board. --- Peter Ujfalusi (1): ASoC: Beagle: Add support for 4 channel sound/soc/omap/omap3beagle.c | 26 ++++++++++++++++++-------- 1 files changed, 18 insertions(+), 8 deletions(-)

2 2

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 24 Apr '09

24 Apr '09

The following changes since commit 9b5b0c01598f9782690b09ce6c49f4ba116dde44: Mark Brown (1): Merge branch 'for-2.6.30' into for-2.6.31 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Eric Miao (2): ASoC: simplify the SSP DMA parameters settings by run-time generation ASoC: change stereo/mono to 32-bit/16-bit for pxa-ssp Joonyoung Shim (2): ASoC: OMAP: Add checking to detect bufferless pcms ASoC: TWL4030: Add VDL path support Mark Brown (16): ASoC: Factor out generic widget power checks ASoC: Factor out DAPM power checks for DACs and ADCs ASoC: Make the DAPM power check an operation on the widget ASoC: Fix offset of freqmode in WM8580 PLL configuration ASoC: Fix WM8580 volume update handling for large register changes ASoC: Add power supply widget to DAPM ASoC: Use DAPM supply widget for WM8903 charge pump ASoC: Support CLK_DSP in WM8903 ASoC: Optimise configuration of WM8903 DC servo ASoC: Actively manage the DC servo for WM8903 ASoC: Remove redundant rate constraint for WM8903 ASoC: Implement WM8903 digital sidetone support Merge commit 'takashi/fix/asoc' into for-2.6.30 Merge branch 'for-2.6.30' into for-2.6.31 ASoC: s3c-i2s-v2 needs to declare a license for modular builds Merge branch 'for-2.6.30' into for-2.6.31 Peter Ujfalusi (2): ASoC: OMAP: Add 4 channel support to mcbsp ASoC: TWL4030: Add 4 channel TDM support Russell King - ARM Linux (1): ASoC: Fix warning in wm9705 Takashi Iwai (1): ASoC: remove non-existing referece to CONFIG_SND_SOC_CODEC_WM8991 Documentation/sound/alsa/soc/dapm.txt | 1 + include/sound/soc-dapm.h | 9 +- sound/soc/codecs/Makefile | 1 - sound/soc/codecs/twl4030.c | 308 +++++++++++++++++++-------------- sound/soc/codecs/twl4030.h | 11 ++ sound/soc/codecs/wm8580.c | 16 +- sound/soc/codecs/wm8903.c | 118 ++++++++----- sound/soc/codecs/wm9705.c | 2 +- sound/soc/omap/omap-mcbsp.c | 5 +- sound/soc/omap/omap-pcm.c | 9 +- sound/soc/pxa/pxa-ssp.c | 203 +++++----------------- sound/soc/s3c24xx/s3c-i2s-v2.c | 3 +- sound/soc/soc-dapm.c | 139 +++++++++++----- 13 files changed, 438 insertions(+), 387 deletions(-)

4 24

Re: [alsa-devel] salsa error message setting hwparams
by Clemens Ladisch 24 Apr '09

24 Apr '09

(please don't top-post) maarten van es wrote: > Thanks, but that's not it. I make sure I set the correct sample rate > with amixer first. And what about the sample format? I looked at the drivers; the two sound cards do not have a common sample format. Best regards, Clemens

1 0

[alsa-devel] salsa error message setting hwparams
by maarten van es 24 Apr '09

24 Apr '09

Hi everyone, I am running out of ideas of why I am getting the error "operation not permitted" when I try setting hw parameters with one of my audio cards. I have a custom linux system based on uclibc with salsa for audio and use the default device for playback. This sytem works fine on a via C7 based pc with via audio. When I try an exact copy of the system on an athlon64 based system with a HDPS9632 I get this error message. The card works fine with other linux systems. I tested several (recent) kernels with my system, all the same result. Any ideas why I might be getting this message? Maarten

2 1

[alsa-devel] [PATCH] sc6000: fix older card initialization
by Krzysztof Helt 24 Apr '09

24 Apr '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The last patch to handle newer cards like SC7000 broke initialization of the SC6000. Fix this. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/sc6000.c | 22 ++++++++++++---------- 1 files changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index 983ab7e..c803b2e 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -391,7 +391,6 @@ static int __devinit sc6000_init_board(char __iomem *vport, int config = mss_config | sc6000_mpu_irq_to_softcfg(mpu_irq[dev]); int err; - int cfg[2]; int old = 0; err = sc6000_dsp_reset(vport); @@ -421,11 +420,18 @@ static int __devinit sc6000_init_board(char __iomem *vport, answer, version[0], version[1]); /* set configuration */ - sc6000_hw_cfg_encode(vport, &cfg[0], port[dev], mpu_port[dev], - mss_port[dev]); - if (sc6000_hw_cfg_write(vport, cfg) < 0) { - snd_printk(KERN_ERR "sc6000_hw_cfg_write: failed!\n"); - return -EIO; + sc6000_write(vport, COMMAND_5C); + if (sc6000_read(vport) < 0) + old = 1; + + if (!old) { + int cfg[2]; + sc6000_hw_cfg_encode(vport, &cfg[0], port[dev], mpu_port[dev], + mss_port[dev]); + if (sc6000_hw_cfg_write(vport, cfg) < 0) { + snd_printk(KERN_ERR "sc6000_hw_cfg_write: failed!\n"); + return -EIO; + } } err = sc6000_setup_board(vport, config); if (err < 0) { @@ -434,10 +440,6 @@ static int __devinit sc6000_init_board(char __iomem *vport, } sc6000_dsp_reset(vport); - sc6000_write(vport, COMMAND_5C); - if (sc6000_read(vport) < 0) - old = 1; - sc6000_dsp_reset(vport); if (!old) { sc6000_write(vport, COMMAND_60); -- 1.6.0.3 ---------------------------------------------------------------------- Dzwonki na komork�! Sprawdz >> http://link.interia.pl/f2127

2 1

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 24 Apr '09

24 Apr '09

The following changes since commit a1992db55d80297544a65160ddb98afba45f7759: Takashi Iwai (1): ASoC: remove non-existing referece to CONFIG_SND_SOC_CODEC_WM8991 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Mark Brown (4): ASoC: Fix offset of freqmode in WM8580 PLL configuration ASoC: Fix WM8580 volume update handling for large register changes Merge commit 'takashi/fix/asoc' into for-2.6.30 ASoC: s3c-i2s-v2 needs to declare a license for modular builds sound/soc/codecs/wm8580.c | 16 ++++++++++------ sound/soc/s3c24xx/s3c-i2s-v2.c | 3 ++- 2 files changed, 12 insertions(+), 7 deletions(-)

2 4

[alsa-devel] trouble with alsa, wolfson, and TI OMAP35xx McBSP
by twebb 23 Apr '09

23 Apr '09

I'm having difficulties getting a Wolfson codec running on McBSP1 of an OMAP 35xx. Here are the particulars: 1) Using Linux kernel 2.6.27-omap1, with an ALSA sound driver ported from a Wolfson supplied source tree labeled 2.6.29 (“dev” branch) 2) Using the TI OMAP3530 SOC married to a Wolfson WM8978 codec via McBSP1. I2C control, I2S pcm data. 3) We are running the codec digital block at 1.8V and the analog block at 3.3V 4) The WM8978 codec is synthesizing FCLK and BCLK, codec is the master. McBsp is the slave. 5) FCLK=44.1KHz, BCLK= 32 * FCLK, MCLK = 13Mhz 6) Framing is I2S stereo, 16 bit word I'm seeing two main problems: Problem 1: The WM8978 sample rate PLL does not seem to be stable while DCVDD = DBVDD = 1.8V. FCLK mean = 44.5KHz and jittering. Increasing DCVDD and DBVDD voltage above 2V increases PLL stability. We have a workaround for this for now. Problem 2: Test tone is being presented by the user application, providing a 1Khz tone sampled at 44.1Khz. The data are S16_LE, right channel only. Left channel is quiet. The data seems to slip back and forth from left to right channel. This is reproducable and verified with a scope trace. Anybody have any ideas what might be going wrong here? Traces for codec reg dump and mcbsp are attached. Thanks, twebb

9 11

[alsa-devel] [PATCH 0/3] ASoC: 4 channel support for twl4030 and omap-mcbsp - Try2
by Peter Ujfalusi 23 Apr '09

23 Apr '09

Hello, This series adds support for 4 channel DSP mode (A or B) to OMAP McBSP and also to the twl4030 codec driver. As an example the Beagle board file is modified to be able to use 4 channel mode. The change in twl4030 is made in a way, that it will not enable the 4 channel mode when the codec is in OPTION2 mode. Since some of the bits in the OPTION register are shared and have different effect in OPTION1 and OPTION2 modes, an extra care is taken to prevent unvanted modification to the register. Note: The series is on top of Mark's tree in regard of the twl4030 codec, so it is on top of the 'ASoC: TWL4030: Add VDL path support' from Joonyoung Shim. --- Peter Ujfalusi (2): ASoC: OMAP: Add 4 channel support to mcbsp ASoC: TWL4030: Add 4 channel TDM support Ujfalusi Peter (Nokia-D/Tampere) (1): ASoC: Beagle: Add support for 4 channel mode sound/soc/codecs/twl4030.c | 52 ++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/twl4030.h | 11 +++++++++ sound/soc/omap/omap-mcbsp.c | 5 ++- sound/soc/omap/omap3beagle.c | 26 ++++++++++++++------ 4 files changed, 82 insertions(+), 12 deletions(-)

2 5

[alsa-devel] [PATCH] alsa-info.sh: add command-line flag to print alsa info to stdout
by Luke Yelavich 23 Apr '09

23 Apr '09

Add the --stdout command-line flag to allow displaying the collected alsa information directly to standard output. This is useful for bug information collecting systems, such as Ubuntu's apport. Signed-off-by: Luke Yelavich <themuso(a)ubuntu.com> --- utils/alsa-info.sh | 34 +++++++++++++++++++++++++++++----- 1 files changed, 29 insertions(+), 5 deletions(-) diff --git a/utils/alsa-info.sh b/utils/alsa-info.sh index 0fff671..ee0de16 100755 --- a/utils/alsa-info.sh +++ b/utils/alsa-info.sh @@ -277,6 +277,12 @@ case "$1" in REPEAT="" shift ;; + --stdout) + DIALOG="" + NOUPLOAD="yes" + QUESTION="no" + TOSTDOUT="yes" + ;; esac done @@ -627,6 +633,17 @@ then fi fi ;; + --stdout) + NOUPLOAD="yes" + withdevices + withconfigs + withaplay + withamixer + withalsactl + withlsmod + cat $FILE + rm $FILE + ;; --about) echo "Written/Tested by the following users of #alsa on irc.freenode.net:" echo "" @@ -635,6 +652,7 @@ then echo " gnubien - Various script ideas / Testing" echo " GrueMaster - HDA Intel specific items / Testing" echo " olegfink - Script update function" + echo " TheMuso - display to stdout functionality" exit 0 ;; *) @@ -652,6 +670,8 @@ then echo " --no-upload (do not upload contents to remote server)" echo " --pastebin (use http://pastebin.ca) as remote server" echo " instead www.alsa-project.org" + echo " --stdout (print alsa information to standard output" + echo " instead of a file)" echo " --about (show some information about the script)" echo " --debug (will run the script as normal, but will not" echo " delete $FILE)" @@ -668,7 +688,9 @@ fi if [ -n "$NOUPLOAD" ]; then - mv $FILE $NFILE || exit 1 + if [ -z "$TOSTDOUT" ]; then + mv $FILE $NFILE || exit 1 + fi if [[ -n $DIALOG ]] then @@ -686,10 +708,12 @@ if [ -n "$NOUPLOAD" ]; then echo "Your ALSA information can be seen by looking in $NFILE" echo "" else - echo "You requested that your information was NOT automatically uploaded to the $WWWSERVICE" - echo "" - echo "Your ALSA information can be seen by looking in $NFILE" - echo "" + if [ -z "$TOSTDOUT" ]; then + echo "You requested that your information was NOT automatically uploaded to the $WWWSERVICE" + echo "" + echo "Your ALSA information can be seen by looking in $NFILE" + echo "" + fi fi fi -- 1.6.0.4

2 2

[alsa-devel] [RFC PATCH 5/5] ASoC: simplify the SSP DMA parameters settings by run-time generation
by Eric Miao 23 Apr '09

23 Apr '09

The SSP DMA parameters can actually be easily generated at run-time since they are almost similar except for the FIFO width and direction. Another benefit is the re-use of information from 'struct ssp_device', like SSDR physical FIFO address and DRCMR register index for both directions. Signed-off-by: Eric Miao <eric.miao(a)marvell.com> --- sound/soc/pxa/pxa-ssp.c | 206 ++++++++++------------------------------------- 1 files changed, 43 insertions(+), 163 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 2ce4fb6..fb7235b 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -50,139 +50,6 @@ struct ssp_priv { #endif }; -#define PXA2xx_SSP1_BASE 0x41000000 -#define PXA27x_SSP2_BASE 0x41700000 -#define PXA27x_SSP3_BASE 0x41900000 -#define PXA3xx_SSP4_BASE 0x41a00000 - -static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_out = { - .name = "SSP1 PCM Mono out", - .dev_addr = PXA2xx_SSP1_BASE + SSDR, - .drcmr = &DRCMR(14), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_in = { - .name = "SSP1 PCM Mono in", - .dev_addr = PXA2xx_SSP1_BASE + SSDR, - .drcmr = &DRCMR(13), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_out = { - .name = "SSP1 PCM Stereo out", - .dev_addr = PXA2xx_SSP1_BASE + SSDR, - .drcmr = &DRCMR(14), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_in = { - .name = "SSP1 PCM Stereo in", - .dev_addr = PXA2xx_SSP1_BASE + SSDR, - .drcmr = &DRCMR(13), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_out = { - .name = "SSP2 PCM Mono out", - .dev_addr = PXA27x_SSP2_BASE + SSDR, - .drcmr = &DRCMR(16), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_in = { - .name = "SSP2 PCM Mono in", - .dev_addr = PXA27x_SSP2_BASE + SSDR, - .drcmr = &DRCMR(15), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_out = { - .name = "SSP2 PCM Stereo out", - .dev_addr = PXA27x_SSP2_BASE + SSDR, - .drcmr = &DRCMR(16), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_in = { - .name = "SSP2 PCM Stereo in", - .dev_addr = PXA27x_SSP2_BASE + SSDR, - .drcmr = &DRCMR(15), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_out = { - .name = "SSP3 PCM Mono out", - .dev_addr = PXA27x_SSP3_BASE + SSDR, - .drcmr = &DRCMR(67), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_in = { - .name = "SSP3 PCM Mono in", - .dev_addr = PXA27x_SSP3_BASE + SSDR, - .drcmr = &DRCMR(66), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_out = { - .name = "SSP3 PCM Stereo out", - .dev_addr = PXA27x_SSP3_BASE + SSDR, - .drcmr = &DRCMR(67), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_in = { - .name = "SSP3 PCM Stereo in", - .dev_addr = PXA27x_SSP3_BASE + SSDR, - .drcmr = &DRCMR(66), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_out = { - .name = "SSP4 PCM Mono out", - .dev_addr = PXA3xx_SSP4_BASE + SSDR, - .drcmr = &DRCMR(67), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_in = { - .name = "SSP4 PCM Mono in", - .dev_addr = PXA3xx_SSP4_BASE + SSDR, - .drcmr = &DRCMR(66), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_out = { - .name = "SSP4 PCM Stereo out", - .dev_addr = PXA3xx_SSP4_BASE + SSDR, - .drcmr = &DRCMR(67), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_in = { - .name = "SSP4 PCM Stereo in", - .dev_addr = PXA3xx_SSP4_BASE + SSDR, - .drcmr = &DRCMR(66), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH4, -}; - static void dump_registers(struct ssp_device *ssp) { dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", @@ -194,25 +61,33 @@ static void dump_registers(struct ssp_device *ssp) ssp_read_reg(ssp, SSACD)); } -static struct pxa2xx_pcm_dma_params *ssp_dma_params[4][4] = { - { - &pxa_ssp1_pcm_mono_out, &pxa_ssp1_pcm_mono_in, - &pxa_ssp1_pcm_stereo_out, &pxa_ssp1_pcm_stereo_in, - }, - { - &pxa_ssp2_pcm_mono_out, &pxa_ssp2_pcm_mono_in, - &pxa_ssp2_pcm_stereo_out, &pxa_ssp2_pcm_stereo_in, - }, - { - &pxa_ssp3_pcm_mono_out, &pxa_ssp3_pcm_mono_in, - &pxa_ssp3_pcm_stereo_out, &pxa_ssp3_pcm_stereo_in, - }, - { - &pxa_ssp4_pcm_mono_out, &pxa_ssp4_pcm_mono_in, - &pxa_ssp4_pcm_stereo_out, &pxa_ssp4_pcm_stereo_in, - }, +struct pxa2xx_pcm_dma_data { + struct pxa2xx_pcm_dma_params params; + char name[20]; }; +static struct pxa2xx_pcm_dma_params * +ssp_get_dma_params(struct ssp_device *ssp, int stereo, int out) +{ + struct pxa2xx_pcm_dma_data *dma; + + dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL); + if (dma == NULL) + return NULL; + + snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id, + stereo ? "Stereo" : "Mono", out ? "out" : "in"); + + dma->params.name = dma->name; + dma->params.drcmr = &DRCMR(out ? ssp->drcmr_tx : ssp->drcmr_rx); + dma->params.dcmd = (out ? (DCMD_INCSRCADDR | DCMD_FLOWTRG) : + (DCMD_INCTRGADDR | DCMD_FLOWSRC)) | + (stereo ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16; + dma->params.dev_addr = ssp->phys_base + SSDR; + + return &dma->params; +} + static int pxa_ssp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -225,6 +100,11 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, clk_enable(priv->ssp->clk); ssp_disable(priv->ssp); } + + if (cpu_dai->dma_data) { + kfree(cpu_dai->dma_data); + cpu_dai->dma_data = NULL; + } return ret; } @@ -239,6 +119,11 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, ssp_disable(priv->ssp); clk_disable(priv->ssp->clk); } + + if (cpu_dai->dma_data) { + kfree(cpu_dai->dma_data); + cpu_dai->dma_data = NULL; + } } #ifdef CONFIG_PM @@ -611,25 +496,20 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->ssp; - int dma = 0, chn = params_channels(params); + int chn = params_channels(params); u32 sscr0; u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; - /* select correct DMA params */ - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) - dma = 1; /* capture DMA offset is 1,3 */ - /* Network mode with one active slot (ttsa == 1) can be used - * to force 16-bit frame width on the wire (for S16_LE), even - * with two channels. Use 16-bit DMA transfers for this case. - */ - if (((chn == 2) && (ttsa != 1)) || (width == 32)) - dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */ - - cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma]; + printk("%s\n", __func__); + /* generate correct DMA params */ + if (cpu_dai->dma_data) + kfree(cpu_dai->dma_data); - dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma); + cpu_dai->dma_data = ssp_get_dma_params(ssp, + ((chn == 2) && (ttsa != 1)) || (width == 32), + substream->stream == SNDRV_PCM_STREAM_PLAYBACK); /* we can only change the settings if the port is not in use */ if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) -- 1.6.0.4

3 12

[alsa-devel] [PATCH] keywest: Convert to new-style i2c driver
by Jean Delvare 23 Apr '09

23 Apr '09

The legacy i2c binding model is going away soon, so convert the ppc keywest sound driver to the new model or it will break. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Benjamin Herrenschmidt <benh(a)kernel.crashing.org> Cc: Takashi Iwai <tiwai(a)suse.de> --- Takashi, please push this patch to Linus quickly, as this is blocking the removal of the legacy i2c binding model, which is scheduled for 2.6.30. I did not get any test report for this one, but it's relatively simple so I am confident I got it right. sound/ppc/keywest.c | 82 +++++++++++++++++++++++++-------------------------- 1 file changed, 41 insertions(+), 41 deletions(-) --- linux-2.6.30-rc2.orig/sound/ppc/keywest.c 2009-04-20 22:40:27.000000000 +0200 +++ linux-2.6.30-rc2/sound/ppc/keywest.c 2009-04-20 22:47:34.000000000 +0200 @@ -33,26 +33,25 @@ static struct pmac_keywest *keywest_ctx; -static int keywest_attach_adapter(struct i2c_adapter *adapter); -static int keywest_detach_client(struct i2c_client *client); - -struct i2c_driver keywest_driver = { - .driver = { - .name = "PMac Keywest Audio", - }, - .attach_adapter = &keywest_attach_adapter, - .detach_client = &keywest_detach_client, -}; - - #ifndef i2c_device_name #define i2c_device_name(x) ((x)->name) #endif +static int keywest_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + i2c_set_clientdata(client, keywest_ctx); + return 0; +} + +/* + * This is kind of a hack, best would be to turn powermac to fixed i2c + * bus numbers and declare the sound device as part of platform + * initialization + */ static int keywest_attach_adapter(struct i2c_adapter *adapter) { - int err; - struct i2c_client *new_client; + struct i2c_board_info info; if (! keywest_ctx) return -EINVAL; @@ -60,46 +59,47 @@ static int keywest_attach_adapter(struct if (strncmp(i2c_device_name(adapter), "mac-io", 6)) return 0; /* ignored */ - new_client = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); - if (! new_client) - return -ENOMEM; - - new_client->addr = keywest_ctx->addr; - i2c_set_clientdata(new_client, keywest_ctx); - new_client->adapter = adapter; - new_client->driver = &keywest_driver; - new_client->flags = 0; - - strcpy(i2c_device_name(new_client), keywest_ctx->name); - keywest_ctx->client = new_client; + memset(&info, 0, sizeof(struct i2c_board_info)); + strlcpy(info.type, "keywest", I2C_NAME_SIZE); + info.addr = keywest_ctx->addr; + keywest_ctx->client = i2c_new_device(adapter, &info); - /* Tell the i2c layer a new client has arrived */ - if (i2c_attach_client(new_client)) { - snd_printk(KERN_ERR "tumbler: cannot attach i2c client\n"); - err = -ENODEV; - goto __err; - } - + /* + * Let i2c-core delete that device on driver removal. + * This is safe because i2c-core holds the core_lock mutex for us. + */ + list_add_tail(&keywest_ctx->client->detected, + &keywest_ctx->client->driver->clients); return 0; - - __err: - kfree(new_client); - keywest_ctx->client = NULL; - return err; } -static int keywest_detach_client(struct i2c_client *client) +static int keywest_remove(struct i2c_client *client) { + i2c_set_clientdata(client, NULL); if (! keywest_ctx) return 0; if (client == keywest_ctx->client) keywest_ctx->client = NULL; - i2c_detach_client(client); - kfree(client); return 0; } + +static const struct i2c_device_id keywest_i2c_id[] = { + { "keywest", 0 }, + { } +}; + +struct i2c_driver keywest_driver = { + .driver = { + .name = "PMac Keywest Audio", + }, + .attach_adapter = keywest_attach_adapter, + .probe = keywest_probe, + .remove = keywest_remove, + .id_table = keywest_i2c_id, +}; + /* exported */ void snd_pmac_keywest_cleanup(struct pmac_keywest *i2c) { -- Jean Delvare

3 12

[alsa-devel] [RFC PATCH 4/5] ASoC: remove those unnecessary #ifdef CONFIG_PXA3xx .. #endif
by Eric Miao 23 Apr '09

23 Apr '09

These are unnecessary since the compiler is smart enough to optimize the code away if CONFIG_PXA3xx is not defined, when cpu_is_pxa3xx() will just be expanded to constant 0. Signed-off-by: Eric Miao <eric.miao(a)marvell.com> --- sound/soc/pxa/pxa-ssp.c | 15 --------------- 1 files changed, 0 insertions(+), 15 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index a65993c..2ce4fb6 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -360,24 +360,20 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, case PXA_SSP_AUDIO_DIV_SCDB: val = ssp_read_reg(ssp, SSACD); val &= ~SSACD_SCDB; -#if defined(CONFIG_PXA3xx) if (cpu_is_pxa3xx()) val &= ~SSACD_SCDX8; -#endif switch (div) { case PXA_SSP_CLK_SCDB_1: val |= SSACD_SCDB; break; case PXA_SSP_CLK_SCDB_4: break; -#if defined(CONFIG_PXA3xx) case PXA_SSP_CLK_SCDB_8: if (cpu_is_pxa3xx()) val |= SSACD_SCDX8; else return -EINVAL; break; -#endif default: return -EINVAL; } @@ -403,10 +399,8 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, struct ssp_device *ssp = priv->ssp; u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70; -#if defined(CONFIG_PXA3xx) if (cpu_is_pxa3xx()) ssp_write_reg(ssp, SSACDD, 0); -#endif switch (freq_out) { case 5622000: @@ -431,7 +425,6 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, break; default: -#ifdef CONFIG_PXA3xx /* PXA3xx has a clock ditherer which can be used to generate * a wider range of frequencies - calculate a value for it. */ @@ -452,8 +445,6 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, val, freq_out); break; } -#endif - return -EINVAL; } @@ -652,10 +643,8 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 = ssp_read_reg(ssp, SSCR0); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: -#ifdef CONFIG_PXA3xx if (cpu_is_pxa3xx()) sscr0 |= SSCR0_FPCKE; -#endif sscr0 |= SSCR0_DataSize(16); break; case SNDRV_PCM_FORMAT_S24_LE: @@ -682,7 +671,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, * needed for that mode are only available on PXA3xx. */ -#ifdef CONFIG_PXA3xx if (!cpu_is_pxa3xx()) return -EINVAL; @@ -691,9 +679,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sspsp |= SSPSP_EDMYSTOP(3); sspsp |= SSPSP_DMYSTOP(3); sspsp |= SSPSP_DMYSTRT(1); -#else - return -EINVAL; -#endif } else { /* The frame width is the width the LRCLK is * asserted for; the delay is expressed in -- 1.6.0.4

3 5

[alsa-devel] AD1981B SoundMax
by Daniel Vogel 23 Apr '09

23 Apr '09

Hello Developers, sorry if this question is off topic on this list, but I don't know where else to send the question to. I'm currently developing an audio-application on an ARM board that is equipped with an AD1981B SoundMax chip. Everything works well except the build-in equalizer. Since it is impossible to get usable technical information from Analog-Device my last hope is to find someone on this list who has some experience with the chip in question. My problem started with the datasheet not telling the format of the fixed point numbers for biquad coefficients. It only states that they are 16 bit numbers with 16 bit resolution... Then, when I use an estimated value for a0 that is unequal from zero and all other coefficients are zero, in theory I should hear some sound when I play a PCM file (44.1kHz / 16Bit) . But there is only silence on the output when the equalizer is not muted. Is there something else I have to set up to make the equalizer output pass through? Looking forward for answers. ;-) Thanks in advance! Daniel GO Engineering GmbH - Stolzenbergstr. 13/IV - 76532 Baden-Baden Geschäftsführer: Helmut Gerstner, Dipl.-Ing. (FH) Ralf Wörner, Dipl.-Ing. (FH) Registergericht: Mannheim HRB 201811

1 0

[alsa-devel] [PATCH v2 1/1] ASoC: TWL4030: Add VDL path support
by Joonyoung Shim 23 Apr '09

23 Apr '09

Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 256 ++++++++++++++++++++++---------------------- 1 files changed, 126 insertions(+), 130 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index cc2968c..fdf88df 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -321,104 +321,60 @@ static void twl4030_power_down(struct snd_soc_codec *codec) } /* Earpiece */ -static const char *twl4030_earpiece_texts[] = - {"Off", "DACL1", "DACL2", "DACR1"}; - -static const unsigned int twl4030_earpiece_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_earpiece_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_earpiece_texts), - twl4030_earpiece_texts, - twl4030_earpiece_values); - -static const struct snd_kcontrol_new twl4030_dapm_earpiece_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_earpiece_enum); +static const struct snd_kcontrol_new twl4030_dapm_earpiece_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_EAR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_EAR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_EAR_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_EAR_CTL, 3, 1, 0), +}; /* PreDrive Left */ -static const char *twl4030_predrivel_texts[] = - {"Off", "DACL1", "DACL2", "DACR2"}; - -static const unsigned int twl4030_predrivel_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_predrivel_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_predrivel_texts), - twl4030_predrivel_texts, - twl4030_predrivel_values); - -static const struct snd_kcontrol_new twl4030_dapm_predrivel_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_predrivel_enum); +static const struct snd_kcontrol_new twl4030_dapm_predrivel_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDL_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PREDL_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDL_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDL_CTL, 3, 1, 0), +}; /* PreDrive Right */ -static const char *twl4030_predriver_texts[] = - {"Off", "DACR1", "DACR2", "DACL2"}; - -static const unsigned int twl4030_predriver_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_predriver_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_predriver_texts), - twl4030_predriver_texts, - twl4030_predriver_values); - -static const struct snd_kcontrol_new twl4030_dapm_predriver_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_predriver_enum); +static const struct snd_kcontrol_new twl4030_dapm_predriver_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PREDR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDR_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDR_CTL, 3, 1, 0), +}; /* Headset Left */ -static const char *twl4030_hsol_texts[] = - {"Off", "DACL1", "DACL2"}; - -static const struct soc_enum twl4030_hsol_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1, - ARRAY_SIZE(twl4030_hsol_texts), - twl4030_hsol_texts); - -static const struct snd_kcontrol_new twl4030_dapm_hsol_control = -SOC_DAPM_ENUM("Route", twl4030_hsol_enum); +static const struct snd_kcontrol_new twl4030_dapm_hsol_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_HS_SEL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_HS_SEL, 2, 1, 0), +}; /* Headset Right */ -static const char *twl4030_hsor_texts[] = - {"Off", "DACR1", "DACR2"}; - -static const struct soc_enum twl4030_hsor_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4, - ARRAY_SIZE(twl4030_hsor_texts), - twl4030_hsor_texts); - -static const struct snd_kcontrol_new twl4030_dapm_hsor_control = -SOC_DAPM_ENUM("Route", twl4030_hsor_enum); +static const struct snd_kcontrol_new twl4030_dapm_hsor_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 3, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_HS_SEL, 4, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_HS_SEL, 5, 1, 0), +}; /* Carkit Left */ -static const char *twl4030_carkitl_texts[] = - {"Off", "DACL1", "DACL2"}; - -static const struct soc_enum twl4030_carkitl_enum = - SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1, - ARRAY_SIZE(twl4030_carkitl_texts), - twl4030_carkitl_texts); - -static const struct snd_kcontrol_new twl4030_dapm_carkitl_control = -SOC_DAPM_ENUM("Route", twl4030_carkitl_enum); +static const struct snd_kcontrol_new twl4030_dapm_carkitl_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKL_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PRECKL_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PRECKL_CTL, 2, 1, 0), +}; /* Carkit Right */ -static const char *twl4030_carkitr_texts[] = - {"Off", "DACR1", "DACR2"}; - -static const struct soc_enum twl4030_carkitr_enum = - SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1, - ARRAY_SIZE(twl4030_carkitr_texts), - twl4030_carkitr_texts); - -static const struct snd_kcontrol_new twl4030_dapm_carkitr_control = -SOC_DAPM_ENUM("Route", twl4030_carkitr_enum); +static const struct snd_kcontrol_new twl4030_dapm_carkitr_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PRECKR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PRECKR_CTL, 2, 1, 0), +}; /* Handsfree Left */ static const char *twl4030_handsfreel_texts[] = - {"Voice", "DACL1", "DACL2", "DACR2"}; + {"Voice", "AudioL1", "AudioL2", "AudioR2"}; static const struct soc_enum twl4030_handsfreel_enum = SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0, @@ -430,7 +386,7 @@ SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum); /* Handsfree Right */ static const char *twl4030_handsfreer_texts[] = - {"Voice", "DACR1", "DACR2", "DACL2"}; + {"Voice", "AudioR1", "AudioR2", "AudioL2"}; static const struct soc_enum twl4030_handsfreer_enum = SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0, @@ -829,6 +785,12 @@ static DECLARE_TLV_DB_SCALE(digital_fine_tlv, -6300, 100, 1); static DECLARE_TLV_DB_SCALE(digital_coarse_tlv, 0, 600, 0); /* + * Voice Downlink GAIN volume control: + * from -37 to 12 dB in 1 dB steps (mute instead of -37 dB) + */ +static DECLARE_TLV_DB_SCALE(digital_voice_downlink_tlv, -3700, 100, 1); + +/* * Analog playback gain * -24 dB to 12 dB in 2 dB steps */ @@ -892,6 +854,16 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL, 1, 1, 0), + /* Common voice downlink gain controls */ + SOC_SINGLE_TLV("DAC Voice Digital Downlink Volume", + TWL4030_REG_VRXPGA, 0, 0x31, 0, digital_voice_downlink_tlv), + + SOC_SINGLE_TLV("DAC Voice Analog Downlink Volume", + TWL4030_REG_VDL_APGA_CTL, 3, 0x12, 1, analog_tlv), + + SOC_SINGLE("DAC Voice Analog Downlink Switch", + TWL4030_REG_VDL_APGA_CTL, 1, 1, 0), + /* Separate output gain controls */ SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume", TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL, @@ -956,6 +928,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", + TWL4030_REG_AVDAC_CTL, 4, 0), /* Analog PGAs */ SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, @@ -966,6 +940,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { 0, 0, NULL, 0), SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("VDL_APGA", TWL4030_REG_VDL_APGA_CTL, + 0, 0, NULL, 0), /* Analog bypasses */ SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, @@ -998,26 +974,35 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0), - /* Output MUX controls */ + /* Output MIXER controls */ /* Earpiece */ - SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_earpiece_control), + SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_earpiece_controls[0], + ARRAY_SIZE(twl4030_dapm_earpiece_controls)), /* PreDrivL/R */ - SND_SOC_DAPM_VALUE_MUX("PredriveL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_predrivel_control), - SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_predriver_control), + SND_SOC_DAPM_MIXER("PredriveL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predrivel_controls[0], + ARRAY_SIZE(twl4030_dapm_predrivel_controls)), + SND_SOC_DAPM_MIXER("PredriveR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predriver_controls[0], + ARRAY_SIZE(twl4030_dapm_predriver_controls)), /* HeadsetL/R */ - SND_SOC_DAPM_MUX_E("HeadsetL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_hsol_control, headsetl_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_hsor_control), + SND_SOC_DAPM_MIXER_E("HeadsetL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsol_controls[0], + ARRAY_SIZE(twl4030_dapm_hsol_controls), headsetl_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER("HeadsetR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsor_controls[0], + ARRAY_SIZE(twl4030_dapm_hsor_controls)), /* CarkitL/R */ - SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_carkitl_control), - SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_carkitr_control), + SND_SOC_DAPM_MIXER("CarkitL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitl_controls[0], + ARRAY_SIZE(twl4030_dapm_carkitl_controls)), + SND_SOC_DAPM_MIXER("CarkitR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitr_controls[0], + ARRAY_SIZE(twl4030_dapm_carkitr_controls)), + + /* Output MUX controls */ /* HandsfreeL/R */ SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, &twl4030_dapm_handsfreel_control, handsfree_event, @@ -1082,50 +1067,61 @@ static const struct snd_soc_dapm_route intercon[] = { {"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"}, {"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"}, + {"VDL_APGA", NULL, "DAC Voice"}, + /* Internal playback routings */ /* Earpiece */ - {"Earpiece Mux", "DACL1", "ARXL1_APGA"}, - {"Earpiece Mux", "DACL2", "ARXL2_APGA"}, - {"Earpiece Mux", "DACR1", "ARXR1_APGA"}, + {"Earpiece Mixer", "Voice", "VDL_APGA"}, + {"Earpiece Mixer", "AudioL1", "ARXL1_APGA"}, + {"Earpiece Mixer", "AudioL2", "ARXL2_APGA"}, + {"Earpiece Mixer", "AudioR1", "ARXR1_APGA"}, /* PreDrivL */ - {"PredriveL Mux", "DACL1", "ARXL1_APGA"}, - {"PredriveL Mux", "DACL2", "ARXL2_APGA"}, - {"PredriveL Mux", "DACR2", "ARXR2_APGA"}, + {"PredriveL Mixer", "Voice", "VDL_APGA"}, + {"PredriveL Mixer", "AudioL1", "ARXL1_APGA"}, + {"PredriveL Mixer", "AudioL2", "ARXL2_APGA"}, + {"PredriveL Mixer", "AudioR2", "ARXR2_APGA"}, /* PreDrivR */ - {"PredriveR Mux", "DACR1", "ARXR1_APGA"}, - {"PredriveR Mux", "DACR2", "ARXR2_APGA"}, - {"PredriveR Mux", "DACL2", "ARXL2_APGA"}, + {"PredriveR Mixer", "Voice", "VDL_APGA"}, + {"PredriveR Mixer", "AudioR1", "ARXR1_APGA"}, + {"PredriveR Mixer", "AudioR2", "ARXR2_APGA"}, + {"PredriveR Mixer", "AudioL2", "ARXL2_APGA"}, /* HeadsetL */ - {"HeadsetL Mux", "DACL1", "ARXL1_APGA"}, - {"HeadsetL Mux", "DACL2", "ARXL2_APGA"}, + {"HeadsetL Mixer", "Voice", "VDL_APGA"}, + {"HeadsetL Mixer", "AudioL1", "ARXL1_APGA"}, + {"HeadsetL Mixer", "AudioL2", "ARXL2_APGA"}, /* HeadsetR */ - {"HeadsetR Mux", "DACR1", "ARXR1_APGA"}, - {"HeadsetR Mux", "DACR2", "ARXR2_APGA"}, + {"HeadsetR Mixer", "Voice", "VDL_APGA"}, + {"HeadsetR Mixer", "AudioR1", "ARXR1_APGA"}, + {"HeadsetR Mixer", "AudioR2", "ARXR2_APGA"}, /* CarkitL */ - {"CarkitL Mux", "DACL1", "ARXL1_APGA"}, - {"CarkitL Mux", "DACL2", "ARXL2_APGA"}, + {"CarkitL Mixer", "Voice", "VDL_APGA"}, + {"CarkitL Mixer", "AudioL1", "ARXL1_APGA"}, + {"CarkitL Mixer", "AudioL2", "ARXL2_APGA"}, /* CarkitR */ - {"CarkitR Mux", "DACR1", "ARXR1_APGA"}, - {"CarkitR Mux", "DACR2", "ARXR2_APGA"}, + {"CarkitR Mixer", "Voice", "VDL_APGA"}, + {"CarkitR Mixer", "AudioR1", "ARXR1_APGA"}, + {"CarkitR Mixer", "AudioR2", "ARXR2_APGA"}, /* HandsfreeL */ - {"HandsfreeL Mux", "DACL1", "ARXL1_APGA"}, - {"HandsfreeL Mux", "DACL2", "ARXL2_APGA"}, - {"HandsfreeL Mux", "DACR2", "ARXR2_APGA"}, + {"HandsfreeL Mux", "Voice", "VDL_APGA"}, + {"HandsfreeL Mux", "AudioL1", "ARXL1_APGA"}, + {"HandsfreeL Mux", "AudioL2", "ARXL2_APGA"}, + {"HandsfreeL Mux", "AudioR2", "ARXR2_APGA"}, /* HandsfreeR */ - {"HandsfreeR Mux", "DACR1", "ARXR1_APGA"}, - {"HandsfreeR Mux", "DACR2", "ARXR2_APGA"}, - {"HandsfreeR Mux", "DACL2", "ARXL2_APGA"}, + {"HandsfreeR Mux", "Voice", "VDL_APGA"}, + {"HandsfreeR Mux", "AudioR1", "ARXR1_APGA"}, + {"HandsfreeR Mux", "AudioR2", "ARXR2_APGA"}, + {"HandsfreeR Mux", "AudioL2", "ARXL2_APGA"}, /* outputs */ {"OUTL", NULL, "ARXL2_APGA"}, {"OUTR", NULL, "ARXR2_APGA"}, - {"EARPIECE", NULL, "Earpiece Mux"}, - {"PREDRIVEL", NULL, "PredriveL Mux"}, - {"PREDRIVER", NULL, "PredriveR Mux"}, - {"HSOL", NULL, "HeadsetL Mux"}, - {"HSOR", NULL, "HeadsetR Mux"}, - {"CARKITL", NULL, "CarkitL Mux"}, - {"CARKITR", NULL, "CarkitR Mux"}, + {"EARPIECE", NULL, "Earpiece Mixer"}, + {"PREDRIVEL", NULL, "PredriveL Mixer"}, + {"PREDRIVER", NULL, "PredriveR Mixer"}, + {"HSOL", NULL, "HeadsetL Mixer"}, + {"HSOR", NULL, "HeadsetR Mixer"}, + {"CARKITL", NULL, "CarkitL Mixer"}, + {"CARKITR", NULL, "CarkitR Mixer"}, {"HFL", NULL, "HandsfreeL Mux"}, {"HFR", NULL, "HandsfreeR Mux"}, -- 1.5.6.3

3 2

[alsa-devel] [RFC PATCH 2/5] pxa: remove deprecated corgi_{ssp, lcd}.c
by Eric Miao 23 Apr '09

23 Apr '09

Signed-off-by: Eric Miao <eric.miao(a)marvell.com> --- arch/arm/mach-pxa/Kconfig | 8 - arch/arm/mach-pxa/Makefile | 1 - arch/arm/mach-pxa/corgi_lcd.c | 288 ---------------------------------------- arch/arm/mach-pxa/corgi_ssp.c | 275 -------------------------------------- arch/arm/mach-pxa/sharpsl_pm.c | 9 +- 5 files changed, 2 insertions(+), 579 deletions(-) delete mode 100644 arch/arm/mach-pxa/corgi_lcd.c delete mode 100644 arch/arm/mach-pxa/corgi_ssp.c diff --git a/arch/arm/mach-pxa/Kconfig b/arch/arm/mach-pxa/Kconfig index a225658..d52d8a3 100644 --- a/arch/arm/mach-pxa/Kconfig +++ b/arch/arm/mach-pxa/Kconfig @@ -88,14 +88,6 @@ config PXA_SHARPSL SL-C3000 (Spitz), SL-C3100 (Borzoi) or SL-C6000x (Tosa) handheld computer. -config CORGI_SSP_DEPRECATED - bool - select PXA_SSP - help - This option will include corgi_ssp.c and corgi_lcd.c - that corgi_ts.c and other legacy drivers (corgi_bl.c - and sharpsl_pm.c) may depend on. - config MACH_POODLE bool "Enable Sharp SL-5600 (Poodle) Support" depends on PXA_SHARPSL diff --git a/arch/arm/mach-pxa/Makefile b/arch/arm/mach-pxa/Makefile index 4bcebf7..eb89363 100644 --- a/arch/arm/mach-pxa/Makefile +++ b/arch/arm/mach-pxa/Makefile @@ -41,7 +41,6 @@ obj-$(CONFIG_MACH_COLIBRI320) += colibri-pxa3xx.o colibri-pxa320.o obj-$(CONFIG_MACH_H5000) += h5000.o obj-$(CONFIG_PXA_SHARP_C7xx) += corgi.o sharpsl_pm.o corgi_pm.o obj-$(CONFIG_PXA_SHARP_Cxx00) += spitz.o sharpsl_pm.o spitz_pm.o -obj-$(CONFIG_CORGI_SSP_DEPRECATED) += corgi_ssp.o corgi_lcd.o obj-$(CONFIG_MACH_POODLE) += poodle.o obj-$(CONFIG_MACH_PCM027) += pcm027.o obj-$(CONFIG_MACH_PCM990_BASEBOARD) += pcm990-baseboard.o diff --git a/arch/arm/mach-pxa/corgi_lcd.c b/arch/arm/mach-pxa/corgi_lcd.c deleted file mode 100644 index d9b9631..0000000 --- a/arch/arm/mach-pxa/corgi_lcd.c +++ /dev/null @@ -1,288 +0,0 @@ -/* - * linux/arch/arm/mach-pxa/corgi_lcd.c - * - * Corgi/Spitz LCD Specific Code - * - * Copyright (C) 2005 Richard Purdie - * - * Connectivity: - * Corgi - LCD to ATI Imageon w100 (Wallaby) - * Spitz - LCD to PXA Framebuffer - * - * 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. - * - */ - -#include <linux/delay.h> -#include <linux/kernel.h> -#include <linux/platform_device.h> -#include <linux/module.h> -#include <linux/string.h> -#include <mach/corgi.h> -#include <mach/hardware.h> -#include <mach/sharpsl.h> -#include <mach/spitz.h> -#include <asm/hardware/scoop.h> -#include <asm/mach/sharpsl_param.h> -#include "generic.h" - -/* Register Addresses */ -#define RESCTL_ADRS 0x00 -#define PHACTRL_ADRS 0x01 -#define DUTYCTRL_ADRS 0x02 -#define POWERREG0_ADRS 0x03 -#define POWERREG1_ADRS 0x04 -#define GPOR3_ADRS 0x05 -#define PICTRL_ADRS 0x06 -#define POLCTRL_ADRS 0x07 - -/* Register Bit Definitions */ -#define RESCTL_QVGA 0x01 -#define RESCTL_VGA 0x00 - -#define POWER1_VW_ON 0x01 /* VW Supply FET ON */ -#define POWER1_GVSS_ON 0x02 /* GVSS(-8V) Power Supply ON */ -#define POWER1_VDD_ON 0x04 /* VDD(8V),SVSS(-4V) Power Supply ON */ - -#define POWER1_VW_OFF 0x00 /* VW Supply FET OFF */ -#define POWER1_GVSS_OFF 0x00 /* GVSS(-8V) Power Supply OFF */ -#define POWER1_VDD_OFF 0x00 /* VDD(8V),SVSS(-4V) Power Supply OFF */ - -#define POWER0_COM_DCLK 0x01 /* COM Voltage DC Bias DAC Serial Data Clock */ -#define POWER0_COM_DOUT 0x02 /* COM Voltage DC Bias DAC Serial Data Out */ -#define POWER0_DAC_ON 0x04 /* DAC Power Supply ON */ -#define POWER0_COM_ON 0x08 /* COM Power Supply ON */ -#define POWER0_VCC5_ON 0x10 /* VCC5 Power Supply ON */ - -#define POWER0_DAC_OFF 0x00 /* DAC Power Supply OFF */ -#define POWER0_COM_OFF 0x00 /* COM Power Supply OFF */ -#define POWER0_VCC5_OFF 0x00 /* VCC5 Power Supply OFF */ - -#define PICTRL_INIT_STATE 0x01 -#define PICTRL_INIOFF 0x02 -#define PICTRL_POWER_DOWN 0x04 -#define PICTRL_COM_SIGNAL_OFF 0x08 -#define PICTRL_DAC_SIGNAL_OFF 0x10 - -#define POLCTRL_SYNC_POL_FALL 0x01 -#define POLCTRL_EN_POL_FALL 0x02 -#define POLCTRL_DATA_POL_FALL 0x04 -#define POLCTRL_SYNC_ACT_H 0x08 -#define POLCTRL_EN_ACT_L 0x10 - -#define POLCTRL_SYNC_POL_RISE 0x00 -#define POLCTRL_EN_POL_RISE 0x00 -#define POLCTRL_DATA_POL_RISE 0x00 -#define POLCTRL_SYNC_ACT_L 0x00 -#define POLCTRL_EN_ACT_H 0x00 - -#define PHACTRL_PHASE_MANUAL 0x01 -#define DEFAULT_PHAD_QVGA (9) -#define DEFAULT_COMADJ (125) - -/* - * This is only a psuedo I2C interface. We can't use the standard kernel - * routines as the interface is write only. We just assume the data is acked... - */ -static void lcdtg_ssp_i2c_send(u8 data) -{ - corgi_ssp_lcdtg_send(POWERREG0_ADRS, data); - udelay(10); -} - -static void lcdtg_i2c_send_bit(u8 data) -{ - lcdtg_ssp_i2c_send(data); - lcdtg_ssp_i2c_send(data | POWER0_COM_DCLK); - lcdtg_ssp_i2c_send(data); -} - -static void lcdtg_i2c_send_start(u8 base) -{ - lcdtg_ssp_i2c_send(base | POWER0_COM_DCLK | POWER0_COM_DOUT); - lcdtg_ssp_i2c_send(base | POWER0_COM_DCLK); - lcdtg_ssp_i2c_send(base); -} - -static void lcdtg_i2c_send_stop(u8 base) -{ - lcdtg_ssp_i2c_send(base); - lcdtg_ssp_i2c_send(base | POWER0_COM_DCLK); - lcdtg_ssp_i2c_send(base | POWER0_COM_DCLK | POWER0_COM_DOUT); -} - -static void lcdtg_i2c_send_byte(u8 base, u8 data) -{ - int i; - for (i = 0; i < 8; i++) { - if (data & 0x80) - lcdtg_i2c_send_bit(base | POWER0_COM_DOUT); - else - lcdtg_i2c_send_bit(base); - data <<= 1; - } -} - -static void lcdtg_i2c_wait_ack(u8 base) -{ - lcdtg_i2c_send_bit(base); -} - -static void lcdtg_set_common_voltage(u8 base_data, u8 data) -{ - /* Set Common Voltage to M62332FP via I2C */ - lcdtg_i2c_send_start(base_data); - lcdtg_i2c_send_byte(base_data, 0x9c); - lcdtg_i2c_wait_ack(base_data); - lcdtg_i2c_send_byte(base_data, 0x00); - lcdtg_i2c_wait_ack(base_data); - lcdtg_i2c_send_byte(base_data, data); - lcdtg_i2c_wait_ack(base_data); - lcdtg_i2c_send_stop(base_data); -} - -/* Set Phase Adjust */ -static void lcdtg_set_phadadj(int mode) -{ - int adj; - switch(mode) { - case 480: - case 640: - /* Setting for VGA */ - adj = sharpsl_param.phadadj; - if (adj < 0) { - adj = PHACTRL_PHASE_MANUAL; - } else { - adj = ((adj & 0x0f) << 1) | PHACTRL_PHASE_MANUAL; - } - break; - case 240: - case 320: - default: - /* Setting for QVGA */ - adj = (DEFAULT_PHAD_QVGA << 1) | PHACTRL_PHASE_MANUAL; - break; - } - - corgi_ssp_lcdtg_send(PHACTRL_ADRS, adj); -} - -static int lcd_inited; - -void corgi_lcdtg_hw_init(int mode) -{ - if (!lcd_inited) { - int comadj; - - /* Initialize Internal Logic & Port */ - corgi_ssp_lcdtg_send(PICTRL_ADRS, PICTRL_POWER_DOWN | PICTRL_INIOFF | PICTRL_INIT_STATE - | PICTRL_COM_SIGNAL_OFF | PICTRL_DAC_SIGNAL_OFF); - - corgi_ssp_lcdtg_send(POWERREG0_ADRS, POWER0_COM_DCLK | POWER0_COM_DOUT | POWER0_DAC_OFF - | POWER0_COM_OFF | POWER0_VCC5_OFF); - - corgi_ssp_lcdtg_send(POWERREG1_ADRS, POWER1_VW_OFF | POWER1_GVSS_OFF | POWER1_VDD_OFF); - - /* VDD(+8V), SVSS(-4V) ON */ - corgi_ssp_lcdtg_send(POWERREG1_ADRS, POWER1_VW_OFF | POWER1_GVSS_OFF | POWER1_VDD_ON); - mdelay(3); - - /* DAC ON */ - corgi_ssp_lcdtg_send(POWERREG0_ADRS, POWER0_COM_DCLK | POWER0_COM_DOUT | POWER0_DAC_ON - | POWER0_COM_OFF | POWER0_VCC5_OFF); - - /* INIB = H, INI = L */ - /* PICTL[0] = H , PICTL[1] = PICTL[2] = PICTL[4] = L */ - corgi_ssp_lcdtg_send(PICTRL_ADRS, PICTRL_INIT_STATE | PICTRL_COM_SIGNAL_OFF); - - /* Set Common Voltage */ - comadj = sharpsl_param.comadj; - if (comadj < 0) - comadj = DEFAULT_COMADJ; - lcdtg_set_common_voltage((POWER0_DAC_ON | POWER0_COM_OFF | POWER0_VCC5_OFF), comadj); - - /* VCC5 ON, DAC ON */ - corgi_ssp_lcdtg_send(POWERREG0_ADRS, POWER0_COM_DCLK | POWER0_COM_DOUT | POWER0_DAC_ON | - POWER0_COM_OFF | POWER0_VCC5_ON); - - /* GVSS(-8V) ON, VDD ON */ - corgi_ssp_lcdtg_send(POWERREG1_ADRS, POWER1_VW_OFF | POWER1_GVSS_ON | POWER1_VDD_ON); - mdelay(2); - - /* COM SIGNAL ON (PICTL[3] = L) */ - corgi_ssp_lcdtg_send(PICTRL_ADRS, PICTRL_INIT_STATE); - - /* COM ON, DAC ON, VCC5_ON */ - corgi_ssp_lcdtg_send(POWERREG0_ADRS, POWER0_COM_DCLK | POWER0_COM_DOUT | POWER0_DAC_ON - | POWER0_COM_ON | POWER0_VCC5_ON); - - /* VW ON, GVSS ON, VDD ON */ - corgi_ssp_lcdtg_send(POWERREG1_ADRS, POWER1_VW_ON | POWER1_GVSS_ON | POWER1_VDD_ON); - - /* Signals output enable */ - corgi_ssp_lcdtg_send(PICTRL_ADRS, 0); - - /* Set Phase Adjust */ - lcdtg_set_phadadj(mode); - - /* Initialize for Input Signals from ATI */ - corgi_ssp_lcdtg_send(POLCTRL_ADRS, POLCTRL_SYNC_POL_RISE | POLCTRL_EN_POL_RISE - | POLCTRL_DATA_POL_RISE | POLCTRL_SYNC_ACT_L | POLCTRL_EN_ACT_H); - udelay(1000); - - lcd_inited=1; - } else { - lcdtg_set_phadadj(mode); - } - - switch(mode) { - case 480: - case 640: - /* Set Lcd Resolution (VGA) */ - corgi_ssp_lcdtg_send(RESCTL_ADRS, RESCTL_VGA); - break; - case 240: - case 320: - default: - /* Set Lcd Resolution (QVGA) */ - corgi_ssp_lcdtg_send(RESCTL_ADRS, RESCTL_QVGA); - break; - } -} - -void corgi_lcdtg_suspend(void) -{ - /* 60Hz x 2 frame = 16.7msec x 2 = 33.4 msec */ - mdelay(34); - - /* (1)VW OFF */ - corgi_ssp_lcdtg_send(POWERREG1_ADRS, POWER1_VW_OFF | POWER1_GVSS_ON | POWER1_VDD_ON); - - /* (2)COM OFF */ - corgi_ssp_lcdtg_send(PICTRL_ADRS, PICTRL_COM_SIGNAL_OFF); - corgi_ssp_lcdtg_send(POWERREG0_ADRS, POWER0_DAC_ON | POWER0_COM_OFF | POWER0_VCC5_ON); - - /* (3)Set Common Voltage Bias 0V */ - lcdtg_set_common_voltage(POWER0_DAC_ON | POWER0_COM_OFF | POWER0_VCC5_ON, 0); - - /* (4)GVSS OFF */ - corgi_ssp_lcdtg_send(POWERREG1_ADRS, POWER1_VW_OFF | POWER1_GVSS_OFF | POWER1_VDD_ON); - - /* (5)VCC5 OFF */ - corgi_ssp_lcdtg_send(POWERREG0_ADRS, POWER0_DAC_ON | POWER0_COM_OFF | POWER0_VCC5_OFF); - - /* (6)Set PDWN, INIOFF, DACOFF */ - corgi_ssp_lcdtg_send(PICTRL_ADRS, PICTRL_INIOFF | PICTRL_DAC_SIGNAL_OFF | - PICTRL_POWER_DOWN | PICTRL_COM_SIGNAL_OFF); - - /* (7)DAC OFF */ - corgi_ssp_lcdtg_send(POWERREG0_ADRS, POWER0_DAC_OFF | POWER0_COM_OFF | POWER0_VCC5_OFF); - - /* (8)VDD OFF */ - corgi_ssp_lcdtg_send(POWERREG1_ADRS, POWER1_VW_OFF | POWER1_GVSS_OFF | POWER1_VDD_OFF); - - lcd_inited = 0; -} - diff --git a/arch/arm/mach-pxa/corgi_ssp.c b/arch/arm/mach-pxa/corgi_ssp.c deleted file mode 100644 index a5ee707..0000000 --- a/arch/arm/mach-pxa/corgi_ssp.c +++ /dev/null @@ -1,275 +0,0 @@ -/* - * SSP control code for Sharp Corgi devices - * - * Copyright (c) 2004-2005 Richard Purdie - * - * 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. - * - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/kernel.h> -#include <linux/sched.h> -#include <linux/slab.h> -#include <linux/delay.h> -#include <linux/platform_device.h> -#include <mach/hardware.h> -#include <asm/mach-types.h> - -#include <mach/ssp.h> -#include <mach/pxa2xx-gpio.h> -#include <mach/regs-ssp.h> -#include "sharpsl.h" - -static DEFINE_SPINLOCK(corgi_ssp_lock); -static struct ssp_dev corgi_ssp_dev; -static struct ssp_state corgi_ssp_state; -static struct corgissp_machinfo *ssp_machinfo; - -/* - * There are three devices connected to the SSP interface: - * 1. A touchscreen controller (TI ADS7846 compatible) - * 2. An LCD controller (with some Backlight functionality) - * 3. A battery monitoring IC (Maxim MAX1111) - * - * Each device uses a different speed/mode of communication. - * - * The touchscreen is very sensitive and the most frequently used - * so the port is left configured for this. - * - * Devices are selected using Chip Selects on GPIOs. - */ - -/* - * ADS7846 Routines - */ -unsigned long corgi_ssp_ads7846_putget(ulong data) -{ - unsigned long flag; - u32 ret = 0; - - spin_lock_irqsave(&corgi_ssp_lock, flag); - if (ssp_machinfo->cs_ads7846 >= 0) - GPCR(ssp_machinfo->cs_ads7846) = GPIO_bit(ssp_machinfo->cs_ads7846); - - ssp_write_word(&corgi_ssp_dev,data); - ssp_read_word(&corgi_ssp_dev, &ret); - - if (ssp_machinfo->cs_ads7846 >= 0) - GPSR(ssp_machinfo->cs_ads7846) = GPIO_bit(ssp_machinfo->cs_ads7846); - spin_unlock_irqrestore(&corgi_ssp_lock, flag); - - return ret; -} - -/* - * NOTE: These functions should always be called in interrupt context - * and use the _lock and _unlock functions. They are very time sensitive. - */ -void corgi_ssp_ads7846_lock(void) -{ - spin_lock(&corgi_ssp_lock); - if (ssp_machinfo->cs_ads7846 >= 0) - GPCR(ssp_machinfo->cs_ads7846) = GPIO_bit(ssp_machinfo->cs_ads7846); -} - -void corgi_ssp_ads7846_unlock(void) -{ - if (ssp_machinfo->cs_ads7846 >= 0) - GPSR(ssp_machinfo->cs_ads7846) = GPIO_bit(ssp_machinfo->cs_ads7846); - spin_unlock(&corgi_ssp_lock); -} - -void corgi_ssp_ads7846_put(ulong data) -{ - ssp_write_word(&corgi_ssp_dev,data); -} - -unsigned long corgi_ssp_ads7846_get(void) -{ - u32 ret = 0; - ssp_read_word(&corgi_ssp_dev, &ret); - return ret; -} - -EXPORT_SYMBOL(corgi_ssp_ads7846_putget); -EXPORT_SYMBOL(corgi_ssp_ads7846_lock); -EXPORT_SYMBOL(corgi_ssp_ads7846_unlock); -EXPORT_SYMBOL(corgi_ssp_ads7846_put); -EXPORT_SYMBOL(corgi_ssp_ads7846_get); - - -/* - * LCD/Backlight Routines - */ -unsigned long corgi_ssp_dac_put(ulong data) -{ - unsigned long flag, sscr1 = SSCR1_SPH; - u32 tmp; - - spin_lock_irqsave(&corgi_ssp_lock, flag); - - if (machine_is_spitz() || machine_is_akita() || machine_is_borzoi()) - sscr1 = 0; - - ssp_disable(&corgi_ssp_dev); - ssp_config(&corgi_ssp_dev, (SSCR0_Motorola | (SSCR0_DSS & 0x07 )), sscr1, 0, SSCR0_SerClkDiv(ssp_machinfo->clk_lcdcon)); - ssp_enable(&corgi_ssp_dev); - - if (ssp_machinfo->cs_lcdcon >= 0) - GPCR(ssp_machinfo->cs_lcdcon) = GPIO_bit(ssp_machinfo->cs_lcdcon); - ssp_write_word(&corgi_ssp_dev,data); - /* Read null data back from device to prevent SSP overflow */ - ssp_read_word(&corgi_ssp_dev, &tmp); - if (ssp_machinfo->cs_lcdcon >= 0) - GPSR(ssp_machinfo->cs_lcdcon) = GPIO_bit(ssp_machinfo->cs_lcdcon); - - ssp_disable(&corgi_ssp_dev); - ssp_config(&corgi_ssp_dev, (SSCR0_National | (SSCR0_DSS & 0x0b )), 0, 0, SSCR0_SerClkDiv(ssp_machinfo->clk_ads7846)); - ssp_enable(&corgi_ssp_dev); - - spin_unlock_irqrestore(&corgi_ssp_lock, flag); - - return 0; -} - -void corgi_ssp_lcdtg_send(u8 adrs, u8 data) -{ - corgi_ssp_dac_put(((adrs & 0x07) << 5) | (data & 0x1f)); -} - -void corgi_ssp_blduty_set(int duty) -{ - corgi_ssp_lcdtg_send(0x02,duty); -} - -EXPORT_SYMBOL(corgi_ssp_lcdtg_send); -EXPORT_SYMBOL(corgi_ssp_blduty_set); - -/* - * Max1111 Routines - */ -int corgi_ssp_max1111_get(ulong data) -{ - unsigned long flag; - long voltage = 0, voltage1 = 0, voltage2 = 0; - - spin_lock_irqsave(&corgi_ssp_lock, flag); - if (ssp_machinfo->cs_max1111 >= 0) - GPCR(ssp_machinfo->cs_max1111) = GPIO_bit(ssp_machinfo->cs_max1111); - ssp_disable(&corgi_ssp_dev); - ssp_config(&corgi_ssp_dev, (SSCR0_Motorola | (SSCR0_DSS & 0x07 )), 0, 0, SSCR0_SerClkDiv(ssp_machinfo->clk_max1111)); - ssp_enable(&corgi_ssp_dev); - - udelay(1); - - /* TB1/RB1 */ - ssp_write_word(&corgi_ssp_dev,data); - ssp_read_word(&corgi_ssp_dev, (u32*)&voltage1); /* null read */ - - /* TB12/RB2 */ - ssp_write_word(&corgi_ssp_dev,0); - ssp_read_word(&corgi_ssp_dev, (u32*)&voltage1); - - /* TB13/RB3*/ - ssp_write_word(&corgi_ssp_dev,0); - ssp_read_word(&corgi_ssp_dev, (u32*)&voltage2); - - ssp_disable(&corgi_ssp_dev); - ssp_config(&corgi_ssp_dev, (SSCR0_National | (SSCR0_DSS & 0x0b )), 0, 0, SSCR0_SerClkDiv(ssp_machinfo->clk_ads7846)); - ssp_enable(&corgi_ssp_dev); - if (ssp_machinfo->cs_max1111 >= 0) - GPSR(ssp_machinfo->cs_max1111) = GPIO_bit(ssp_machinfo->cs_max1111); - spin_unlock_irqrestore(&corgi_ssp_lock, flag); - - if (voltage1 & 0xc0 || voltage2 & 0x3f) - voltage = -1; - else - voltage = ((voltage1 << 2) & 0xfc) | ((voltage2 >> 6) & 0x03); - - return voltage; -} - -EXPORT_SYMBOL(corgi_ssp_max1111_get); - -/* - * Support Routines - */ - -void __init corgi_ssp_set_machinfo(struct corgissp_machinfo *machinfo) -{ - ssp_machinfo = machinfo; -} - -static int __init corgi_ssp_probe(struct platform_device *dev) -{ - int ret; - - /* Chip Select - Disable All */ - if (ssp_machinfo->cs_lcdcon >= 0) - pxa_gpio_mode(ssp_machinfo->cs_lcdcon | GPIO_OUT | GPIO_DFLT_HIGH); - if (ssp_machinfo->cs_max1111 >= 0) - pxa_gpio_mode(ssp_machinfo->cs_max1111 | GPIO_OUT | GPIO_DFLT_HIGH); - if (ssp_machinfo->cs_ads7846 >= 0) - pxa_gpio_mode(ssp_machinfo->cs_ads7846 | GPIO_OUT | GPIO_DFLT_HIGH); - - ret = ssp_init(&corgi_ssp_dev, ssp_machinfo->port, 0); - - if (ret) - printk(KERN_ERR "Unable to register SSP handler!\n"); - else { - ssp_disable(&corgi_ssp_dev); - ssp_config(&corgi_ssp_dev, (SSCR0_National | (SSCR0_DSS & 0x0b )), 0, 0, SSCR0_SerClkDiv(ssp_machinfo->clk_ads7846)); - ssp_enable(&corgi_ssp_dev); - } - - return ret; -} - -static int corgi_ssp_remove(struct platform_device *dev) -{ - ssp_exit(&corgi_ssp_dev); - return 0; -} - -static int corgi_ssp_suspend(struct platform_device *dev, pm_message_t state) -{ - ssp_flush(&corgi_ssp_dev); - ssp_save_state(&corgi_ssp_dev,&corgi_ssp_state); - - return 0; -} - -static int corgi_ssp_resume(struct platform_device *dev) -{ - if (ssp_machinfo->cs_lcdcon >= 0) - GPSR(ssp_machinfo->cs_lcdcon) = GPIO_bit(ssp_machinfo->cs_lcdcon); /* High - Disable LCD Control/Timing Gen */ - if (ssp_machinfo->cs_max1111 >= 0) - GPSR(ssp_machinfo->cs_max1111) = GPIO_bit(ssp_machinfo->cs_max1111); /* High - Disable MAX1111*/ - if (ssp_machinfo->cs_ads7846 >= 0) - GPSR(ssp_machinfo->cs_ads7846) = GPIO_bit(ssp_machinfo->cs_ads7846); /* High - Disable ADS7846*/ - ssp_restore_state(&corgi_ssp_dev,&corgi_ssp_state); - ssp_enable(&corgi_ssp_dev); - - return 0; -} - -static struct platform_driver corgissp_driver = { - .probe = corgi_ssp_probe, - .remove = corgi_ssp_remove, - .suspend = corgi_ssp_suspend, - .resume = corgi_ssp_resume, - .driver = { - .name = "corgi-ssp", - }, -}; - -int __init corgi_ssp_init(void) -{ - return platform_driver_register(&corgissp_driver); -} - -arch_initcall(corgi_ssp_init); diff --git a/arch/arm/mach-pxa/sharpsl_pm.c b/arch/arm/mach-pxa/sharpsl_pm.c index 16b4ec6..007ff5e 100644 --- a/arch/arm/mach-pxa/sharpsl_pm.c +++ b/arch/arm/mach-pxa/sharpsl_pm.c @@ -126,22 +126,17 @@ struct battery_thresh spitz_battery_levels_noac[] = { /* * Read MAX1111 ADC */ +extern int max1111_read_channel(int); + int sharpsl_pm_pxa_read_max1111(int channel) { if (machine_is_tosa()) // Ugly, better move this function into another module return 0; -#ifdef CONFIG_CORGI_SSP_DEPRECATED - return corgi_ssp_max1111_get((channel << MAXCTRL_SEL_SH) | MAXCTRL_PD0 | MAXCTRL_PD1 - | MAXCTRL_SGL | MAXCTRL_UNI | MAXCTRL_STR); -#else - extern int max1111_read_channel(int); - /* max1111 accepts channels from 0-3, however, * it is encoded from 0-7 here in the code. */ return max1111_read_channel(channel >> 1); -#endif } void sharpsl_pm_pxa_init(void) -- 1.6.0.4

1 0

[alsa-devel] [RFC PATCH 1/5] pxa: deprecate and remove corgi_ts.c, use ads7846 instead
by Eric Miao 23 Apr '09

23 Apr '09

Signed-off-by: Eric Miao <eric.miao(a)marvell.com> --- drivers/input/touchscreen/Kconfig | 17 -- drivers/input/touchscreen/Makefile | 1 - drivers/input/touchscreen/corgi_ts.c | 385 ---------------------------------- 3 files changed, 0 insertions(+), 403 deletions(-) delete mode 100644 drivers/input/touchscreen/corgi_ts.c diff --git a/drivers/input/touchscreen/Kconfig b/drivers/input/touchscreen/Kconfig index bb6486a..b513845 100644 --- a/drivers/input/touchscreen/Kconfig +++ b/drivers/input/touchscreen/Kconfig @@ -41,23 +41,6 @@ config TOUCHSCREEN_BITSY To compile this driver as a module, choose M here: the module will be called h3600_ts_input. -config TOUCHSCREEN_CORGI - tristate "SharpSL (Corgi and Spitz series) touchscreen driver (DEPRECATED)" - depends on PXA_SHARPSL - select CORGI_SSP_DEPRECATED - default y - help - Say Y here to enable the driver for the touchscreen on the - Sharp SL-C7xx and SL-Cxx00 series of PDAs. - - If unsure, say N. - - To compile this driver as a module, choose M here: the - module will be called corgi_ts. - - NOTE: this driver is deprecated, try enable SPI and generic - ADS7846-based touchscreen driver. - config TOUCHSCREEN_DA9034 tristate "Touchscreen support for Dialog Semiconductor DA9034" depends on PMIC_DA903X diff --git a/drivers/input/touchscreen/Makefile b/drivers/input/touchscreen/Makefile index d3375af..f7e6d2d 100644 --- a/drivers/input/touchscreen/Makefile +++ b/drivers/input/touchscreen/Makefile @@ -9,7 +9,6 @@ wm97xx-ts-y := wm97xx-core.o obj-$(CONFIG_TOUCHSCREEN_ADS7846) += ads7846.o obj-$(CONFIG_TOUCHSCREEN_ATMEL_TSADCC) += atmel_tsadcc.o obj-$(CONFIG_TOUCHSCREEN_BITSY) += h3600_ts_input.o -obj-$(CONFIG_TOUCHSCREEN_CORGI) += corgi_ts.o obj-$(CONFIG_TOUCHSCREEN_GUNZE) += gunze.o obj-$(CONFIG_TOUCHSCREEN_ELO) += elo.o obj-$(CONFIG_TOUCHSCREEN_FUJITSU) += fujitsu_ts.o diff --git a/drivers/input/touchscreen/corgi_ts.c b/drivers/input/touchscreen/corgi_ts.c deleted file mode 100644 index 94a1919..0000000 --- a/drivers/input/touchscreen/corgi_ts.c +++ /dev/null @@ -1,385 +0,0 @@ -/* - * Touchscreen driver for Sharp SL-C7xx and SL-Cxx00 models - * - * Copyright (c) 2004-2005 Richard Purdie - * - * 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. - * - */ - - -#include <linux/delay.h> -#include <linux/platform_device.h> -#include <linux/init.h> -#include <linux/input.h> -#include <linux/interrupt.h> -#include <linux/module.h> -#include <linux/slab.h> -#include <linux/irq.h> - -#include <mach/sharpsl.h> -#include <mach/hardware.h> -#include <mach/pxa2xx-gpio.h> - - -#define PWR_MODE_ACTIVE 0 -#define PWR_MODE_SUSPEND 1 - -#define X_AXIS_MAX 3830 -#define X_AXIS_MIN 150 -#define Y_AXIS_MAX 3830 -#define Y_AXIS_MIN 190 -#define PRESSURE_MIN 0 -#define PRESSURE_MAX 15000 - -struct ts_event { - short pressure; - short x; - short y; -}; - -struct corgi_ts { - struct input_dev *input; - struct timer_list timer; - struct ts_event tc; - int pendown; - int power_mode; - int irq_gpio; - struct corgits_machinfo *machinfo; -}; - -#ifdef CONFIG_PXA25x -#define CCNT(a) asm volatile ("mrc p14, 0, %0, C1, C0, 0" : "=r"(a)) -#define PMNC_GET(x) asm volatile ("mrc p14, 0, %0, C0, C0, 0" : "=r"(x)) -#define PMNC_SET(x) asm volatile ("mcr p14, 0, %0, C0, C0, 0" : : "r"(x)) -#endif -#ifdef CONFIG_PXA27x -#define CCNT(a) asm volatile ("mrc p14, 0, %0, C1, C1, 0" : "=r"(a)) -#define PMNC_GET(x) asm volatile ("mrc p14, 0, %0, C0, C1, 0" : "=r"(x)) -#define PMNC_SET(x) asm volatile ("mcr p14, 0, %0, C0, C1, 0" : : "r"(x)) -#endif - -/* ADS7846 Touch Screen Controller bit definitions */ -#define ADSCTRL_PD0 (1u << 0) /* PD0 */ -#define ADSCTRL_PD1 (1u << 1) /* PD1 */ -#define ADSCTRL_DFR (1u << 2) /* SER/DFR */ -#define ADSCTRL_MOD (1u << 3) /* Mode */ -#define ADSCTRL_ADR_SH 4 /* Address setting */ -#define ADSCTRL_STS (1u << 7) /* Start Bit */ - -/* External Functions */ -extern unsigned int get_clk_frequency_khz(int info); - -static unsigned long calc_waittime(struct corgi_ts *corgi_ts) -{ - unsigned long hsync_invperiod = corgi_ts->machinfo->get_hsync_invperiod(); - - if (hsync_invperiod) - return get_clk_frequency_khz(0)*1000/hsync_invperiod; - else - return 0; -} - -static int sync_receive_data_send_cmd(struct corgi_ts *corgi_ts, int doRecive, int doSend, - unsigned int address, unsigned long wait_time) -{ - unsigned long timer1 = 0, timer2, pmnc = 0; - int pos = 0; - - if (wait_time && doSend) { - PMNC_GET(pmnc); - if (!(pmnc & 0x01)) - PMNC_SET(0x01); - - /* polling HSync */ - corgi_ts->machinfo->wait_hsync(); - /* get CCNT */ - CCNT(timer1); - } - - if (doRecive) - pos = corgi_ssp_ads7846_get(); - - if (doSend) { - int cmd = ADSCTRL_PD0 | ADSCTRL_PD1 | (address << ADSCTRL_ADR_SH) | ADSCTRL_STS; - /* dummy command */ - corgi_ssp_ads7846_put(cmd); - corgi_ssp_ads7846_get(); - - if (wait_time) { - /* Wait after HSync */ - CCNT(timer2); - if (timer2-timer1 > wait_time) { - /* too slow - timeout, try again */ - corgi_ts->machinfo->wait_hsync(); - /* get CCNT */ - CCNT(timer1); - /* Wait after HSync */ - CCNT(timer2); - } - while (timer2 - timer1 < wait_time) - CCNT(timer2); - } - corgi_ssp_ads7846_put(cmd); - if (wait_time && !(pmnc & 0x01)) - PMNC_SET(pmnc); - } - return pos; -} - -static int read_xydata(struct corgi_ts *corgi_ts) -{ - unsigned int x, y, z1, z2; - unsigned long flags, wait_time; - - /* critical section */ - local_irq_save(flags); - corgi_ssp_ads7846_lock(); - wait_time = calc_waittime(corgi_ts); - - /* Y-axis */ - sync_receive_data_send_cmd(corgi_ts, 0, 1, 1u, wait_time); - - /* Y-axis */ - sync_receive_data_send_cmd(corgi_ts, 1, 1, 1u, wait_time); - - /* X-axis */ - y = sync_receive_data_send_cmd(corgi_ts, 1, 1, 5u, wait_time); - - /* Z1 */ - x = sync_receive_data_send_cmd(corgi_ts, 1, 1, 3u, wait_time); - - /* Z2 */ - z1 = sync_receive_data_send_cmd(corgi_ts, 1, 1, 4u, wait_time); - z2 = sync_receive_data_send_cmd(corgi_ts, 1, 0, 4u, wait_time); - - /* Power-Down Enable */ - corgi_ssp_ads7846_put((1u << ADSCTRL_ADR_SH) | ADSCTRL_STS); - corgi_ssp_ads7846_get(); - - corgi_ssp_ads7846_unlock(); - local_irq_restore(flags); - - if (x== 0 || y == 0 || z1 == 0 || (x * (z2 - z1) / z1) >= 15000) { - corgi_ts->tc.pressure = 0; - return 0; - } - - corgi_ts->tc.x = x; - corgi_ts->tc.y = y; - corgi_ts->tc.pressure = (x * (z2 - z1)) / z1; - return 1; -} - -static void new_data(struct corgi_ts *corgi_ts) -{ - struct input_dev *dev = corgi_ts->input; - - if (corgi_ts->power_mode != PWR_MODE_ACTIVE) - return; - - if (!corgi_ts->tc.pressure && corgi_ts->pendown == 0) - return; - - input_report_abs(dev, ABS_X, corgi_ts->tc.x); - input_report_abs(dev, ABS_Y, corgi_ts->tc.y); - input_report_abs(dev, ABS_PRESSURE, corgi_ts->tc.pressure); - input_report_key(dev, BTN_TOUCH, corgi_ts->pendown); - input_sync(dev); -} - -static void ts_interrupt_main(struct corgi_ts *corgi_ts, int isTimer) -{ - if ((GPLR(IRQ_TO_GPIO(corgi_ts->irq_gpio)) & GPIO_bit(IRQ_TO_GPIO(corgi_ts->irq_gpio))) == 0) { - /* Disable Interrupt */ - set_irq_type(corgi_ts->irq_gpio, IRQ_TYPE_NONE); - if (read_xydata(corgi_ts)) { - corgi_ts->pendown = 1; - new_data(corgi_ts); - } - mod_timer(&corgi_ts->timer, jiffies + HZ / 100); - } else { - if (corgi_ts->pendown == 1 || corgi_ts->pendown == 2) { - mod_timer(&corgi_ts->timer, jiffies + HZ / 100); - corgi_ts->pendown++; - return; - } - - if (corgi_ts->pendown) { - corgi_ts->tc.pressure = 0; - new_data(corgi_ts); - } - - /* Enable Falling Edge */ - set_irq_type(corgi_ts->irq_gpio, IRQ_TYPE_EDGE_FALLING); - corgi_ts->pendown = 0; - } -} - -static void corgi_ts_timer(unsigned long data) -{ - struct corgi_ts *corgits_data = (struct corgi_ts *) data; - - ts_interrupt_main(corgits_data, 1); -} - -static irqreturn_t ts_interrupt(int irq, void *dev_id) -{ - struct corgi_ts *corgits_data = dev_id; - - ts_interrupt_main(corgits_data, 0); - return IRQ_HANDLED; -} - -#ifdef CONFIG_PM -static int corgits_suspend(struct platform_device *dev, pm_message_t state) -{ - struct corgi_ts *corgi_ts = platform_get_drvdata(dev); - - if (corgi_ts->pendown) { - del_timer_sync(&corgi_ts->timer); - corgi_ts->tc.pressure = 0; - new_data(corgi_ts); - corgi_ts->pendown = 0; - } - corgi_ts->power_mode = PWR_MODE_SUSPEND; - - corgi_ssp_ads7846_putget((1u << ADSCTRL_ADR_SH) | ADSCTRL_STS); - - return 0; -} - -static int corgits_resume(struct platform_device *dev) -{ - struct corgi_ts *corgi_ts = platform_get_drvdata(dev); - - corgi_ssp_ads7846_putget((4u << ADSCTRL_ADR_SH) | ADSCTRL_STS); - /* Enable Falling Edge */ - set_irq_type(corgi_ts->irq_gpio, IRQ_TYPE_EDGE_FALLING); - corgi_ts->power_mode = PWR_MODE_ACTIVE; - - return 0; -} -#else -#define corgits_suspend NULL -#define corgits_resume NULL -#endif - -static int __devinit corgits_probe(struct platform_device *pdev) -{ - struct corgi_ts *corgi_ts; - struct input_dev *input_dev; - int err = -ENOMEM; - - corgi_ts = kzalloc(sizeof(struct corgi_ts), GFP_KERNEL); - input_dev = input_allocate_device(); - if (!corgi_ts || !input_dev) - goto fail1; - - platform_set_drvdata(pdev, corgi_ts); - - corgi_ts->machinfo = pdev->dev.platform_data; - corgi_ts->irq_gpio = platform_get_irq(pdev, 0); - - if (corgi_ts->irq_gpio < 0) { - err = -ENODEV; - goto fail1; - } - - corgi_ts->input = input_dev; - - init_timer(&corgi_ts->timer); - corgi_ts->timer.data = (unsigned long) corgi_ts; - corgi_ts->timer.function = corgi_ts_timer; - - input_dev->name = "Corgi Touchscreen"; - input_dev->phys = "corgits/input0"; - input_dev->id.bustype = BUS_HOST; - input_dev->id.vendor = 0x0001; - input_dev->id.product = 0x0002; - input_dev->id.version = 0x0100; - input_dev->dev.parent = &pdev->dev; - - input_dev->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); - input_dev->keybit[BIT_WORD(BTN_TOUCH)] = BIT_MASK(BTN_TOUCH); - input_set_abs_params(input_dev, ABS_X, X_AXIS_MIN, X_AXIS_MAX, 0, 0); - input_set_abs_params(input_dev, ABS_Y, Y_AXIS_MIN, Y_AXIS_MAX, 0, 0); - input_set_abs_params(input_dev, ABS_PRESSURE, PRESSURE_MIN, PRESSURE_MAX, 0, 0); - - pxa_gpio_mode(IRQ_TO_GPIO(corgi_ts->irq_gpio) | GPIO_IN); - - /* Initiaize ADS7846 Difference Reference mode */ - corgi_ssp_ads7846_putget((1u << ADSCTRL_ADR_SH) | ADSCTRL_STS); - mdelay(5); - corgi_ssp_ads7846_putget((3u << ADSCTRL_ADR_SH) | ADSCTRL_STS); - mdelay(5); - corgi_ssp_ads7846_putget((4u << ADSCTRL_ADR_SH) | ADSCTRL_STS); - mdelay(5); - corgi_ssp_ads7846_putget((5u << ADSCTRL_ADR_SH) | ADSCTRL_STS); - mdelay(5); - - if (request_irq(corgi_ts->irq_gpio, ts_interrupt, IRQF_DISABLED, "ts", corgi_ts)) { - err = -EBUSY; - goto fail1; - } - - err = input_register_device(corgi_ts->input); - if (err) - goto fail2; - - corgi_ts->power_mode = PWR_MODE_ACTIVE; - - /* Enable Falling Edge */ - set_irq_type(corgi_ts->irq_gpio, IRQ_TYPE_EDGE_FALLING); - - return 0; - - fail2: free_irq(corgi_ts->irq_gpio, corgi_ts); - fail1: input_free_device(input_dev); - kfree(corgi_ts); - return err; -} - -static int __devexit corgits_remove(struct platform_device *pdev) -{ - struct corgi_ts *corgi_ts = platform_get_drvdata(pdev); - - free_irq(corgi_ts->irq_gpio, corgi_ts); - del_timer_sync(&corgi_ts->timer); - corgi_ts->machinfo->put_hsync(); - input_unregister_device(corgi_ts->input); - kfree(corgi_ts); - - return 0; -} - -static struct platform_driver corgits_driver = { - .probe = corgits_probe, - .remove = __devexit_p(corgits_remove), - .suspend = corgits_suspend, - .resume = corgits_resume, - .driver = { - .name = "corgi-ts", - .owner = THIS_MODULE, - }, -}; - -static int __init corgits_init(void) -{ - return platform_driver_register(&corgits_driver); -} - -static void __exit corgits_exit(void) -{ - platform_driver_unregister(&corgits_driver); -} - -module_init(corgits_init); -module_exit(corgits_exit); - -MODULE_AUTHOR("Richard Purdie <rpurdie(a)rpsys.net>"); -MODULE_DESCRIPTION("Corgi TouchScreen Driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:corgi-ts"); -- 1.6.0.4

1 0

[alsa-devel] [RFC PATCH 0/5] PXA SSP code cleanup
by Eric Miao 23 Apr '09

23 Apr '09

Remove legacy SSP API by deprecating the only user now - corgi_ts.c, (which could be now well substituted by ads7846 after the sync patch) sound/soc/pxa/pxa-ssp.c is modified and simplified a bit. I guess sound/soc/pxa/pxa-ssp.c will still have to undergo the changes as ASoCv2 progress, hopefully that should be a little bit easier now. -- Cheers - eric

1 0

[PATCH] ASoC: simplify the SSP DMA parameters settings by
by Eric Miao 23 Apr '09

23 Apr '09

run-time generation The SSP DMA parameters can actually be easily generated at run-time since they are almost similar except for the FIFO width and direction. Another benefit is the re-use of information from 'struct ssp_device', like SSDR physical FIFO address and DRCMR register index for both directions. Signed-off-by: Eric Miao <eric.miao(a)marvell.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Reviewed-by: pHilipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 203 ++++++++++------------------------------------- 1 files changed, 43 insertions(+), 160 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index b9b61dd..fb8cacc 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -50,139 +50,6 @@ struct ssp_priv { #endif }; -#define PXA2xx_SSP1_BASE 0x41000000 -#define PXA27x_SSP2_BASE 0x41700000 -#define PXA27x_SSP3_BASE 0x41900000 -#define PXA3xx_SSP4_BASE 0x41a00000 - -static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_out = { - .name = "SSP1 PCM Mono out", - .dev_addr = PXA2xx_SSP1_BASE + SSDR, - .drcmr = &DRCMR(14), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_in = { - .name = "SSP1 PCM Mono in", - .dev_addr = PXA2xx_SSP1_BASE + SSDR, - .drcmr = &DRCMR(13), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_out = { - .name = "SSP1 PCM Stereo out", - .dev_addr = PXA2xx_SSP1_BASE + SSDR, - .drcmr = &DRCMR(14), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_in = { - .name = "SSP1 PCM Stereo in", - .dev_addr = PXA2xx_SSP1_BASE + SSDR, - .drcmr = &DRCMR(13), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_out = { - .name = "SSP2 PCM Mono out", - .dev_addr = PXA27x_SSP2_BASE + SSDR, - .drcmr = &DRCMR(16), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_in = { - .name = "SSP2 PCM Mono in", - .dev_addr = PXA27x_SSP2_BASE + SSDR, - .drcmr = &DRCMR(15), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_out = { - .name = "SSP2 PCM Stereo out", - .dev_addr = PXA27x_SSP2_BASE + SSDR, - .drcmr = &DRCMR(16), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_in = { - .name = "SSP2 PCM Stereo in", - .dev_addr = PXA27x_SSP2_BASE + SSDR, - .drcmr = &DRCMR(15), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_out = { - .name = "SSP3 PCM Mono out", - .dev_addr = PXA27x_SSP3_BASE + SSDR, - .drcmr = &DRCMR(67), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_in = { - .name = "SSP3 PCM Mono in", - .dev_addr = PXA27x_SSP3_BASE + SSDR, - .drcmr = &DRCMR(66), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_out = { - .name = "SSP3 PCM Stereo out", - .dev_addr = PXA27x_SSP3_BASE + SSDR, - .drcmr = &DRCMR(67), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_in = { - .name = "SSP3 PCM Stereo in", - .dev_addr = PXA27x_SSP3_BASE + SSDR, - .drcmr = &DRCMR(66), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_out = { - .name = "SSP4 PCM Mono out", - .dev_addr = PXA3xx_SSP4_BASE + SSDR, - .drcmr = &DRCMR(67), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_in = { - .name = "SSP4 PCM Mono in", - .dev_addr = PXA3xx_SSP4_BASE + SSDR, - .drcmr = &DRCMR(66), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_out = { - .name = "SSP4 PCM Stereo out", - .dev_addr = PXA3xx_SSP4_BASE + SSDR, - .drcmr = &DRCMR(67), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_in = { - .name = "SSP4 PCM Stereo in", - .dev_addr = PXA3xx_SSP4_BASE + SSDR, - .drcmr = &DRCMR(66), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH4, -}; - static void dump_registers(struct ssp_device *ssp) { dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", @@ -194,25 +61,33 @@ static void dump_registers(struct ssp_device *ssp) ssp_read_reg(ssp, SSACD)); } -static struct pxa2xx_pcm_dma_params *ssp_dma_params[4][4] = { - { - &pxa_ssp1_pcm_mono_out, &pxa_ssp1_pcm_mono_in, - &pxa_ssp1_pcm_stereo_out, &pxa_ssp1_pcm_stereo_in, - }, - { - &pxa_ssp2_pcm_mono_out, &pxa_ssp2_pcm_mono_in, - &pxa_ssp2_pcm_stereo_out, &pxa_ssp2_pcm_stereo_in, - }, - { - &pxa_ssp3_pcm_mono_out, &pxa_ssp3_pcm_mono_in, - &pxa_ssp3_pcm_stereo_out, &pxa_ssp3_pcm_stereo_in, - }, - { - &pxa_ssp4_pcm_mono_out, &pxa_ssp4_pcm_mono_in, - &pxa_ssp4_pcm_stereo_out, &pxa_ssp4_pcm_stereo_in, - }, +struct pxa2xx_pcm_dma_data { + struct pxa2xx_pcm_dma_params params; + char name[20]; }; +static struct pxa2xx_pcm_dma_params * +ssp_get_dma_params(struct ssp_device *ssp, int stereo, int out) +{ + struct pxa2xx_pcm_dma_data *dma; + + dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL); + if (dma == NULL) + return NULL; + + snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id, + stereo ? "Stereo" : "Mono", out ? "out" : "in"); + + dma->params.name = dma->name; + dma->params.drcmr = &DRCMR(out ? ssp->drcmr_tx : ssp->drcmr_rx); + dma->params.dcmd = (out ? (DCMD_INCSRCADDR | DCMD_FLOWTRG) : + (DCMD_INCTRGADDR | DCMD_FLOWSRC)) | + (stereo ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16; + dma->params.dev_addr = ssp->phys_base + SSDR; + + return &dma->params; +} + static int pxa_ssp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -227,6 +102,11 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, clk_enable(priv->dev.ssp->clk); ssp_disable(&priv->dev); } + + if (cpu_dai->dma_data) { + kfree(cpu_dai->dma_data); + cpu_dai->dma_data = NULL; + } return ret; } @@ -241,6 +121,11 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, ssp_disable(&priv->dev); clk_disable(priv->dev.ssp->clk); } + + if (cpu_dai->dma_data) { + kfree(cpu_dai->dma_data); + cpu_dai->dma_data = NULL; + } } #ifdef CONFIG_PM @@ -653,25 +538,23 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; - int dma = 0, chn = params_channels(params); + int chn = params_channels(params); u32 sscr0; u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; - /* select correct DMA params */ - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) - dma = 1; /* capture DMA offset is 1,3 */ + /* generate correct DMA params */ + if (cpu_dai->dma_data) + kfree(cpu_dai->dma_data); + /* Network mode with one active slot (ttsa == 1) can be used * to force 16-bit frame width on the wire (for S16_LE), even * with two channels. Use 16-bit DMA transfers for this case. */ - if (((chn == 2) && (ttsa != 1)) || (width == 32)) - dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */ - - cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma]; - - dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma); + cpu_dai->dma_data = ssp_get_dma_params(ssp, + ((chn == 2) && (ttsa != 1)) || (width == 32), + substream->stream == SNDRV_PCM_STREAM_PLAYBACK); /* we can only change the settings if the port is not in use */ if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) -- 1.6.0.4 --0016364ee0220e5a4e0468352f08 Content-Type: text/x-diff; charset=US-ASCII; name="0001-ASoC-simplify-the-SSP-DMA-parameters-settings-by-ru.patch" Content-Disposition: attachment; filename="0001-ASoC-simplify-the-SSP-DMA-parameters-settings-by-ru.patch" Content-Transfer-Encoding: base64 X-Attachment-Id: f_ftv880y00 RnJvbSA2NTViYWQ4ZTcyZTNhYTEzZjBkZGY5MTBkNWY0YjlmMDY2ZWM5Y2MzIE1vbiBTZXAgMTcg MDA6MDA6MDAgMjAwMQpGcm9tOiBFcmljIE1pYW8gPGVyaWMubWlhb0BtYXJ2ZWxsLmNvbT4KRGF0 ZTogVGh1LCAyMyBBcHIgMjAwOSAxMDo1ODozNCArMDgwMApTdWJqZWN0OiBbUEFUQ0hdIEFTb0M6 IHNpbXBsaWZ5IHRoZSBTU1AgRE1BIHBhcmFtZXRlcnMgc2V0dGluZ3MgYnkgcnVuLXRpbWUgZ2Vu ZXJhdGlvbgoKVGhlIFNTUCBETUEgcGFyYW1ldGVycyBjYW4gYWN0dWFsbHkgYmUgZWFzaWx5IGdl bmVyYXRlZCBhdCBydW4tdGltZSBzaW5jZQp0aGV5IGFyZSBhbG1vc3Qgc2ltaWxhciBleGNlcHQg Zm9yIHRoZSBGSUZPIHdpZHRoIGFuZCBkaXJlY3Rpb24uIEFub3RoZXIKYmVuZWZpdCBpcyB0aGUg cmUtdXNlIG9mIGluZm9ybWF0aW9uIGZyb20gJ3N0cnVjdCBzc3BfZGV2aWNlJywgbGlrZSBTU0RS CnBoeXNpY2FsIEZJRk8gYWRkcmVzcyBhbmQgRFJDTVIgcmVnaXN0ZXIgaW5kZXggZm9yIGJvdGgg ZGlyZWN0aW9ucy4KClNpZ25lZC1vZmYtYnk6IEVyaWMgTWlhbyA8ZXJpYy5taWFvQG1hcnZlbGwu Y29tPgpTaWduZWQtb2ZmLWJ5OiBNYXJrIEJyb3duIDxicm9vbmllQG9wZW5zb3VyY2Uud29sZnNv bm1pY3JvLmNvbT4KUmV2aWV3ZWQtYnk6IHBIaWxpcHAgWmFiZWwgPHBoaWxpcHAuemFiZWxAZ21h aWwuY29tPgotLS0KIHNvdW5kL3NvYy9weGEvcHhhLXNzcC5jIHwgIDIwMyArKysrKysrKysrLS0t LS0tLS0tLS0tLS0tLS0tLS0tLS0tLS0tLS0tLS0tLS0tLQogMSBmaWxlcyBjaGFuZ2VkLCA0MyBp bnNlcnRpb25zKCspLCAxNjAgZGVsZXRpb25zKC0pCgpkaWZmIC0tZ2l0IGEvc291bmQvc29jL3B4 YS9weGEtc3NwLmMgYi9zb3VuZC9zb2MvcHhhL3B4YS1zc3AuYwppbmRleCBiOWI2MWRkLi5mYjhj YWNjIDEwMDY0NAotLS0gYS9zb3VuZC9zb2MvcHhhL3B4YS1zc3AuYworKysgYi9zb3VuZC9zb2Mv cHhhL3B4YS1zc3AuYwpAQCAtNTAsMTM5ICs1MCw2IEBAIHN0cnVjdCBzc3BfcHJpdiB7CiAjZW5k aWYKIH07CiAKLSNkZWZpbmUgUFhBMnh4X1NTUDFfQkFTRQkweDQxMDAwMDAwCi0jZGVmaW5lIFBY QTI3eF9TU1AyX0JBU0UJMHg0MTcwMDAwMAotI2RlZmluZSBQWEEyN3hfU1NQM19CQVNFCTB4NDE5 MDAwMDAKLSNkZWZpbmUgUFhBM3h4X1NTUDRfQkFTRQkweDQxYTAwMDAwCi0KLXN0YXRpYyBzdHJ1 Y3QgcHhhMnh4X3BjbV9kbWFfcGFyYW1zIHB4YV9zc3AxX3BjbV9tb25vX291dCA9IHsKLQkubmFt ZQkJCT0gIlNTUDEgUENNIE1vbm8gb3V0IiwKLQkuZGV2X2FkZHIJCT0gUFhBMnh4X1NTUDFfQkFT RSArIFNTRFIsCi0JLmRyY21yCQkJPSAmRFJDTVIoMTQpLAotCS5kY21kCQkJPSBEQ01EX0lOQ1NS Q0FERFIgfCBEQ01EX0ZMT1dUUkcgfAotCQkJCSAgRENNRF9CVVJTVDE2IHwgRENNRF9XSURUSDIs Ci19OwotCi1zdGF0aWMgc3RydWN0IHB4YTJ4eF9wY21fZG1hX3BhcmFtcyBweGFfc3NwMV9wY21f bW9ub19pbiA9IHsKLQkubmFtZQkJCT0gIlNTUDEgUENNIE1vbm8gaW4iLAotCS5kZXZfYWRkcgkJ PSBQWEEyeHhfU1NQMV9CQVNFICsgU1NEUiwKLQkuZHJjbXIJCQk9ICZEUkNNUigxMyksCi0JLmRj bWQJCQk9IERDTURfSU5DVFJHQUREUiB8IERDTURfRkxPV1NSQyB8Ci0JCQkJICBEQ01EX0JVUlNU MTYgfCBEQ01EX1dJRFRIMiwKLX07Ci0KLXN0YXRpYyBzdHJ1Y3QgcHhhMnh4X3BjbV9kbWFfcGFy YW1zIHB4YV9zc3AxX3BjbV9zdGVyZW9fb3V0ID0gewotCS5uYW1lCQkJPSAiU1NQMSBQQ00gU3Rl cmVvIG91dCIsCi0JLmRldl9hZGRyCQk9IFBYQTJ4eF9TU1AxX0JBU0UgKyBTU0RSLAotCS5kcmNt cgkJCT0gJkRSQ01SKDE0KSwKLQkuZGNtZAkJCT0gRENNRF9JTkNTUkNBRERSIHwgRENNRF9GTE9X VFJHIHwKLQkJCQkgIERDTURfQlVSU1QxNiB8IERDTURfV0lEVEg0LAotfTsKLQotc3RhdGljIHN0 cnVjdCBweGEyeHhfcGNtX2RtYV9wYXJhbXMgcHhhX3NzcDFfcGNtX3N0ZXJlb19pbiA9IHsKLQku bmFtZQkJCT0gIlNTUDEgUENNIFN0ZXJlbyBpbiIsCi0JLmRldl9hZGRyCQk9IFBYQTJ4eF9TU1Ax X0JBU0UgKyBTU0RSLAotCS5kcmNtcgkJCT0gJkRSQ01SKDEzKSwKLQkuZGNtZAkJCT0gRENNRF9J TkNUUkdBRERSIHwgRENNRF9GTE9XU1JDIHwKLQkJCQkgIERDTURfQlVSU1QxNiB8IERDTURfV0lE VEg0LAotfTsKLQotc3RhdGljIHN0cnVjdCBweGEyeHhfcGNtX2RtYV9wYXJhbXMgcHhhX3NzcDJf cGNtX21vbm9fb3V0ID0gewotCS5uYW1lCQkJPSAiU1NQMiBQQ00gTW9ubyBvdXQiLAotCS5kZXZf YWRkcgkJPSBQWEEyN3hfU1NQMl9CQVNFICsgU1NEUiwKLQkuZHJjbXIJCQk9ICZEUkNNUigxNiks Ci0JLmRjbWQJCQk9IERDTURfSU5DU1JDQUREUiB8IERDTURfRkxPV1RSRyB8Ci0JCQkJICBEQ01E X0JVUlNUMTYgfCBEQ01EX1dJRFRIMiwKLX07Ci0KLXN0YXRpYyBzdHJ1Y3QgcHhhMnh4X3BjbV9k bWFfcGFyYW1zIHB4YV9zc3AyX3BjbV9tb25vX2luID0gewotCS5uYW1lCQkJPSAiU1NQMiBQQ00g TW9ubyBpbiIsCi0JLmRldl9hZGRyCQk9IFBYQTI3eF9TU1AyX0JBU0UgKyBTU0RSLAotCS5kcmNt cgkJCT0gJkRSQ01SKDE1KSwKLQkuZGNtZAkJCT0gRENNRF9JTkNUUkdBRERSIHwgRENNRF9GTE9X U1JDIHwKLQkJCQkgIERDTURfQlVSU1QxNiB8IERDTURfV0lEVEgyLAotfTsKLQotc3RhdGljIHN0 cnVjdCBweGEyeHhfcGNtX2RtYV9wYXJhbXMgcHhhX3NzcDJfcGNtX3N0ZXJlb19vdXQgPSB7Ci0J Lm5hbWUJCQk9ICJTU1AyIFBDTSBTdGVyZW8gb3V0IiwKLQkuZGV2X2FkZHIJCT0gUFhBMjd4X1NT UDJfQkFTRSArIFNTRFIsCi0JLmRyY21yCQkJPSAmRFJDTVIoMTYpLAotCS5kY21kCQkJPSBEQ01E X0lOQ1NSQ0FERFIgfCBEQ01EX0ZMT1dUUkcgfAotCQkJCSAgRENNRF9CVVJTVDE2IHwgRENNRF9X SURUSDQsCi19OwotCi1zdGF0aWMgc3RydWN0IHB4YTJ4eF9wY21fZG1hX3BhcmFtcyBweGFfc3Nw Ml9wY21fc3RlcmVvX2luID0gewotCS5uYW1lCQkJPSAiU1NQMiBQQ00gU3RlcmVvIGluIiwKLQku ZGV2X2FkZHIJCT0gUFhBMjd4X1NTUDJfQkFTRSArIFNTRFIsCi0JLmRyY21yCQkJPSAmRFJDTVIo MTUpLAotCS5kY21kCQkJPSBEQ01EX0lOQ1RSR0FERFIgfCBEQ01EX0ZMT1dTUkMgfAotCQkJCSAg RENNRF9CVVJTVDE2IHwgRENNRF9XSURUSDQsCi19OwotCi1zdGF0aWMgc3RydWN0IHB4YTJ4eF9w Y21fZG1hX3BhcmFtcyBweGFfc3NwM19wY21fbW9ub19vdXQgPSB7Ci0JLm5hbWUJCQk9ICJTU1Az IFBDTSBNb25vIG91dCIsCi0JLmRldl9hZGRyCQk9IFBYQTI3eF9TU1AzX0JBU0UgKyBTU0RSLAot CS5kcmNtcgkJCT0gJkRSQ01SKDY3KSwKLQkuZGNtZAkJCT0gRENNRF9JTkNTUkNBRERSIHwgRENN RF9GTE9XVFJHIHwKLQkJCQkgIERDTURfQlVSU1QxNiB8IERDTURfV0lEVEgyLAotfTsKLQotc3Rh dGljIHN0cnVjdCBweGEyeHhfcGNtX2RtYV9wYXJhbXMgcHhhX3NzcDNfcGNtX21vbm9faW4gPSB7 Ci0JLm5hbWUJCQk9ICJTU1AzIFBDTSBNb25vIGluIiwKLQkuZGV2X2FkZHIJCT0gUFhBMjd4X1NT UDNfQkFTRSArIFNTRFIsCi0JLmRyY21yCQkJPSAmRFJDTVIoNjYpLAotCS5kY21kCQkJPSBEQ01E X0lOQ1RSR0FERFIgfCBEQ01EX0ZMT1dTUkMgfAotCQkJCSAgRENNRF9CVVJTVDE2IHwgRENNRF9X SURUSDIsCi19OwotCi1zdGF0aWMgc3RydWN0IHB4YTJ4eF9wY21fZG1hX3BhcmFtcyBweGFfc3Nw M19wY21fc3RlcmVvX291dCA9IHsKLQkubmFtZQkJCT0gIlNTUDMgUENNIFN0ZXJlbyBvdXQiLAot CS5kZXZfYWRkcgkJPSBQWEEyN3hfU1NQM19CQVNFICsgU1NEUiwKLQkuZHJjbXIJCQk9ICZEUkNN Uig2NyksCi0JLmRjbWQJCQk9IERDTURfSU5DU1JDQUREUiB8IERDTURfRkxPV1RSRyB8Ci0JCQkJ ICBEQ01EX0JVUlNUMTYgfCBEQ01EX1dJRFRINCwKLX07Ci0KLXN0YXRpYyBzdHJ1Y3QgcHhhMnh4 X3BjbV9kbWFfcGFyYW1zIHB4YV9zc3AzX3BjbV9zdGVyZW9faW4gPSB7Ci0JLm5hbWUJCQk9ICJT U1AzIFBDTSBTdGVyZW8gaW4iLAotCS5kZXZfYWRkcgkJPSBQWEEyN3hfU1NQM19CQVNFICsgU1NE UiwKLQkuZHJjbXIJCQk9ICZEUkNNUig2NiksCi0JLmRjbWQJCQk9IERDTURfSU5DVFJHQUREUiB8 IERDTURfRkxPV1NSQyB8Ci0JCQkJICBEQ01EX0JVUlNUMTYgfCBEQ01EX1dJRFRINCwKLX07Ci0K LXN0YXRpYyBzdHJ1Y3QgcHhhMnh4X3BjbV9kbWFfcGFyYW1zIHB4YV9zc3A0X3BjbV9tb25vX291 dCA9IHsKLQkubmFtZQkJCT0gIlNTUDQgUENNIE1vbm8gb3V0IiwKLQkuZGV2X2FkZHIJCT0gUFhB M3h4X1NTUDRfQkFTRSArIFNTRFIsCi0JLmRyY21yCQkJPSAmRFJDTVIoNjcpLAotCS5kY21kCQkJ PSBEQ01EX0lOQ1NSQ0FERFIgfCBEQ01EX0ZMT1dUUkcgfAotCQkJCSAgRENNRF9CVVJTVDE2IHwg RENNRF9XSURUSDIsCi19OwotCi1zdGF0aWMgc3RydWN0IHB4YTJ4eF9wY21fZG1hX3BhcmFtcyBw eGFfc3NwNF9wY21fbW9ub19pbiA9IHsKLQkubmFtZQkJCT0gIlNTUDQgUENNIE1vbm8gaW4iLAot CS5kZXZfYWRkcgkJPSBQWEEzeHhfU1NQNF9CQVNFICsgU1NEUiwKLQkuZHJjbXIJCQk9ICZEUkNN Uig2NiksCi0JLmRjbWQJCQk9IERDTURfSU5DVFJHQUREUiB8IERDTURfRkxPV1NSQyB8Ci0JCQkJ ICBEQ01EX0JVUlNUMTYgfCBEQ01EX1dJRFRIMiwKLX07Ci0KLXN0YXRpYyBzdHJ1Y3QgcHhhMnh4 X3BjbV9kbWFfcGFyYW1zIHB4YV9zc3A0X3BjbV9zdGVyZW9fb3V0ID0gewotCS5uYW1lCQkJPSAi U1NQNCBQQ00gU3RlcmVvIG91dCIsCi0JLmRldl9hZGRyCQk9IFBYQTN4eF9TU1A0X0JBU0UgKyBT U0RSLAotCS5kcmNtcgkJCT0gJkRSQ01SKDY3KSwKLQkuZGNtZAkJCT0gRENNRF9JTkNTUkNBRERS IHwgRENNRF9GTE9XVFJHIHwKLQkJCQkgIERDTURfQlVSU1QxNiB8IERDTURfV0lEVEg0LAotfTsK LQotc3RhdGljIHN0cnVjdCBweGEyeHhfcGNtX2RtYV9wYXJhbXMgcHhhX3NzcDRfcGNtX3N0ZXJl b19pbiA9IHsKLQkubmFtZQkJCT0gIlNTUDQgUENNIFN0ZXJlbyBpbiIsCi0JLmRldl9hZGRyCQk9 IFBYQTN4eF9TU1A0X0JBU0UgKyBTU0RSLAotCS5kcmNtcgkJCT0gJkRSQ01SKDY2KSwKLQkuZGNt ZAkJCT0gRENNRF9JTkNUUkdBRERSIHwgRENNRF9GTE9XU1JDIHwKLQkJCQkgIERDTURfQlVSU1Qx NiB8IERDTURfV0lEVEg0LAotfTsKLQogc3RhdGljIHZvaWQgZHVtcF9yZWdpc3RlcnMoc3RydWN0 IHNzcF9kZXZpY2UgKnNzcCkKIHsKIAlkZXZfZGJnKCZzc3AtPnBkZXYtPmRldiwgIlNTQ1IwIDB4 JTA4eCBTU0NSMSAweCUwOHggU1NUTyAweCUwOHhcbiIsCkBAIC0xOTQsMjUgKzYxLDMzIEBAIHN0 YXRpYyB2b2lkIGR1bXBfcmVnaXN0ZXJzKHN0cnVjdCBzc3BfZGV2aWNlICpzc3ApCiAJCSBzc3Bf cmVhZF9yZWcoc3NwLCBTU0FDRCkpOwogfQogCi1zdGF0aWMgc3RydWN0IHB4YTJ4eF9wY21fZG1h X3BhcmFtcyAqc3NwX2RtYV9wYXJhbXNbNF1bNF0gPSB7Ci0JewotCQkmcHhhX3NzcDFfcGNtX21v bm9fb3V0LCAmcHhhX3NzcDFfcGNtX21vbm9faW4sCi0JCSZweGFfc3NwMV9wY21fc3RlcmVvX291 dCwgJnB4YV9zc3AxX3BjbV9zdGVyZW9faW4sCi0JfSwKLQl7Ci0JCSZweGFfc3NwMl9wY21fbW9u b19vdXQsICZweGFfc3NwMl9wY21fbW9ub19pbiwKLQkJJnB4YV9zc3AyX3BjbV9zdGVyZW9fb3V0 LCAmcHhhX3NzcDJfcGNtX3N0ZXJlb19pbiwKLQl9LAotCXsKLQkJJnB4YV9zc3AzX3BjbV9tb25v X291dCwgJnB4YV9zc3AzX3BjbV9tb25vX2luLAotCQkmcHhhX3NzcDNfcGNtX3N0ZXJlb19vdXQs ICZweGFfc3NwM19wY21fc3RlcmVvX2luLAotCX0sCi0JewotCQkmcHhhX3NzcDRfcGNtX21vbm9f b3V0LCAmcHhhX3NzcDRfcGNtX21vbm9faW4sCi0JCSZweGFfc3NwNF9wY21fc3RlcmVvX291dCwg JnB4YV9zc3A0X3BjbV9zdGVyZW9faW4sCi0JfSwKK3N0cnVjdCBweGEyeHhfcGNtX2RtYV9kYXRh IHsKKwlzdHJ1Y3QgcHhhMnh4X3BjbV9kbWFfcGFyYW1zIHBhcmFtczsKKwljaGFyIG5hbWVbMjBd OwogfTsKIAorc3RhdGljIHN0cnVjdCBweGEyeHhfcGNtX2RtYV9wYXJhbXMgKgorc3NwX2dldF9k bWFfcGFyYW1zKHN0cnVjdCBzc3BfZGV2aWNlICpzc3AsIGludCBzdGVyZW8sIGludCBvdXQpCit7 CisJc3RydWN0IHB4YTJ4eF9wY21fZG1hX2RhdGEgKmRtYTsKKworCWRtYSA9IGt6YWxsb2Moc2l6 ZW9mKHN0cnVjdCBweGEyeHhfcGNtX2RtYV9kYXRhKSwgR0ZQX0tFUk5FTCk7CisJaWYgKGRtYSA9 PSBOVUxMKQorCQlyZXR1cm4gTlVMTDsKKworCXNucHJpbnRmKGRtYS0+bmFtZSwgMjAsICJTU1Al ZCBQQ00gJXMgJXMiLCBzc3AtPnBvcnRfaWQsCisJCQlzdGVyZW8gPyAiU3RlcmVvIiA6ICJNb25v Iiwgb3V0ID8gIm91dCIgOiAiaW4iKTsKKworCWRtYS0+cGFyYW1zLm5hbWUgPSBkbWEtPm5hbWU7 CisJZG1hLT5wYXJhbXMuZHJjbXIgPSAmRFJDTVIob3V0ID8gc3NwLT5kcmNtcl90eCA6IHNzcC0+ ZHJjbXJfcngpOworCWRtYS0+cGFyYW1zLmRjbWQgPSAob3V0ID8gKERDTURfSU5DU1JDQUREUiB8 IERDTURfRkxPV1RSRykgOgorCQkJCSAgKERDTURfSU5DVFJHQUREUiB8IERDTURfRkxPV1NSQykp IHwKKwkJCShzdGVyZW8gPyBEQ01EX1dJRFRINCA6IERDTURfV0lEVEgyKSB8IERDTURfQlVSU1Qx NjsKKwlkbWEtPnBhcmFtcy5kZXZfYWRkciA9IHNzcC0+cGh5c19iYXNlICsgU1NEUjsKKworCXJl dHVybiAmZG1hLT5wYXJhbXM7Cit9CisKIHN0YXRpYyBpbnQgcHhhX3NzcF9zdGFydHVwKHN0cnVj dCBzbmRfcGNtX3N1YnN0cmVhbSAqc3Vic3RyZWFtLAogCQkJICAgc3RydWN0IHNuZF9zb2NfZGFp ICpkYWkpCiB7CkBAIC0yMjcsNiArMTAyLDExIEBAIHN0YXRpYyBpbnQgcHhhX3NzcF9zdGFydHVw KHN0cnVjdCBzbmRfcGNtX3N1YnN0cmVhbSAqc3Vic3RyZWFtLAogCQljbGtfZW5hYmxlKHByaXYt PmRldi5zc3AtPmNsayk7CiAJCXNzcF9kaXNhYmxlKCZwcml2LT5kZXYpOwogCX0KKworCWlmIChj cHVfZGFpLT5kbWFfZGF0YSkgeworCQlrZnJlZShjcHVfZGFpLT5kbWFfZGF0YSk7CisJCWNwdV9k YWktPmRtYV9kYXRhID0gTlVMTDsKKwl9CiAJcmV0dXJuIHJldDsKIH0KIApAQCAtMjQxLDYgKzEy MSwxMSBAQCBzdGF0aWMgdm9pZCBweGFfc3NwX3NodXRkb3duKHN0cnVjdCBzbmRfcGNtX3N1YnN0 cmVhbSAqc3Vic3RyZWFtLAogCQlzc3BfZGlzYWJsZSgmcHJpdi0+ZGV2KTsKIAkJY2xrX2Rpc2Fi bGUocHJpdi0+ZGV2LnNzcC0+Y2xrKTsKIAl9CisKKwlpZiAoY3B1X2RhaS0+ZG1hX2RhdGEpIHsK KwkJa2ZyZWUoY3B1X2RhaS0+ZG1hX2RhdGEpOworCQljcHVfZGFpLT5kbWFfZGF0YSA9IE5VTEw7 CisJfQogfQogCiAjaWZkZWYgQ09ORklHX1BNCkBAIC02NTMsMjUgKzUzOCwyMyBAQCBzdGF0aWMg aW50IHB4YV9zc3BfaHdfcGFyYW1zKHN0cnVjdCBzbmRfcGNtX3N1YnN0cmVhbSAqc3Vic3RyZWFt LAogCXN0cnVjdCBzbmRfc29jX2RhaSAqY3B1X2RhaSA9IHJ0ZC0+ZGFpLT5jcHVfZGFpOwogCXN0 cnVjdCBzc3BfcHJpdiAqcHJpdiA9IGNwdV9kYWktPnByaXZhdGVfZGF0YTsKIAlzdHJ1Y3Qgc3Nw X2RldmljZSAqc3NwID0gcHJpdi0+ZGV2LnNzcDsKLQlpbnQgZG1hID0gMCwgY2huID0gcGFyYW1z X2NoYW5uZWxzKHBhcmFtcyk7CisJaW50IGNobiA9IHBhcmFtc19jaGFubmVscyhwYXJhbXMpOwog CXUzMiBzc2NyMDsKIAl1MzIgc3Nwc3A7CiAJaW50IHdpZHRoID0gc25kX3BjbV9mb3JtYXRfcGh5 c2ljYWxfd2lkdGgocGFyYW1zX2Zvcm1hdChwYXJhbXMpKTsKIAlpbnQgdHRzYSA9IHNzcF9yZWFk X3JlZyhzc3AsIFNTVFNBKSAmIDB4ZjsKIAotCS8qIHNlbGVjdCBjb3JyZWN0IERNQSBwYXJhbXMg Ki8KLQlpZiAoc3Vic3RyZWFtLT5zdHJlYW0gIT0gU05EUlZfUENNX1NUUkVBTV9QTEFZQkFDSykK LQkJZG1hID0gMTsgLyogY2FwdHVyZSBETUEgb2Zmc2V0IGlzIDEsMyAqLworCS8qIGdlbmVyYXRl IGNvcnJlY3QgRE1BIHBhcmFtcyAqLworCWlmIChjcHVfZGFpLT5kbWFfZGF0YSkKKwkJa2ZyZWUo Y3B1X2RhaS0+ZG1hX2RhdGEpOworCiAJLyogTmV0d29yayBtb2RlIHdpdGggb25lIGFjdGl2ZSBz bG90ICh0dHNhID09IDEpIGNhbiBiZSB1c2VkCiAJICogdG8gZm9yY2UgMTYtYml0IGZyYW1lIHdp ZHRoIG9uIHRoZSB3aXJlIChmb3IgUzE2X0xFKSwgZXZlbgogCSAqIHdpdGggdHdvIGNoYW5uZWxz LiBVc2UgMTYtYml0IERNQSB0cmFuc2ZlcnMgZm9yIHRoaXMgY2FzZS4KIAkgKi8KLQlpZiAoKChj aG4gPT0gMikgJiYgKHR0c2EgIT0gMSkpIHx8ICh3aWR0aCA9PSAzMikpCi0JCWRtYSArPSAyOyAv KiAzMi1iaXQgRE1BIG9mZnNldCBpcyAyLCAxNi1iaXQgaXMgMCAqLwotCi0JY3B1X2RhaS0+ZG1h X2RhdGEgPSBzc3BfZG1hX3BhcmFtc1tjcHVfZGFpLT5pZF1bZG1hXTsKLQotCWRldl9kYmcoJnNz cC0+cGRldi0+ZGV2LCAicHhhX3NzcF9od19wYXJhbXM6IGRtYSAlZFxuIiwgZG1hKTsKKwljcHVf ZGFpLT5kbWFfZGF0YSA9IHNzcF9nZXRfZG1hX3BhcmFtcyhzc3AsCisJCQkoKGNobiA9PSAyKSAm JiAodHRzYSAhPSAxKSkgfHwgKHdpZHRoID09IDMyKSwKKwkJCXN1YnN0cmVhbS0+c3RyZWFtID09 IFNORFJWX1BDTV9TVFJFQU1fUExBWUJBQ0spOwogCiAJLyogd2UgY2FuIG9ubHkgY2hhbmdlIHRo ZSBzZXR0aW5ncyBpZiB0aGUgcG9ydCBpcyBub3QgaW4gdXNlICovCiAJaWYgKHNzcF9yZWFkX3Jl Zyhzc3AsIFNTQ1IwKSAmIFNTQ1IwX1NTRSkKLS0gCjEuNi4wLjQKCg== --0016364ee0220e5a4e0468352f08--

1 0

[PATCH] ASoC: simplify the SSP DMA parameters settings by
by Eric Miao 23 Apr '09

23 Apr '09

run-time generation The SSP DMA parameters can actually be easily generated at run-time since they are almost similar except for the FIFO width and direction. Another benefit is the re-use of information from 'struct ssp_device', like SSDR physical FIFO address and DRCMR register index for both directions. In addition, the name of the DMA parameter has been changed to something like: "SSP3 PCM 16-bit out" or "SSP3 PCM 32-bit in" instead of the original "Stereo" and "Mono". This is more consistent with the code. Signed-off-by: Eric Miao <eric.miao(a)marvell.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Reviewed-by: pHilipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 203 ++++++++++------------------------------------- 1 files changed, 43 insertions(+), 160 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index b9b61dd..6fc7876 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -50,139 +50,6 @@ struct ssp_priv { #endif }; -#define PXA2xx_SSP1_BASE 0x41000000 -#define PXA27x_SSP2_BASE 0x41700000 -#define PXA27x_SSP3_BASE 0x41900000 -#define PXA3xx_SSP4_BASE 0x41a00000 - -static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_out = { - .name = "SSP1 PCM Mono out", - .dev_addr = PXA2xx_SSP1_BASE + SSDR, - .drcmr = &DRCMR(14), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_in = { - .name = "SSP1 PCM Mono in", - .dev_addr = PXA2xx_SSP1_BASE + SSDR, - .drcmr = &DRCMR(13), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_out = { - .name = "SSP1 PCM Stereo out", - .dev_addr = PXA2xx_SSP1_BASE + SSDR, - .drcmr = &DRCMR(14), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_in = { - .name = "SSP1 PCM Stereo in", - .dev_addr = PXA2xx_SSP1_BASE + SSDR, - .drcmr = &DRCMR(13), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_out = { - .name = "SSP2 PCM Mono out", - .dev_addr = PXA27x_SSP2_BASE + SSDR, - .drcmr = &DRCMR(16), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_in = { - .name = "SSP2 PCM Mono in", - .dev_addr = PXA27x_SSP2_BASE + SSDR, - .drcmr = &DRCMR(15), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_out = { - .name = "SSP2 PCM Stereo out", - .dev_addr = PXA27x_SSP2_BASE + SSDR, - .drcmr = &DRCMR(16), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_in = { - .name = "SSP2 PCM Stereo in", - .dev_addr = PXA27x_SSP2_BASE + SSDR, - .drcmr = &DRCMR(15), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_out = { - .name = "SSP3 PCM Mono out", - .dev_addr = PXA27x_SSP3_BASE + SSDR, - .drcmr = &DRCMR(67), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_in = { - .name = "SSP3 PCM Mono in", - .dev_addr = PXA27x_SSP3_BASE + SSDR, - .drcmr = &DRCMR(66), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_out = { - .name = "SSP3 PCM Stereo out", - .dev_addr = PXA27x_SSP3_BASE + SSDR, - .drcmr = &DRCMR(67), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_in = { - .name = "SSP3 PCM Stereo in", - .dev_addr = PXA27x_SSP3_BASE + SSDR, - .drcmr = &DRCMR(66), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_out = { - .name = "SSP4 PCM Mono out", - .dev_addr = PXA3xx_SSP4_BASE + SSDR, - .drcmr = &DRCMR(67), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_in = { - .name = "SSP4 PCM Mono in", - .dev_addr = PXA3xx_SSP4_BASE + SSDR, - .drcmr = &DRCMR(66), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_out = { - .name = "SSP4 PCM Stereo out", - .dev_addr = PXA3xx_SSP4_BASE + SSDR, - .drcmr = &DRCMR(67), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_in = { - .name = "SSP4 PCM Stereo in", - .dev_addr = PXA3xx_SSP4_BASE + SSDR, - .drcmr = &DRCMR(66), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH4, -}; - static void dump_registers(struct ssp_device *ssp) { dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", @@ -194,25 +61,33 @@ static void dump_registers(struct ssp_device *ssp) ssp_read_reg(ssp, SSACD)); } -static struct pxa2xx_pcm_dma_params *ssp_dma_params[4][4] = { - { - &pxa_ssp1_pcm_mono_out, &pxa_ssp1_pcm_mono_in, - &pxa_ssp1_pcm_stereo_out, &pxa_ssp1_pcm_stereo_in, - }, - { - &pxa_ssp2_pcm_mono_out, &pxa_ssp2_pcm_mono_in, - &pxa_ssp2_pcm_stereo_out, &pxa_ssp2_pcm_stereo_in, - }, - { - &pxa_ssp3_pcm_mono_out, &pxa_ssp3_pcm_mono_in, - &pxa_ssp3_pcm_stereo_out, &pxa_ssp3_pcm_stereo_in, - }, - { - &pxa_ssp4_pcm_mono_out, &pxa_ssp4_pcm_mono_in, - &pxa_ssp4_pcm_stereo_out, &pxa_ssp4_pcm_stereo_in, - }, +struct pxa2xx_pcm_dma_data { + struct pxa2xx_pcm_dma_params params; + char name[20]; }; +static struct pxa2xx_pcm_dma_params * +ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) +{ + struct pxa2xx_pcm_dma_data *dma; + + dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL); + if (dma == NULL) + return NULL; + + snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id, + width4 ? "32-bit" : "16-bit", out ? "out" : "in"); + + dma->params.name = dma->name; + dma->params.drcmr = &DRCMR(out ? ssp->drcmr_tx : ssp->drcmr_rx); + dma->params.dcmd = (out ? (DCMD_INCSRCADDR | DCMD_FLOWTRG) : + (DCMD_INCTRGADDR | DCMD_FLOWSRC)) | + (width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16; + dma->params.dev_addr = ssp->phys_base + SSDR; + + return &dma->params; +} + static int pxa_ssp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -227,6 +102,11 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, clk_enable(priv->dev.ssp->clk); ssp_disable(&priv->dev); } + + if (cpu_dai->dma_data) { + kfree(cpu_dai->dma_data); + cpu_dai->dma_data = NULL; + } return ret; } @@ -241,6 +121,11 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, ssp_disable(&priv->dev); clk_disable(priv->dev.ssp->clk); } + + if (cpu_dai->dma_data) { + kfree(cpu_dai->dma_data); + cpu_dai->dma_data = NULL; + } } #ifdef CONFIG_PM @@ -653,25 +538,23 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; - int dma = 0, chn = params_channels(params); + int chn = params_channels(params); u32 sscr0; u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; - /* select correct DMA params */ - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) - dma = 1; /* capture DMA offset is 1,3 */ + /* generate correct DMA params */ + if (cpu_dai->dma_data) + kfree(cpu_dai->dma_data); + /* Network mode with one active slot (ttsa == 1) can be used * to force 16-bit frame width on the wire (for S16_LE), even * with two channels. Use 16-bit DMA transfers for this case. */ - if (((chn == 2) && (ttsa != 1)) || (width == 32)) - dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */ - - cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma]; - - dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma); + cpu_dai->dma_data = ssp_get_dma_params(ssp, + ((chn == 2) && (ttsa != 1)) || (width == 32), + substream->stream == SNDRV_PCM_STREAM_PLAYBACK); /* we can only change the settings if the port is not in use */ if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) -- 1.6.0.4 --0016364eef74c9234b046835b0db Content-Type: text/x-diff; charset=US-ASCII; name="0001-ASoC-simplify-the-SSP-DMA-parameters-settings-by-ru.patch" Content-Disposition: attachment; filename="0001-ASoC-simplify-the-SSP-DMA-parameters-settings-by-ru.patch" Content-Transfer-Encoding: base64 X-Attachment-Id: f_ftv9ixzn0 RnJvbSAzZGVhZjNjZTQzOTc5MDY2OGZkM2YwNDE0MjMyNmNmODRjYjcxYjM5IE1vbiBTZXAgMTcg MDA6MDA6MDAgMjAwMQpGcm9tOiBFcmljIE1pYW8gPGVyaWMubWlhb0BtYXJ2ZWxsLmNvbT4KRGF0 ZTogVGh1LCAyMyBBcHIgMjAwOSAxMDo1ODozNCArMDgwMApTdWJqZWN0OiBbUEFUQ0hdIEFTb0M6 IHNpbXBsaWZ5IHRoZSBTU1AgRE1BIHBhcmFtZXRlcnMgc2V0dGluZ3MgYnkgcnVuLXRpbWUgZ2Vu ZXJhdGlvbgoKVGhlIFNTUCBETUEgcGFyYW1ldGVycyBjYW4gYWN0dWFsbHkgYmUgZWFzaWx5IGdl bmVyYXRlZCBhdCBydW4tdGltZSBzaW5jZQp0aGV5IGFyZSBhbG1vc3Qgc2ltaWxhciBleGNlcHQg Zm9yIHRoZSBGSUZPIHdpZHRoIGFuZCBkaXJlY3Rpb24uIEFub3RoZXIKYmVuZWZpdCBpcyB0aGUg cmUtdXNlIG9mIGluZm9ybWF0aW9uIGZyb20gJ3N0cnVjdCBzc3BfZGV2aWNlJywgbGlrZSBTU0RS CnBoeXNpY2FsIEZJRk8gYWRkcmVzcyBhbmQgRFJDTVIgcmVnaXN0ZXIgaW5kZXggZm9yIGJvdGgg ZGlyZWN0aW9ucy4KCkluIGFkZGl0aW9uLCB0aGUgbmFtZSBvZiB0aGUgRE1BIHBhcmFtZXRlciBo YXMgYmVlbiBjaGFuZ2VkIHRvIHNvbWV0aGluZwpsaWtlOgoKICAiU1NQMyBQQ00gMTYtYml0IG91 dCIgb3IgIlNTUDMgUENNIDMyLWJpdCBpbiIKCmluc3RlYWQgb2YgdGhlIG9yaWdpbmFsICJTdGVy ZW8iIGFuZCAiTW9ubyIuIFRoaXMgaXMgbW9yZSBjb25zaXN0ZW50IHdpdGgKdGhlIGNvZGUuCgpT aWduZWQtb2ZmLWJ5OiBFcmljIE1pYW8gPGVyaWMubWlhb0BtYXJ2ZWxsLmNvbT4KU2lnbmVkLW9m Zi1ieTogTWFyayBCcm93biA8YnJvb25pZUBvcGVuc291cmNlLndvbGZzb25taWNyby5jb20+ClJl dmlld2VkLWJ5OiBwSGlsaXBwIFphYmVsIDxwaGlsaXBwLnphYmVsQGdtYWlsLmNvbT4KLS0tCiBz b3VuZC9zb2MvcHhhL3B4YS1zc3AuYyB8ICAyMDMgKysrKysrKysrKy0tLS0tLS0tLS0tLS0tLS0t LS0tLS0tLS0tLS0tLS0tLS0tLS0KIDEgZmlsZXMgY2hhbmdlZCwgNDMgaW5zZXJ0aW9ucygrKSwg MTYwIGRlbGV0aW9ucygtKQoKZGlmZiAtLWdpdCBhL3NvdW5kL3NvYy9weGEvcHhhLXNzcC5jIGIv c291bmQvc29jL3B4YS9weGEtc3NwLmMKaW5kZXggYjliNjFkZC4uNmZjNzg3NiAxMDA2NDQKLS0t IGEvc291bmQvc29jL3B4YS9weGEtc3NwLmMKKysrIGIvc291bmQvc29jL3B4YS9weGEtc3NwLmMK QEAgLTUwLDEzOSArNTAsNiBAQCBzdHJ1Y3Qgc3NwX3ByaXYgewogI2VuZGlmCiB9OwogCi0jZGVm aW5lIFBYQTJ4eF9TU1AxX0JBU0UJMHg0MTAwMDAwMAotI2RlZmluZSBQWEEyN3hfU1NQMl9CQVNF CTB4NDE3MDAwMDAKLSNkZWZpbmUgUFhBMjd4X1NTUDNfQkFTRQkweDQxOTAwMDAwCi0jZGVmaW5l IFBYQTN4eF9TU1A0X0JBU0UJMHg0MWEwMDAwMAotCi1zdGF0aWMgc3RydWN0IHB4YTJ4eF9wY21f ZG1hX3BhcmFtcyBweGFfc3NwMV9wY21fbW9ub19vdXQgPSB7Ci0JLm5hbWUJCQk9ICJTU1AxIFBD TSBNb25vIG91dCIsCi0JLmRldl9hZGRyCQk9IFBYQTJ4eF9TU1AxX0JBU0UgKyBTU0RSLAotCS5k cmNtcgkJCT0gJkRSQ01SKDE0KSwKLQkuZGNtZAkJCT0gRENNRF9JTkNTUkNBRERSIHwgRENNRF9G TE9XVFJHIHwKLQkJCQkgIERDTURfQlVSU1QxNiB8IERDTURfV0lEVEgyLAotfTsKLQotc3RhdGlj IHN0cnVjdCBweGEyeHhfcGNtX2RtYV9wYXJhbXMgcHhhX3NzcDFfcGNtX21vbm9faW4gPSB7Ci0J Lm5hbWUJCQk9ICJTU1AxIFBDTSBNb25vIGluIiwKLQkuZGV2X2FkZHIJCT0gUFhBMnh4X1NTUDFf QkFTRSArIFNTRFIsCi0JLmRyY21yCQkJPSAmRFJDTVIoMTMpLAotCS5kY21kCQkJPSBEQ01EX0lO Q1RSR0FERFIgfCBEQ01EX0ZMT1dTUkMgfAotCQkJCSAgRENNRF9CVVJTVDE2IHwgRENNRF9XSURU SDIsCi19OwotCi1zdGF0aWMgc3RydWN0IHB4YTJ4eF9wY21fZG1hX3BhcmFtcyBweGFfc3NwMV9w Y21fc3RlcmVvX291dCA9IHsKLQkubmFtZQkJCT0gIlNTUDEgUENNIFN0ZXJlbyBvdXQiLAotCS5k ZXZfYWRkcgkJPSBQWEEyeHhfU1NQMV9CQVNFICsgU1NEUiwKLQkuZHJjbXIJCQk9ICZEUkNNUigx NCksCi0JLmRjbWQJCQk9IERDTURfSU5DU1JDQUREUiB8IERDTURfRkxPV1RSRyB8Ci0JCQkJICBE Q01EX0JVUlNUMTYgfCBEQ01EX1dJRFRINCwKLX07Ci0KLXN0YXRpYyBzdHJ1Y3QgcHhhMnh4X3Bj bV9kbWFfcGFyYW1zIHB4YV9zc3AxX3BjbV9zdGVyZW9faW4gPSB7Ci0JLm5hbWUJCQk9ICJTU1Ax IFBDTSBTdGVyZW8gaW4iLAotCS5kZXZfYWRkcgkJPSBQWEEyeHhfU1NQMV9CQVNFICsgU1NEUiwK LQkuZHJjbXIJCQk9ICZEUkNNUigxMyksCi0JLmRjbWQJCQk9IERDTURfSU5DVFJHQUREUiB8IERD TURfRkxPV1NSQyB8Ci0JCQkJICBEQ01EX0JVUlNUMTYgfCBEQ01EX1dJRFRINCwKLX07Ci0KLXN0 YXRpYyBzdHJ1Y3QgcHhhMnh4X3BjbV9kbWFfcGFyYW1zIHB4YV9zc3AyX3BjbV9tb25vX291dCA9 IHsKLQkubmFtZQkJCT0gIlNTUDIgUENNIE1vbm8gb3V0IiwKLQkuZGV2X2FkZHIJCT0gUFhBMjd4 X1NTUDJfQkFTRSArIFNTRFIsCi0JLmRyY21yCQkJPSAmRFJDTVIoMTYpLAotCS5kY21kCQkJPSBE Q01EX0lOQ1NSQ0FERFIgfCBEQ01EX0ZMT1dUUkcgfAotCQkJCSAgRENNRF9CVVJTVDE2IHwgRENN RF9XSURUSDIsCi19OwotCi1zdGF0aWMgc3RydWN0IHB4YTJ4eF9wY21fZG1hX3BhcmFtcyBweGFf c3NwMl9wY21fbW9ub19pbiA9IHsKLQkubmFtZQkJCT0gIlNTUDIgUENNIE1vbm8gaW4iLAotCS5k ZXZfYWRkcgkJPSBQWEEyN3hfU1NQMl9CQVNFICsgU1NEUiwKLQkuZHJjbXIJCQk9ICZEUkNNUigx NSksCi0JLmRjbWQJCQk9IERDTURfSU5DVFJHQUREUiB8IERDTURfRkxPV1NSQyB8Ci0JCQkJICBE Q01EX0JVUlNUMTYgfCBEQ01EX1dJRFRIMiwKLX07Ci0KLXN0YXRpYyBzdHJ1Y3QgcHhhMnh4X3Bj bV9kbWFfcGFyYW1zIHB4YV9zc3AyX3BjbV9zdGVyZW9fb3V0ID0gewotCS5uYW1lCQkJPSAiU1NQ MiBQQ00gU3RlcmVvIG91dCIsCi0JLmRldl9hZGRyCQk9IFBYQTI3eF9TU1AyX0JBU0UgKyBTU0RS LAotCS5kcmNtcgkJCT0gJkRSQ01SKDE2KSwKLQkuZGNtZAkJCT0gRENNRF9JTkNTUkNBRERSIHwg RENNRF9GTE9XVFJHIHwKLQkJCQkgIERDTURfQlVSU1QxNiB8IERDTURfV0lEVEg0LAotfTsKLQot c3RhdGljIHN0cnVjdCBweGEyeHhfcGNtX2RtYV9wYXJhbXMgcHhhX3NzcDJfcGNtX3N0ZXJlb19p biA9IHsKLQkubmFtZQkJCT0gIlNTUDIgUENNIFN0ZXJlbyBpbiIsCi0JLmRldl9hZGRyCQk9IFBY QTI3eF9TU1AyX0JBU0UgKyBTU0RSLAotCS5kcmNtcgkJCT0gJkRSQ01SKDE1KSwKLQkuZGNtZAkJ CT0gRENNRF9JTkNUUkdBRERSIHwgRENNRF9GTE9XU1JDIHwKLQkJCQkgIERDTURfQlVSU1QxNiB8 IERDTURfV0lEVEg0LAotfTsKLQotc3RhdGljIHN0cnVjdCBweGEyeHhfcGNtX2RtYV9wYXJhbXMg cHhhX3NzcDNfcGNtX21vbm9fb3V0ID0gewotCS5uYW1lCQkJPSAiU1NQMyBQQ00gTW9ubyBvdXQi LAotCS5kZXZfYWRkcgkJPSBQWEEyN3hfU1NQM19CQVNFICsgU1NEUiwKLQkuZHJjbXIJCQk9ICZE UkNNUig2NyksCi0JLmRjbWQJCQk9IERDTURfSU5DU1JDQUREUiB8IERDTURfRkxPV1RSRyB8Ci0J CQkJICBEQ01EX0JVUlNUMTYgfCBEQ01EX1dJRFRIMiwKLX07Ci0KLXN0YXRpYyBzdHJ1Y3QgcHhh Mnh4X3BjbV9kbWFfcGFyYW1zIHB4YV9zc3AzX3BjbV9tb25vX2luID0gewotCS5uYW1lCQkJPSAi U1NQMyBQQ00gTW9ubyBpbiIsCi0JLmRldl9hZGRyCQk9IFBYQTI3eF9TU1AzX0JBU0UgKyBTU0RS LAotCS5kcmNtcgkJCT0gJkRSQ01SKDY2KSwKLQkuZGNtZAkJCT0gRENNRF9JTkNUUkdBRERSIHwg RENNRF9GTE9XU1JDIHwKLQkJCQkgIERDTURfQlVSU1QxNiB8IERDTURfV0lEVEgyLAotfTsKLQot c3RhdGljIHN0cnVjdCBweGEyeHhfcGNtX2RtYV9wYXJhbXMgcHhhX3NzcDNfcGNtX3N0ZXJlb19v dXQgPSB7Ci0JLm5hbWUJCQk9ICJTU1AzIFBDTSBTdGVyZW8gb3V0IiwKLQkuZGV2X2FkZHIJCT0g UFhBMjd4X1NTUDNfQkFTRSArIFNTRFIsCi0JLmRyY21yCQkJPSAmRFJDTVIoNjcpLAotCS5kY21k CQkJPSBEQ01EX0lOQ1NSQ0FERFIgfCBEQ01EX0ZMT1dUUkcgfAotCQkJCSAgRENNRF9CVVJTVDE2 IHwgRENNRF9XSURUSDQsCi19OwotCi1zdGF0aWMgc3RydWN0IHB4YTJ4eF9wY21fZG1hX3BhcmFt cyBweGFfc3NwM19wY21fc3RlcmVvX2luID0gewotCS5uYW1lCQkJPSAiU1NQMyBQQ00gU3RlcmVv IGluIiwKLQkuZGV2X2FkZHIJCT0gUFhBMjd4X1NTUDNfQkFTRSArIFNTRFIsCi0JLmRyY21yCQkJ PSAmRFJDTVIoNjYpLAotCS5kY21kCQkJPSBEQ01EX0lOQ1RSR0FERFIgfCBEQ01EX0ZMT1dTUkMg fAotCQkJCSAgRENNRF9CVVJTVDE2IHwgRENNRF9XSURUSDQsCi19OwotCi1zdGF0aWMgc3RydWN0 IHB4YTJ4eF9wY21fZG1hX3BhcmFtcyBweGFfc3NwNF9wY21fbW9ub19vdXQgPSB7Ci0JLm5hbWUJ CQk9ICJTU1A0IFBDTSBNb25vIG91dCIsCi0JLmRldl9hZGRyCQk9IFBYQTN4eF9TU1A0X0JBU0Ug KyBTU0RSLAotCS5kcmNtcgkJCT0gJkRSQ01SKDY3KSwKLQkuZGNtZAkJCT0gRENNRF9JTkNTUkNB RERSIHwgRENNRF9GTE9XVFJHIHwKLQkJCQkgIERDTURfQlVSU1QxNiB8IERDTURfV0lEVEgyLAot fTsKLQotc3RhdGljIHN0cnVjdCBweGEyeHhfcGNtX2RtYV9wYXJhbXMgcHhhX3NzcDRfcGNtX21v bm9faW4gPSB7Ci0JLm5hbWUJCQk9ICJTU1A0IFBDTSBNb25vIGluIiwKLQkuZGV2X2FkZHIJCT0g UFhBM3h4X1NTUDRfQkFTRSArIFNTRFIsCi0JLmRyY21yCQkJPSAmRFJDTVIoNjYpLAotCS5kY21k CQkJPSBEQ01EX0lOQ1RSR0FERFIgfCBEQ01EX0ZMT1dTUkMgfAotCQkJCSAgRENNRF9CVVJTVDE2 IHwgRENNRF9XSURUSDIsCi19OwotCi1zdGF0aWMgc3RydWN0IHB4YTJ4eF9wY21fZG1hX3BhcmFt cyBweGFfc3NwNF9wY21fc3RlcmVvX291dCA9IHsKLQkubmFtZQkJCT0gIlNTUDQgUENNIFN0ZXJl byBvdXQiLAotCS5kZXZfYWRkcgkJPSBQWEEzeHhfU1NQNF9CQVNFICsgU1NEUiwKLQkuZHJjbXIJ CQk9ICZEUkNNUig2NyksCi0JLmRjbWQJCQk9IERDTURfSU5DU1JDQUREUiB8IERDTURfRkxPV1RS RyB8Ci0JCQkJICBEQ01EX0JVUlNUMTYgfCBEQ01EX1dJRFRINCwKLX07Ci0KLXN0YXRpYyBzdHJ1 Y3QgcHhhMnh4X3BjbV9kbWFfcGFyYW1zIHB4YV9zc3A0X3BjbV9zdGVyZW9faW4gPSB7Ci0JLm5h bWUJCQk9ICJTU1A0IFBDTSBTdGVyZW8gaW4iLAotCS5kZXZfYWRkcgkJPSBQWEEzeHhfU1NQNF9C QVNFICsgU1NEUiwKLQkuZHJjbXIJCQk9ICZEUkNNUig2NiksCi0JLmRjbWQJCQk9IERDTURfSU5D VFJHQUREUiB8IERDTURfRkxPV1NSQyB8Ci0JCQkJICBEQ01EX0JVUlNUMTYgfCBEQ01EX1dJRFRI NCwKLX07Ci0KIHN0YXRpYyB2b2lkIGR1bXBfcmVnaXN0ZXJzKHN0cnVjdCBzc3BfZGV2aWNlICpz c3ApCiB7CiAJZGV2X2RiZygmc3NwLT5wZGV2LT5kZXYsICJTU0NSMCAweCUwOHggU1NDUjEgMHgl MDh4IFNTVE8gMHglMDh4XG4iLApAQCAtMTk0LDI1ICs2MSwzMyBAQCBzdGF0aWMgdm9pZCBkdW1w X3JlZ2lzdGVycyhzdHJ1Y3Qgc3NwX2RldmljZSAqc3NwKQogCQkgc3NwX3JlYWRfcmVnKHNzcCwg U1NBQ0QpKTsKIH0KIAotc3RhdGljIHN0cnVjdCBweGEyeHhfcGNtX2RtYV9wYXJhbXMgKnNzcF9k bWFfcGFyYW1zWzRdWzRdID0gewotCXsKLQkJJnB4YV9zc3AxX3BjbV9tb25vX291dCwgJnB4YV9z c3AxX3BjbV9tb25vX2luLAotCQkmcHhhX3NzcDFfcGNtX3N0ZXJlb19vdXQsICZweGFfc3NwMV9w Y21fc3RlcmVvX2luLAotCX0sCi0JewotCQkmcHhhX3NzcDJfcGNtX21vbm9fb3V0LCAmcHhhX3Nz cDJfcGNtX21vbm9faW4sCi0JCSZweGFfc3NwMl9wY21fc3RlcmVvX291dCwgJnB4YV9zc3AyX3Bj bV9zdGVyZW9faW4sCi0JfSwKLQl7Ci0JCSZweGFfc3NwM19wY21fbW9ub19vdXQsICZweGFfc3Nw M19wY21fbW9ub19pbiwKLQkJJnB4YV9zc3AzX3BjbV9zdGVyZW9fb3V0LCAmcHhhX3NzcDNfcGNt X3N0ZXJlb19pbiwKLQl9LAotCXsKLQkJJnB4YV9zc3A0X3BjbV9tb25vX291dCwgJnB4YV9zc3A0 X3BjbV9tb25vX2luLAotCQkmcHhhX3NzcDRfcGNtX3N0ZXJlb19vdXQsICZweGFfc3NwNF9wY21f c3RlcmVvX2luLAotCX0sCitzdHJ1Y3QgcHhhMnh4X3BjbV9kbWFfZGF0YSB7CisJc3RydWN0IHB4 YTJ4eF9wY21fZG1hX3BhcmFtcyBwYXJhbXM7CisJY2hhciBuYW1lWzIwXTsKIH07CiAKK3N0YXRp YyBzdHJ1Y3QgcHhhMnh4X3BjbV9kbWFfcGFyYW1zICoKK3NzcF9nZXRfZG1hX3BhcmFtcyhzdHJ1 Y3Qgc3NwX2RldmljZSAqc3NwLCBpbnQgd2lkdGg0LCBpbnQgb3V0KQoreworCXN0cnVjdCBweGEy eHhfcGNtX2RtYV9kYXRhICpkbWE7CisKKwlkbWEgPSBremFsbG9jKHNpemVvZihzdHJ1Y3QgcHhh Mnh4X3BjbV9kbWFfZGF0YSksIEdGUF9LRVJORUwpOworCWlmIChkbWEgPT0gTlVMTCkKKwkJcmV0 dXJuIE5VTEw7CisKKwlzbnByaW50ZihkbWEtPm5hbWUsIDIwLCAiU1NQJWQgUENNICVzICVzIiwg c3NwLT5wb3J0X2lkLAorCQkJd2lkdGg0ID8gIjMyLWJpdCIgOiAiMTYtYml0Iiwgb3V0ID8gIm91 dCIgOiAiaW4iKTsKKworCWRtYS0+cGFyYW1zLm5hbWUgPSBkbWEtPm5hbWU7CisJZG1hLT5wYXJh bXMuZHJjbXIgPSAmRFJDTVIob3V0ID8gc3NwLT5kcmNtcl90eCA6IHNzcC0+ZHJjbXJfcngpOwor CWRtYS0+cGFyYW1zLmRjbWQgPSAob3V0ID8gKERDTURfSU5DU1JDQUREUiB8IERDTURfRkxPV1RS RykgOgorCQkJCSAgKERDTURfSU5DVFJHQUREUiB8IERDTURfRkxPV1NSQykpIHwKKwkJCSh3aWR0 aDQgPyBEQ01EX1dJRFRINCA6IERDTURfV0lEVEgyKSB8IERDTURfQlVSU1QxNjsKKwlkbWEtPnBh cmFtcy5kZXZfYWRkciA9IHNzcC0+cGh5c19iYXNlICsgU1NEUjsKKworCXJldHVybiAmZG1hLT5w YXJhbXM7Cit9CisKIHN0YXRpYyBpbnQgcHhhX3NzcF9zdGFydHVwKHN0cnVjdCBzbmRfcGNtX3N1 YnN0cmVhbSAqc3Vic3RyZWFtLAogCQkJICAgc3RydWN0IHNuZF9zb2NfZGFpICpkYWkpCiB7CkBA IC0yMjcsNiArMTAyLDExIEBAIHN0YXRpYyBpbnQgcHhhX3NzcF9zdGFydHVwKHN0cnVjdCBzbmRf cGNtX3N1YnN0cmVhbSAqc3Vic3RyZWFtLAogCQljbGtfZW5hYmxlKHByaXYtPmRldi5zc3AtPmNs ayk7CiAJCXNzcF9kaXNhYmxlKCZwcml2LT5kZXYpOwogCX0KKworCWlmIChjcHVfZGFpLT5kbWFf ZGF0YSkgeworCQlrZnJlZShjcHVfZGFpLT5kbWFfZGF0YSk7CisJCWNwdV9kYWktPmRtYV9kYXRh ID0gTlVMTDsKKwl9CiAJcmV0dXJuIHJldDsKIH0KIApAQCAtMjQxLDYgKzEyMSwxMSBAQCBzdGF0 aWMgdm9pZCBweGFfc3NwX3NodXRkb3duKHN0cnVjdCBzbmRfcGNtX3N1YnN0cmVhbSAqc3Vic3Ry ZWFtLAogCQlzc3BfZGlzYWJsZSgmcHJpdi0+ZGV2KTsKIAkJY2xrX2Rpc2FibGUocHJpdi0+ZGV2 LnNzcC0+Y2xrKTsKIAl9CisKKwlpZiAoY3B1X2RhaS0+ZG1hX2RhdGEpIHsKKwkJa2ZyZWUoY3B1 X2RhaS0+ZG1hX2RhdGEpOworCQljcHVfZGFpLT5kbWFfZGF0YSA9IE5VTEw7CisJfQogfQogCiAj aWZkZWYgQ09ORklHX1BNCkBAIC02NTMsMjUgKzUzOCwyMyBAQCBzdGF0aWMgaW50IHB4YV9zc3Bf aHdfcGFyYW1zKHN0cnVjdCBzbmRfcGNtX3N1YnN0cmVhbSAqc3Vic3RyZWFtLAogCXN0cnVjdCBz bmRfc29jX2RhaSAqY3B1X2RhaSA9IHJ0ZC0+ZGFpLT5jcHVfZGFpOwogCXN0cnVjdCBzc3BfcHJp diAqcHJpdiA9IGNwdV9kYWktPnByaXZhdGVfZGF0YTsKIAlzdHJ1Y3Qgc3NwX2RldmljZSAqc3Nw ID0gcHJpdi0+ZGV2LnNzcDsKLQlpbnQgZG1hID0gMCwgY2huID0gcGFyYW1zX2NoYW5uZWxzKHBh cmFtcyk7CisJaW50IGNobiA9IHBhcmFtc19jaGFubmVscyhwYXJhbXMpOwogCXUzMiBzc2NyMDsK IAl1MzIgc3Nwc3A7CiAJaW50IHdpZHRoID0gc25kX3BjbV9mb3JtYXRfcGh5c2ljYWxfd2lkdGgo cGFyYW1zX2Zvcm1hdChwYXJhbXMpKTsKIAlpbnQgdHRzYSA9IHNzcF9yZWFkX3JlZyhzc3AsIFNT VFNBKSAmIDB4ZjsKIAotCS8qIHNlbGVjdCBjb3JyZWN0IERNQSBwYXJhbXMgKi8KLQlpZiAoc3Vi c3RyZWFtLT5zdHJlYW0gIT0gU05EUlZfUENNX1NUUkVBTV9QTEFZQkFDSykKLQkJZG1hID0gMTsg LyogY2FwdHVyZSBETUEgb2Zmc2V0IGlzIDEsMyAqLworCS8qIGdlbmVyYXRlIGNvcnJlY3QgRE1B IHBhcmFtcyAqLworCWlmIChjcHVfZGFpLT5kbWFfZGF0YSkKKwkJa2ZyZWUoY3B1X2RhaS0+ZG1h X2RhdGEpOworCiAJLyogTmV0d29yayBtb2RlIHdpdGggb25lIGFjdGl2ZSBzbG90ICh0dHNhID09 IDEpIGNhbiBiZSB1c2VkCiAJICogdG8gZm9yY2UgMTYtYml0IGZyYW1lIHdpZHRoIG9uIHRoZSB3 aXJlIChmb3IgUzE2X0xFKSwgZXZlbgogCSAqIHdpdGggdHdvIGNoYW5uZWxzLiBVc2UgMTYtYml0 IERNQSB0cmFuc2ZlcnMgZm9yIHRoaXMgY2FzZS4KIAkgKi8KLQlpZiAoKChjaG4gPT0gMikgJiYg KHR0c2EgIT0gMSkpIHx8ICh3aWR0aCA9PSAzMikpCi0JCWRtYSArPSAyOyAvKiAzMi1iaXQgRE1B IG9mZnNldCBpcyAyLCAxNi1iaXQgaXMgMCAqLwotCi0JY3B1X2RhaS0+ZG1hX2RhdGEgPSBzc3Bf ZG1hX3BhcmFtc1tjcHVfZGFpLT5pZF1bZG1hXTsKLQotCWRldl9kYmcoJnNzcC0+cGRldi0+ZGV2 LCAicHhhX3NzcF9od19wYXJhbXM6IGRtYSAlZFxuIiwgZG1hKTsKKwljcHVfZGFpLT5kbWFfZGF0 YSA9IHNzcF9nZXRfZG1hX3BhcmFtcyhzc3AsCisJCQkoKGNobiA9PSAyKSAmJiAodHRzYSAhPSAx KSkgfHwgKHdpZHRoID09IDMyKSwKKwkJCXN1YnN0cmVhbS0+c3RyZWFtID09IFNORFJWX1BDTV9T VFJFQU1fUExBWUJBQ0spOwogCiAJLyogd2UgY2FuIG9ubHkgY2hhbmdlIHRoZSBzZXR0aW5ncyBp ZiB0aGUgcG9ydCBpcyBub3QgaW4gdXNlICovCiAJaWYgKHNzcF9yZWFkX3JlZyhzc3AsIFNTQ1Iw KSAmIFNTQ1IwX1NTRSkKLS0gCjEuNi4wLjQKCg== --0016364eef74c9234b046835b0db--

1 0

[alsa-devel] inserting operations while in rate conversion
by Maris 22 Apr '09

22 Apr '09

I'm running uClinux with ALSA in an Analog Devices' Blackfin CPU environment. Actually I'm checking possibilities in doing special things while ALSA is going to do automatic rate conversion. Specific description: ADC is sampled with 32 or 64 ksps rate. The corresponding hardware driver is specified to operate at one of these rates (which I'll specify later). The application reads the ADC input stream at the output end of the ALSA processing as a stream with a rate of 8 ksps. I need some information about inside details of ALSA's automatic rate conversion (pointers to appropriate information would be helpful). What I want to do: insert a digital anti-aliasing filter algorithm which limits the effective bandwidth of the ADC stream to 4 kHz prior to actually have the rate converted. Of course, I could also add the filter algorithm in the hardware device driver - at the cost of not inherently being available as a hardware independent algorithm. Some detail info: - ADC = AD73311 (speech codec). Basic drivers that allow e.g. arecord & aplay are available. - CPU: convergent DSP/microcontroller core @ 500 MHz and fixed point processing. Thanks for attention. BTW: this is my first dig into ALSA details... Rob Code snippet: (currently only one rate is defined as a correspondence to the hardware driver rate) #define AD73311_RATES (SNDRV_PCM_RATE_8000 | \ SNDRV_PCM_RATE_16000 | \ SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_64000) struct snd_soc_dai ad73311_dai = { .name = "AD73311", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 1, .rates = SNDRV_PCM_RATE_32000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 1, .rates = SNDRV_PCM_RATE_32000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }; EXPORT_SYMBOL_GPL(ad73311_dai);

1 0

[alsa-devel] bug in salsa async handler + patch
by maarten van es 22 Apr '09

22 Apr '09

Hi, I couldn't find where or how to submit patches for salsa, so here goes. This patch resolves random program crashes when using libsalsa with an async handler: snd_async_del_handler does not correctly maintain its list of async handlers. This patch is for salsa-lib-0.0.21, the latest stable version. change src/async.c, line 118 async_list = h; to src/async.c, line 118 async_list = h->next; Maarten

2 1

Re: [alsa-devel] [stable] [PATCH] us122l: add snd_us122l_free()
by Takashi Iwai 22 Apr '09

22 Apr '09

At Tue, 21 Apr 2009 22:20:43 +0000, Chris Wright wrote: > > * Takashi Iwai (tiwai(a)suse.de) wrote: > > At Fri, 17 Apr 2009 13:38:24 +0200, > > Karsten Wiese wrote: > > > > > > Am Freitag 17 April 2009 schrieb Takashi Iwai: > > > > At Thu, 16 Apr 2009 19:38:50 +0200, > > > > Karsten Wiese wrote: > > > > > > > > > > Use it to clean up snd_us122l_card_used[]. > > > > > > > > > > Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> > > > > > Cc: stable(a)kernel.org > > > > > > > > Is it really for stable kernel? Any bug report for this? > > > > > > This really is for stable too. The bug it fixes happens if one does: > > > > > > Start on a pc with onboard soundcard at index 0. > > > $ modprobe snd_usb_us122l modprobe snd_usb_us122l index=2 > > > plug US122L soundcard > > > $ cat /proc/asound/devices > > > last command shows the US122L soundcard is at index 2. > > > > > > unplug US122L soundcard > > > plug US122L soundcard > > > $ cat /proc/asound/devices > > > last command shows the US122L soundcard is at index 1 now, which is the bug > > > the patch fixes. > > > > OK, then please write up the problem to be fixed in the changelog, > > too. > > This looks like it's still outstanding issue. It's already in sound git tree which I'll send a pull request soon later. The relevant commit contains Cc to stable(a)kernel.org, so that it can be handled easily (automatically?)). thanks, Takashi

2 1

[alsa-devel] iMac 9,1 patch work
by Micah Bucy 22 Apr '09

22 Apr '09

I¹m working on a patch for the iMac 9,1 and have the output working. Input is giving me problems. The particular setup is using patch_realtek.c from hda. I noticed that the built-in microphone works if it uses alc883 with model=auto specified. I¹m just not sure what the relevant code is that alc883_auto is using for capture. Any pointers would be helpful. -Micah

2 1

[alsa-devel] [PATCH] ASoC: OMAP: Add checking to detect bufferless pcms
by Joonyoung Shim 22 Apr '09

22 Apr '09

Add checking in hw_params and prepare to detect bufferless pcms(i.e. BT <--> codec). Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/omap/omap-pcm.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 07cf7f4..6454e15 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -87,8 +87,10 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data; int err = 0; + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ if (!dma_data) - return -ENODEV; + return 0; snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); runtime->dma_bytes = params_buffer_bytes(params); @@ -134,6 +136,11 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) struct omap_pcm_dma_data *dma_data = prtd->dma_data; struct omap_dma_channel_params dma_params; + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!prtd->dma_data) + return 0; + memset(&dma_params, 0, sizeof(dma_params)); /* * Note: Regardless of interface data formats supported by OMAP McBSP -- 1.5.6.3

4 6

[alsa-devel] [PATCH v2 0/1] ASoC: TWL4030: Add VDL path support
by Joonyoung Shim 22 Apr '09

22 Apr '09

Hi, I added and modified some DAPMs for VDL path of the twl4030 codec. This patch was posted in the past, but i resend the patch modified more. Please review. Thanks.

1 0

[alsa-devel] Asterisk 1.4.17 and alsa 1.0.19 dmix plugin problem.
by Nicola Mfb 21 Apr '09

21 Apr '09

Hi! I'm sorry if this is an old issue but I did not found help in archives/alsaweb/google and I'm new to alsa, so this report is what I guessed reading sources and compiling test apps for about 2 days, and the following assumptions may be completely wrong. Asterisk has an alsa channel that works fine when using plughw:0,0 both for read and for writing, but if use plug:dmix/plug:dsnoop or default with a dmix/dsnoop wrapper in /etc/asound.conf it does not emit any sound. It uses select on poll fd descriptor to know if the pcm is ready to be sent output samples. The select will never return as it seems the dmix plugin is not able to signal when is ready. I noted that using snd_pcm_wait works well but cannot be used in asterisk as it needs to select on two fd. I guess that snd_pcm_wait with dmix check the shared memory and not the fd descriptor. So should poll/select be avoided completely in applications (as they does not know if the default pcm is a dmix) and asterisk alsa channel be rewritten or is there a workaround? Best Regards Nicola

2 5

Re: [alsa-devel] Cirrus Logic CS4207
by Takashi Iwai 21 Apr '09

21 Apr '09

At Tue, 21 Apr 2009 09:24:31 +0200, Peter-Jan Deweirdt wrote: > > Hi, > > This is the patch that I've changed to the 2.6.29 kernel. However I > need some more help for playing the audio. Since I don't really know > what is happening. > > It's detecting my audio chip. (a part of dmesg) > [ 5.171478] Advanced Linux Sound Architecture Driver Version 1.0.18a. > [ 5.176635] HDA Intel 0000:00:1b.0: setting latency timer to 64 > [ 5.220417] ALSA device list: > [ 5.220434] #0: HDA Intel MID at 0xd0010000 irq 22 > > So it looks like my audio card is found. Some more outputs > > lspci -v > 00:1b.0 Audio device: Intel Corporation System Controller Hub (SCH > Poulsbo) HD Audio Controller (rev 06) > Subsystem: Intel Corporation Unknown device 8119 > Flags: bus master, fast devsel, latency 0, IRQ 22 > Memory at d0010000 (64-bit, non-prefetchable) [size=16K] > Capabilities: [50] Power Management version 2 > Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 > Kernel driver in use: HDA Intel > > > [root@localhost ~]# cat /proc/asound/cards > 0 [MID ]: HDA-Intel - HDA Intel MID > HDA Intel MID at 0xd0010000 irq 22 > [root@localhost ~]# > > [root@localhost ~]# aplay -l > **** List of PLAYBACK Hardware Devices **** > card 0: MID [HDA Intel MID], device 0: CS4207 Analog [CS4207 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: MID [HDA Intel MID], device 1: CS4207 Digital [CS4207 Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > [root@localhost ~]# > > > [root@localhost ~]# aplay -L > front:CARD=MID,DEV=0 > HDA Intel MID, CS4207 Analog > Front speakers > surround40:CARD=MID,DEV=0 > HDA Intel MID, CS4207 Analog > 4.0 Surround output to Front and Rear speakers > surround41:CARD=MID,DEV=0 > HDA Intel MID, CS4207 Analog > 4.1 Surround output to Front, Rear and Subwoofer speakers > surround50:CARD=MID,DEV=0 > HDA Intel MID, CS4207 Analog > 5.0 Surround output to Front, Center and Rear speakers > surround51:CARD=MID,DEV=0 > HDA Intel MID, CS4207 Analog > 5.1 Surround output to Front, Center, Rear and Subwoofer speakers > surround71:CARD=MID,DEV=0 > HDA Intel MID, CS4207 Analog > 7.1 Surround output to Front, Center, Side, Rear and Woofer > speakers > iec958:CARD=MID,DEV=0 > HDA Intel MID, CS4207 Digital > IEC958 (S/PDIF) Digital Audio Output > null > Discard all samples (playback) or generate zero samples (capture) > pulse > PulseAudio Sound Server > default > Default > [root@localhost ~]# > > [root@localhost ~]# ls -al /dev/snd/ > total 0 > drwxr-xr-x 2 root root 200 Mar 24 20:00 . > drwxr-xr-x 15 root root 3360 Mar 24 20:00 .. > crw-rw---- 1 root audio 116, 9 Mar 24 20:00 controlC0 > crw-rw---- 1 root audio 116, 8 Mar 24 20:00 hwC0D0 > crw-rw---- 1 root audio 116, 7 Mar 24 20:00 pcmC0D0c > crw-rw---- 1 root audio 116, 6 Mar 24 20:58 pcmC0D0p > crw-rw---- 1 root audio 116, 5 Mar 24 20:00 pcmC0D1c > crw-rw---- 1 root audio 116, 4 Mar 24 20:30 pcmC0D1p > crw-rw---- 1 root audio 116, 3 Mar 24 20:00 seq > crw-rw---- 1 root audio 116, 2 Mar 24 20:00 timer > [root@localhost ~]# All look good, so far. > The problems that I'm having is: > - How to play audio? % aplay foo.wav > - I am seeing 2 playback devices (what is correct) but no headphones The streams are just for analog and digital outputs. The analog outputs are switched either by the automatic HP jack detection or manual switch (if any). > - How can I see the difference between the MIC in and LINE IN? Via mixer controls. Takashi

2 3

[alsa-devel] [PATCH] AOA: Convert onyx and tas codecs to new-style i2c drivers
by Jean Delvare 21 Apr '09

21 Apr '09

The legacy i2c binding model is going away soon, so convert the AOA codec drivers to the new model or they'll break. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Tested-by: Johannes Berg <johannes(a)sipsolutions.net> Tested-by: Andreas Schwab <schwab(a)linux-m68k.org> Cc: Benjamin Herrenschmidt <benh(a)kernel.crashing.org> Cc: Takashi Iwai <tiwai(a)suse.de> --- Takashi, please push this patch to Linus quickly, as this is blocking the removal of the legacy i2c binding model, which is scheduled for 2.6.30. sound/aoa/codecs/onyx.c | 76 ++++++++++++++++++++++++++++++++--------------- sound/aoa/codecs/tas.c | 66 +++++++++++++++++++++++++--------------- 2 files changed, 95 insertions(+), 47 deletions(-) --- linux-2.6.30-rc2.orig/sound/aoa/codecs/onyx.c 2009-04-15 09:55:20.000000000 +0200 +++ linux-2.6.30-rc2/sound/aoa/codecs/onyx.c 2009-04-15 14:07:50.000000000 +0200 @@ -47,7 +47,7 @@ MODULE_DESCRIPTION("pcm3052 (onyx) codec struct onyx { /* cache registers 65 to 80, they are write-only! */ u8 cache[16]; - struct i2c_client i2c; + struct i2c_client *i2c; struct aoa_codec codec; u32 initialised:1, spdif_locked:1, @@ -72,7 +72,7 @@ static int onyx_read_register(struct ony *value = onyx->cache[reg-FIRSTREGISTER]; return 0; } - v = i2c_smbus_read_byte_data(&onyx->i2c, reg); + v = i2c_smbus_read_byte_data(onyx->i2c, reg); if (v < 0) return -1; *value = (u8)v; @@ -84,7 +84,7 @@ static int onyx_write_register(struct on { int result; - result = i2c_smbus_write_byte_data(&onyx->i2c, reg, value); + result = i2c_smbus_write_byte_data(onyx->i2c, reg, value); if (!result) onyx->cache[reg-FIRSTREGISTER] = value; return result; @@ -996,12 +996,45 @@ static void onyx_exit_codec(struct aoa_c onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx); } -static struct i2c_driver onyx_driver; - static int onyx_create(struct i2c_adapter *adapter, struct device_node *node, int addr) { + struct i2c_board_info info; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + strlcpy(info.type, "aoa_codec_onyx", I2C_NAME_SIZE); + info.addr = addr; + info.platform_data = node; + client = i2c_new_device(adapter, &info); + if (!client) + return -ENODEV; + + /* + * We know the driver is already loaded, so the device should be + * already bound. If not it means binding failed, which suggests + * the device doesn't really exist and should be deleted. + * Ideally this would be replaced by better checks _before_ + * instantiating the device. + */ + if (!client->driver) { + i2c_unregister_device(client); + return -ENODEV; + } + + /* + * Let i2c-core delete that device on driver removal. + * This is safe because i2c-core holds the core_lock mutex for us. + */ + list_add_tail(&client->detected, &client->driver->clients); + return 0; +} + +static int onyx_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device_node *node = client->dev.platform_data; struct onyx *onyx; u8 dummy; @@ -1011,20 +1044,12 @@ static int onyx_create(struct i2c_adapte return -ENOMEM; mutex_init(&onyx->mutex); - onyx->i2c.driver = &onyx_driver; - onyx->i2c.adapter = adapter; - onyx->i2c.addr = addr & 0x7f; - strlcpy(onyx->i2c.name, "onyx audio codec", I2C_NAME_SIZE); - - if (i2c_attach_client(&onyx->i2c)) { - printk(KERN_ERR PFX "failed to attach to i2c\n"); - goto fail; - } + onyx->i2c = client; + i2c_set_clientdata(client, onyx); /* we try to read from register ONYX_REG_CONTROL * to check if the codec is present */ if (onyx_read_register(onyx, ONYX_REG_CONTROL, &dummy) != 0) { - i2c_detach_client(&onyx->i2c); printk(KERN_ERR PFX "failed to read control register\n"); goto fail; } @@ -1036,14 +1061,14 @@ static int onyx_create(struct i2c_adapte onyx->codec.node = of_node_get(node); if (aoa_codec_register(&onyx->codec)) { - i2c_detach_client(&onyx->i2c); goto fail; } printk(KERN_DEBUG PFX "created and attached onyx instance\n"); return 0; fail: + i2c_set_clientdata(client, NULL); kfree(onyx); - return -EINVAL; + return -ENODEV; } static int onyx_i2c_attach(struct i2c_adapter *adapter) @@ -1080,28 +1105,33 @@ static int onyx_i2c_attach(struct i2c_ad return onyx_create(adapter, NULL, 0x47); } -static int onyx_i2c_detach(struct i2c_client *client) +static int onyx_i2c_remove(struct i2c_client *client) { - struct onyx *onyx = container_of(client, struct onyx, i2c); - int err; + struct onyx *onyx = i2c_get_clientdata(client); - if ((err = i2c_detach_client(client))) - return err; aoa_codec_unregister(&onyx->codec); of_node_put(onyx->codec.node); if (onyx->codec_info) kfree(onyx->codec_info); + i2c_set_clientdata(client, onyx); kfree(onyx); return 0; } +static const struct i2c_device_id onyx_i2c_id[] = { + { "aoa_codec_onyx", 0 }, + { } +}; + static struct i2c_driver onyx_driver = { .driver = { .name = "aoa_codec_onyx", .owner = THIS_MODULE, }, .attach_adapter = onyx_i2c_attach, - .detach_client = onyx_i2c_detach, + .probe = onyx_i2c_probe, + .remove = onyx_i2c_remove, + .id_table = onyx_i2c_id, }; static int __init onyx_init(void) --- linux-2.6.30-rc2.orig/sound/aoa/codecs/tas.c 2009-04-15 09:55:20.000000000 +0200 +++ linux-2.6.30-rc2/sound/aoa/codecs/tas.c 2009-04-15 09:56:39.000000000 +0200 @@ -82,7 +82,7 @@ MODULE_DESCRIPTION("tas codec driver for struct tas { struct aoa_codec codec; - struct i2c_client i2c; + struct i2c_client *i2c; u32 mute_l:1, mute_r:1 , controls_created:1 , drc_enabled:1, @@ -108,9 +108,9 @@ static struct tas *codec_to_tas(struct a static inline int tas_write_reg(struct tas *tas, u8 reg, u8 len, u8 *data) { if (len == 1) - return i2c_smbus_write_byte_data(&tas->i2c, reg, *data); + return i2c_smbus_write_byte_data(tas->i2c, reg, *data); else - return i2c_smbus_write_i2c_block_data(&tas->i2c, reg, len, data); + return i2c_smbus_write_i2c_block_data(tas->i2c, reg, len, data); } static void tas3004_set_drc(struct tas *tas) @@ -882,12 +882,34 @@ static void tas_exit_codec(struct aoa_co } -static struct i2c_driver tas_driver; - static int tas_create(struct i2c_adapter *adapter, struct device_node *node, int addr) { + struct i2c_board_info info; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + strlcpy(info.type, "aoa_codec_tas", I2C_NAME_SIZE); + info.addr = addr; + info.platform_data = node; + + client = i2c_new_device(adapter, &info); + if (!client) + return -ENODEV; + + /* + * Let i2c-core delete that device on driver removal. + * This is safe because i2c-core holds the core_lock mutex for us. + */ + list_add_tail(&client->detected, &client->driver->clients); + return 0; +} + +static int tas_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device_node *node = client->dev.platform_data; struct tas *tas; tas = kzalloc(sizeof(struct tas), GFP_KERNEL); @@ -896,17 +918,11 @@ static int tas_create(struct i2c_adapter return -ENOMEM; mutex_init(&tas->mtx); - tas->i2c.driver = &tas_driver; - tas->i2c.adapter = adapter; - tas->i2c.addr = addr; + tas->i2c = client; + i2c_set_clientdata(client, tas); + /* seems that half is a saner default */ tas->drc_range = TAS3004_DRC_MAX / 2; - strlcpy(tas->i2c.name, "tas audio codec", I2C_NAME_SIZE); - - if (i2c_attach_client(&tas->i2c)) { - printk(KERN_ERR PFX "failed to attach to i2c\n"); - goto fail; - } strlcpy(tas->codec.name, "tas", MAX_CODEC_NAME_LEN); tas->codec.owner = THIS_MODULE; @@ -915,14 +931,12 @@ static int tas_create(struct i2c_adapter tas->codec.node = of_node_get(node); if (aoa_codec_register(&tas->codec)) { - goto detach; + goto fail; } printk(KERN_DEBUG "snd-aoa-codec-tas: tas found, addr 0x%02x on %s\n", - addr, node->full_name); + (unsigned int)client->addr, node->full_name); return 0; - detach: - i2c_detach_client(&tas->i2c); fail: mutex_destroy(&tas->mtx); kfree(tas); @@ -970,14 +984,11 @@ static int tas_i2c_attach(struct i2c_ada return -ENODEV; } -static int tas_i2c_detach(struct i2c_client *client) +static int tas_i2c_remove(struct i2c_client *client) { - struct tas *tas = container_of(client, struct tas, i2c); - int err; + struct tas *tas = i2c_get_clientdata(client); u8 tmp = TAS_ACR_ANALOG_PDOWN; - if ((err = i2c_detach_client(client))) - return err; aoa_codec_unregister(&tas->codec); of_node_put(tas->codec.node); @@ -989,13 +1000,20 @@ static int tas_i2c_detach(struct i2c_cli return 0; } +static const struct i2c_device_id tas_i2c_id[] = { + { "aoa_codec_tas", 0 }, + { } +}; + static struct i2c_driver tas_driver = { .driver = { .name = "aoa_codec_tas", .owner = THIS_MODULE, }, .attach_adapter = tas_i2c_attach, - .detach_client = tas_i2c_detach, + .probe = tas_i2c_probe, + .remove = tas_i2c_remove, + .id_table = tas_i2c_id, }; static int __init tas_init(void) -- Jean Delvare

3 4

Re: [alsa-devel] [PATCH] keywest: Convert to new-style i2c driver
by Takashi Iwai 21 Apr '09

21 Apr '09

At Tue, 21 Apr 2009 19:33:00 +1000, Paul Mackerras wrote: > > Takashi Iwai writes: > > > At least, the conversion patch Jean posted can be in 2.6.30, I think. > > Really? What regression, security hole or serious bug are you going > to tell Linus that it fixes? :) Build warning fixes :) Takashi

1 0

Re: [alsa-devel] 2.6.29 regression: left audio channel broken after resume from suspend with Intel HDA
by Takashi Iwai 21 Apr '09

21 Apr '09

At Tue, 21 Apr 2009 09:32:19 +0200, Tino Keitel wrote: > > On Tue, Apr 21, 2009 at 07:24:40 +0200, Takashi Iwai wrote: > > The commit is fcad94a4c71c36a05f4d5c6dcb174534b4e0b136: > > ALSA: hda - Fix the cmd cache keys for amp verbs > > Thanks for the reply. However, the patch doesn't work. I just got > distorted sound again. Then please run alsa-info.sh (with --no-upload option) before and after suspend, and attach the generated files. Also, check the latest Linus tree whether the problem still appears. thanks, Takashi

1 0

[alsa-devel] Getting M-Audio JamLab simple controls to work
by Greg Erwin 21 Apr '09

21 Apr '09

I have an M-Audio JamLab usb audio interface. Long story short, it works for the most part, but trying to use the simple controls via amixer or alsamixer don't work. I've filed a bug report about it a little while back that has quite a bit of information in it: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4415 Anyway, I feel like it would't take a whole lot to get working (I hope), but the thing is, I bet very few people actually have one. I know a little bit of C, and will be glad to help as much as I can.

2 1

Re: [alsa-devel] 2.6.29 regression: left audio channel broken after resume from suspend with Intel HDA
by Takashi Iwai 21 Apr '09

21 Apr '09

At Mon, 20 Apr 2009 21:46:59 +0200, Tino Keitel wrote: > > On Mon, Apr 20, 2009 at 21:41:06 +0200, Takashi Iwai wrote: > > [...] > > > Could you try the latest Linus git tree? A fix patch was merged > > after 2.6.30-rc2. The patch will be merged to later stable kernel, too. > > What commit is this? Can it be applied to 2.6.29? As it isn't perfectly > reproducable, I'd like to run 2.6.29 with the patch applied and see if > the problem disappears. The commit is fcad94a4c71c36a05f4d5c6dcb174534b4e0b136: ALSA: hda - Fix the cmd cache keys for amp verbs Takashi

1 0

[alsa-devel] PS3 Eye Microphone support
by Mihai Tanasescu 21 Apr '09

21 Apr '09

Hello guys, I don't know if this is the proper list to address the question but here it goes and sorry if I'm wrong. I wanted to ask if there is any support for the Playstation3 Eye USB Camera Microphone provided by alsa at the moment or if there are plans to support it. Currently from what I could figure out: root@Mihai:~# cat /proc/asound/cards 0 [Audigy ]: Audigy - Audigy 1 [SB0090] Audigy 1 [SB0090] (rev.3, serial:0x531102) at 0xb880, irq 23 1 [CameraB304061 ]: USB-Audio - USB Camera-B3.04.06.1 OmniVision Technologies, Inc. USB Camera-B3.04.06.1 at usb-0000:00:1d.7-5.1, hi the Microphone Camera is seen there. I successfully recorded something using Audacity and the OSS driver that was reported in the recording options. What I can't do is control the sound levels via the gnome volume control applet (which uses alsactl store and restore). Also, alsamixer won't work. root@Mihai:~# alsactl store alsactl: get_control:262: Cannot read control '2,0,0,Mic Capture Volume,0': Invalid argument root@Mihai:~# alsamixer -c 1 alsamixer: function snd_mixer_load failed: Invalid argument Cheers, Mihai

1 0

Re: [alsa-devel] 2.6.29 regression: left audio channel broken after resume from suspend with Intel HDA
by Takashi Iwai 20 Apr '09

20 Apr '09

At Mon, 20 Apr 2009 20:40:24 +0200, Tino Keitel wrote: > > Hi, > > since 2.6.29 and also with 2.6.29.1 the left audio channel of the > digital output of my Mac mini Core 2 Duo is noisy. This didn't happen > with 2.6.28. Unloading snd-hda-intel and reloading it does fix the > noisy sound. > > This happened with some older versions of the Linux kernel (with an > older Mac mini Core Duo, I don't remeber which kernel version), but got > fixed at some point. Now it happens again with 2.6.29. Could you try the latest Linus git tree? A fix patch was merged after 2.6.30-rc2. The patch will be merged to later stable kernel, too. Takashi

1 0

Re: [alsa-devel] 2.6.29 regression: left audio channel broken after resume from suspend with Intel HDA
by Rafael J. Wysocki 20 Apr '09

20 Apr '09

On Monday 20 April 2009, Tino Keitel wrote: > Hi, > > since 2.6.29 and also with 2.6.29.1 the left audio channel of the > digital output of my Mac mini Core 2 Duo is noisy. This didn't happen > with 2.6.28. Unloading snd-hda-intel and reloading it does fix the > noisy sound. > > This happened with some older versions of the Linux kernel (with an > older Mac mini Core Duo, I don't remeber which kernel version), but got > fixed at some point. Now it happens again with 2.6.29. > > Here is the dmesg output for the sound hardware: > > HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 > HDA Intel 0000:00:1b.0: setting latency timer to 64 > hda_codec: STAC922x, Apple subsys_id=100 > input: HDA Intel Line In at Ext Left Jack as /class/input/input18 > input: HDA Intel Mic at Ext Left Jack as /class/input/input19 > input: HDA Intel HP Out at Ext Left Jack as /class/input/input20 > > Any hints what I could try to fix this, maybe some commit that I should > try to revert? Well, no. Still, you might try the current Linus' tree. Best, Rafael

1 0

[alsa-devel] vi-like key bindings for alsamixer
by Simon Hengel 20 Apr '09

20 Apr '09

Hello List, I realized that 'j' and 'k' are not used nor reserved for future use with alsamixer. So I decided to use them as aliases for 'up' and 'down'. Is this deemed suitable for the masses? Cheers, Simon Hengel

4 6

[alsa-devel] No sound with Creative Sound Blaster X-Fi Xtreme Audio Notebook (CA0110-IBG)
by Ivan Mironov 20 Apr '09

20 Apr '09

Hello. I'm trying to get sound from my Creative Sound Blaster X-Fi Xtreme Audio Notebook (it's expresscard) with fresh ALSA drivers (I tried both alsa-driver-snapshot and alsa-driver-unstable-snapshot from 16.04.2009). I can change mixer values in alsamixer, but no sound output. For example: ------------------------------------------------------------ $ speaker-test -D plughw:1,0 speaker-test 1.0.19 Playback device is plughw:1,0 Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Playback open error: -22,Invalid argument Playback open error: -22,Invalid argument Playback open error: -22,Invalid argument ^C ------------------------------------------------------------ at this time ALSA kernel driver gives an error in dmesg: ------------------------------------------------------------ ... ALSA /home/ivan/stuff/alsa-driver/acore/pcm_native.c:2040: snd_pcm_hw_constraints_complete failed ALSA /home/ivan/stuff/alsa-driver/acore/pcm_native.c:2040: snd_pcm_hw_constraints_complete failed ALSA /home/ivan/stuff/alsa-driver/acore/pcm_native.c:2040: snd_pcm_hw_constraints_complete failed ... ------------------------------------------------------------ What should I do? Here is some information: ------------------------------------------------------------ $ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.19. Compiled on Apr 17 2009 for kernel 2.6.29.1 (SMP). $ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf3600000 irq 22 1 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xc2000000 irq 16 ------------------------------------------------------------ ------------------------------------------------------------ 03:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG Subsystem: Creative Labs Device 0018 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (500ns min, 5000ns max) Interrupt: pin A routed to IRQ 16 Region 0: Memory at c2000000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel ------------------------------------------------------------ Information from the alsa-info.sh: http://www.alsa-project.org/db/?f=a2683b00960fd3d8d32c44cc360aa0df5c413bdc Kernel log in the attachment. Sorry for my terrible english.

2 8

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 20 Apr '09

20 Apr '09

The following changes since commit 8d98f2246d7c622198ae0b8ca66f1c82b8a25377: Mark Brown (1): Merge branch 'for-2.6.30' into for-2.6.31 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Jarkko Nikula (1): ASoC: OMAP: Update contact addresses Joonyoung Shim (1): ASoC: TWL4030: Add support Voice DAI Mark Brown (1): Merge branch 'for-2.6.30' into for-2.6.31 Peter Ujfalusi (1): ASoC: TWL4030: Fix for the constraint handling Philipp Zabel (1): ASoC: pxa-ssp: Don't use SSCR0_SerClkDiv and SSCR0_SCR sound/soc/codecs/twl4030.c | 258 +++++++++++++++++++++++++++++++++++++---- sound/soc/codecs/twl4030.h | 18 +++- sound/soc/omap/n810.c | 4 +- sound/soc/omap/omap-mcbsp.c | 5 +- sound/soc/omap/omap-mcbsp.h | 3 +- sound/soc/omap/omap-pcm.c | 5 +- sound/soc/omap/omap-pcm.h | 3 +- sound/soc/omap/omap2evm.c | 2 +- sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/omap3pandora.c | 4 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/sdp3430.c | 2 +- sound/soc/pxa/pxa-ssp.c | 36 +++++-- 13 files changed, 297 insertions(+), 47 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 20 Apr '09

20 Apr '09

The following changes since commit 002fbad829883a3ad9b185077f9e54c37dc7502d: Jarkko Nikula (1): ASoC: OMAP: Fix FS polarity in OSK5912 machine driver are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Jarkko Nikula (1): ASoC: OMAP: Update contact addresses Philipp Zabel (1): ASoC: pxa-ssp: Don't use SSCR0_SerClkDiv and SSCR0_SCR Russell King - ARM Linux (1): ASoC: Fix warning in wm9705 sound/soc/codecs/wm9705.c | 2 +- sound/soc/omap/n810.c | 4 ++-- sound/soc/omap/omap-mcbsp.c | 5 +++-- sound/soc/omap/omap-mcbsp.h | 3 ++- sound/soc/omap/omap-pcm.c | 5 +++-- sound/soc/omap/omap-pcm.h | 3 ++- sound/soc/pxa/pxa-ssp.c | 36 ++++++++++++++++++++++++++++-------- 7 files changed, 41 insertions(+), 17 deletions(-)

2 1

[alsa-devel] [PATCH] Atiixp: Add SSID for mute_led quirk (unknown HP model)
by Daniel T Chen 20 Apr '09

20 Apr '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Subject says it all. Briefly, use the mute_led quirk for another HP model. Thanks, Dan -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) iD8DBQFJ7FfZe9GwFciKvaMRAleaAJ9mmtLUW7HgwjZCHkV1Mk4/dbV+MgCgjDEm ZD/KsZbyDYJB4Egq+fzDXc8= =qaDE -----END PGP SIGNATURE-----

2 1

[alsa-devel] [PATCH v2] us122l: add snd_us122l_free()
by Karsten Wiese 20 Apr '09

20 Apr '09

Use it to clean up snd_us122l_card_used[]. Without patch unplugging of an US122L soundcard didn't reset the corresponding element of snd_us122l_card_used[] to 0. The (SNDRV_CARDS + 1)th plugging in did not result in creating the soundcard device anymore. Index values supplied with the modprobe command line were not used correctly anymore after the first unplugging of an US122L. Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> Cc: stable(a)kernel.org --- sound/usb/usx2y/us122l.c | 16 ++++++++++++---- 1 files changed, 12 insertions(+), 4 deletions(-) Hi Takashi, this is Version 2 of the patch with a better changelog. Good enough? Thanks, Karsten diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 012ff1f..e0385ce 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2007, 2008 Karsten Wiese <fzu(a)wemgehoertderstaat.de> + * Copyright (C) 2007-2009 Karsten Wiese <fzu(a)wemgehoertderstaat.de> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -26,7 +26,7 @@ #include "us122l.h" MODULE_AUTHOR("Karsten Wiese <fzu(a)wemgehoertderstaat.de>"); -MODULE_DESCRIPTION("TASCAM "NAME_ALLCAPS" Version 0.5"); +MODULE_DESCRIPTION("TASCAM "NAME_ALLCAPS" Version 0.6"); MODULE_LICENSE("GPL"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ @@ -474,6 +474,14 @@ static bool us122l_create_card(struct snd_card *card) return true; } +static void snd_us122l_free(struct snd_card *card) +{ + struct us122l *us122l = US122L(card); + int index = us122l->chip.index; + if (index >= 0 && index < SNDRV_CARDS) + snd_us122l_card_used[index] = 0; +} + static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) { int dev; @@ -490,7 +498,7 @@ static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) if (err < 0) return err; snd_us122l_card_used[US122L(card)->chip.index = dev] = 1; - + card->private_free = snd_us122l_free; US122L(card)->chip.dev = device; US122L(card)->chip.card = card; mutex_init(&US122L(card)->mutex); @@ -584,7 +592,7 @@ static void snd_us122l_disconnect(struct usb_interface *intf) } usb_put_intf(intf); - usb_put_dev(US122L(card)->chip.dev); + usb_put_dev(us122l->chip.dev); while (atomic_read(&us122l->mmap_count)) msleep(500); -- 1.6.0.6

2 3

[alsa-devel] [PATCH v2 1/1] ASoC: TWL4030: Add support Voice DAI
by Joonyoung Shim 20 Apr '09

20 Apr '09

Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 173 +++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/twl4030.h | 18 ++++- sound/soc/omap/omap2evm.c | 2 +- sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/omap3pandora.c | 4 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/sdp3430.c | 2 +- 7 files changed, 191 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index a1b76d7..cc2968c 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1484,6 +1484,144 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int twl4030_voice_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u8 infreq; + u8 mode; + + /* If the system master clock is not 26MHz, the voice PCM interface is + * not avilable. + */ + infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL) + & TWL4030_APLL_INFREQ; + + if (infreq != TWL4030_APLL_INFREQ_26000KHZ) { + printk(KERN_ERR "TWL4030 voice startup: " + "MCLK is not 26MHz, call set_sysclk() on init\n"); + return -EINVAL; + } + + /* If the codec mode is not option2, the voice PCM interface is not + * avilable. + */ + mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) + & TWL4030_OPT_MODE; + + if (mode != TWL4030_OPTION_2) { + printk(KERN_ERR "TWL4030 voice startup: " + "the codec mode is not option2\n"); + return -EINVAL; + } + + return 0; +} + +static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u8 old_mode, mode; + + /* bit rate */ + old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) + & ~(TWL4030_CODECPDZ); + mode = old_mode; + + switch (params_rate(params)) { + case 8000: + mode &= ~(TWL4030_SEL_16K); + break; + case 16000: + mode |= TWL4030_SEL_16K; + break; + default: + printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n", + params_rate(params)); + return -EINVAL; + } + + if (mode != old_mode) { + /* change rate and set CODECPDZ */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_codec_enable(codec, 1); + } + + return 0; +} + +static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 infreq; + + switch (freq) { + case 26000000: + infreq = TWL4030_APLL_INFREQ_26000KHZ; + break; + default: + printk(KERN_ERR "TWL4030 voice set sysclk: unknown rate %d\n", + freq); + return -EINVAL; + } + + infreq |= TWL4030_APLL_EN; + twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq); + + return 0; +} + +static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 old_format, format; + + /* get format */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF); + format = old_format; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFM: + format &= ~(TWL4030_VIF_SLAVE_EN); + break; + case SND_SOC_DAIFMT_CBS_CFS: + format |= TWL4030_VIF_SLAVE_EN; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + format &= ~(TWL4030_VIF_FORMAT); + break; + case SND_SOC_DAIFMT_NB_IF: + format |= TWL4030_VIF_FORMAT; + break; + default: + return -EINVAL; + } + + if (format != old_format) { + /* change format and set CODECPDZ */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_VOICE_IF, format); + twl4030_codec_enable(codec, 1); + } + + return 0; +} + #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) @@ -1495,7 +1633,15 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { .set_fmt = twl4030_set_dai_fmt, }; -struct snd_soc_dai twl4030_dai = { +static struct snd_soc_dai_ops twl4030_dai_voice_ops = { + .startup = twl4030_voice_startup, + .hw_params = twl4030_voice_hw_params, + .set_sysclk = twl4030_voice_set_dai_sysclk, + .set_fmt = twl4030_voice_set_dai_fmt, +}; + +struct snd_soc_dai twl4030_dai[] = { +{ .name = "twl4030", .playback = { .stream_name = "Playback", @@ -1510,6 +1656,23 @@ struct snd_soc_dai twl4030_dai = { .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, .ops = &twl4030_dai_ops, +}, +{ + .name = "twl4030 Voice", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &twl4030_dai_voice_ops, +}, }; EXPORT_SYMBOL_GPL(twl4030_dai); @@ -1550,8 +1713,8 @@ static int twl4030_init(struct snd_soc_device *socdev) codec->read = twl4030_read_reg_cache; codec->write = twl4030_write; codec->set_bias_level = twl4030_set_bias_level; - codec->dai = &twl4030_dai; - codec->num_dai = 1; + codec->dai = twl4030_dai; + codec->num_dai = ARRAY_SIZE(twl4030_dai), codec->reg_cache_size = sizeof(twl4030_reg); codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), GFP_KERNEL); @@ -1645,13 +1808,13 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); static int __init twl4030_modinit(void) { - return snd_soc_register_dai(&twl4030_dai); + return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); } module_init(twl4030_modinit); static void __exit twl4030_exit(void) { - snd_soc_unregister_dai(&twl4030_dai); + snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); } module_exit(twl4030_exit); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index cb63765..981ec60 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -113,6 +113,8 @@ #define TWL4030_SEL_16K 0x04 #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 +#define TWL4030_OPTION_1 (1 << 0) +#define TWL4030_OPTION_2 (0 << 0) /* TWL4030_REG_MICBIAS_CTL (0x04) Fields */ @@ -171,6 +173,17 @@ #define TWL4030_CLK256FS_EN 0x02 #define TWL4030_AIF_EN 0x01 +/* VOICE_IF (0x0F) Fields */ + +#define TWL4030_VIF_SLAVE_EN 0x80 +#define TWL4030_VIF_DIN_EN 0x40 +#define TWL4030_VIF_DOUT_EN 0x20 +#define TWL4030_VIF_SWAP 0x10 +#define TWL4030_VIF_FORMAT 0x08 +#define TWL4030_VIF_TRI_EN 0x04 +#define TWL4030_VIF_SUB_EN 0x02 +#define TWL4030_VIF_EN 0x01 + /* EAR_CTL (0x21) */ #define TWL4030_EAR_GAIN 0x30 @@ -236,7 +249,10 @@ #define TWL4030_SMOOTH_ANAVOL_EN 0x02 #define TWL4030_DIGMIC_LR_SWAP_EN 0x01 -extern struct snd_soc_dai twl4030_dai; +#define TWL4030_DAI_HIFI 0 +#define TWL4030_DAI_VOICE 1 + +extern struct snd_soc_dai twl4030_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_twl4030; #endif /* End of __TWL4030_AUDIO_H__ */ diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c index 0c2322d..027e1a4 100644 --- a/sound/soc/omap/omap2evm.c +++ b/sound/soc/omap/omap2evm.c @@ -86,7 +86,7 @@ static struct snd_soc_dai_link omap2evm_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap2evm_ops, }; diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index fd24a4a..6aa428e 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link omap3beagle_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3beagle_ops, }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index fe282d4..ad219aa 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -228,14 +228,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { .name = "PCM1773", .stream_name = "HiFi Out", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3pandora_out_ops, .init = omap3pandora_out_init, }, { .name = "TWL4030", .stream_name = "Line/Mic In", .cpu_dai = &omap_mcbsp_dai[1], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3pandora_in_ops, .init = omap3pandora_in_init, } diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index a72dc4e..ec4f8fd 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link overo_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &overo_ops, }; diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 10f1c86..1c79741 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -197,7 +197,7 @@ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, }; -- 1.5.6.3

3 2

[alsa-devel] [PATCH 0/1] ASoC: TWL4030: Constraint setting rework
by Peter Ujfalusi 20 Apr '09

20 Apr '09

Hello, The current implementation of constraints in TWL4030 codec needs some tuning. An interesting case can be generated by using gstreamer: gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false What would (since the current implementation blocks it) happen in this case is: playback stream: open capture stream: open capture stream: hw_params playback_stream: hw_params In capture open we set the constraints to the capture stream based on the playback streams runtime parameters, and they are 0 (rate = 0, sample_bits = 0), since the hw_param has not been called for that stream yet... The following patch fixes this issue among other possible scenarios regarding to constraint handling. Note that I'm not using the just introduced symmetric_rates feature of ASoC introduced by Mark Brown recently, since I think that also suffers from this problem (or not, I have to try it out first to be sure about it) --- Peter Ujfalusi (1): ASoC: TWL4030: Fix for the constraint handling sound/soc/codecs/twl4030.c | 85 ++++++++++++++++++++++++++++++++++---------- 1 files changed, 66 insertions(+), 19 deletions(-)

2 5

[alsa-devel] [PATCH v2 0/1] ASoC: TWL4030: Add support Voice DAI
by Joonyoung Shim 20 Apr '09

20 Apr '09

Hi, I added Voice DAI to support the PCM voice interface of the twl4030 codec. I post this patch modified through the reviews Please review again. Thanks.

1 0

[alsa-devel] Speaker and Headphone Output problem with ALC1200
by Thomas Pietrzak 20 Apr '09

20 Apr '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have a MSI GX720-061 laptop, and I unfortunately experiment the "Speaker and Headphone Output" problem. The sound card is the following: $ lspci |grep Audio 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) $ aplay -l **** Liste des PLAYBACK périphériques **** carte 0: Intel [HDA Intel], périphérique 0 : ALC1200 Analog [ALC1200 Analog] Sous-périphériques: 0/1 Sous-périphérique: #0: subdevice #0 carte 0: Intel [HDA Intel], périphérique 1 : ALC1200 Digital [ALC1200 Digital] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 carte 0: Intel [HDA Intel], périphérique 6 : Si3054 Modem [Si3054 Modem] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 Note that in Ubuntu Intrepid it was recognized as ALC888. However my problem remains in Jaunty. These are the alsa packages I have (Ubuntu Jaunty): $ dpkg -l |grep alsa ii alsa-base 1.0.18.dfsg-1ubuntu6 ALSA driver configuration files ii alsa-oss 1.0.17-1 ALSA wrapper for OSS applications ii alsa-tools 1.0.18-1ubuntu2 Console based ALSA utilities for specific ha ii alsa-tools-gui 1.0.18-1ubuntu2 GUI based ALSA utilities for specific hardwa ii alsa-utils 1.0.18-1ubuntu10 ALSA utilities ii alsamixergui 0.9.0rc2-1-9 graphical soundcard mixer for ALSA soundcard ii alsaplayer-alsa 0.99.80-3ubuntu1 PCM player designed for ALSA (ALSA output mo ii alsaplayer-common 0.99.80-3ubuntu1 PCM player designed for ALSA (common files) ii alsaplayer-gtk 0.99.80-3ubuntu1 PCM player designed for ALSA (GTK version) ii bluez-alsa 4.32-0ubuntu3 Bluetooth audio support ii gnome-alsamixer 0.9.7~cvs.20060916.ds.1-2 ALSA sound mixer for GNOME ii gstreamer0.10-alsa 0.10.22-4 GStreamer plugin for ALSA ii libesd-alsa0 0.2.40-0ubuntu3 Enlightened Sound Daemon (ALSA) - Shared lib ii libpt-1.10.10-plugins-alsa 1.10.10-2ubuntu3 Portable Windows Library Audio Plugin for th ii libpt2.4.2-plugins-alsa 2.4.2-2ubuntu1 PTLib audio plugin for the ALSA Interface rc libsdl1.2debian-alsa 1.2.13-2ubuntu1 Simple DirectMedia Layer (with X11 and ALSA This is my kernel: $ uname -a Linux rigel 2.6.28-11-generic #38-Ubuntu SMP Fri Mar 27 10:01:17 UTC 2009 x86_64 GNU/Linux My problem is that I can't have both 7.1 sound and headphones working. If I add this line to /etc/modprobe.d/alsa-base.conf: options snd-hda-intel model=6stack-dig position_fix=0 I have 7.1 sound. But if I plug my headphones, I have sound on both headphones and PC speakers. The only way to mute the speakers and still have sound in the headphones is to uncheck "headphones" in the settings tab of the gnome mixer, and to mute "center" and "LFE". I tried the option probe_mask=1 that is recommended for ALC1200, but it didn't changed anything. I found a solution by googleing. It suggested to add this instead: options snd-hda-intel model=targa-2ch-dig options snd-hda-intel enable_msi=1 In this case I only have headphones when I plug my headphones. However I don't have 7.1 sound anymore, but only stereo. Can someone help me to make my headphones work normally with 7.1 sound? Please feel free to ask me any information you need. Many thanks, - --- Tom -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAknsLSoACgkQngTxdcSw9xxLXACcCX/k6FOFDoU7BWNUUj9UlDTA Sp0AnjjbC42TpssAREpB7nmaVOYORDQe =STMu -----END PGP SIGNATURE----- upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.56 !!################################ !!Script ran on: Mon Apr 20 07:54:18 UTC 2009 !!Linux Distribution !!------------------ Ubuntu 9.04 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 9.04" !!Kernel Information !!------------------ Kernel release: 2.6.28-11-generic Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.18rc3 Library version: 1.0.18 Utilities version: 1.0.18 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No aRts: Installed - Yes (/usr/bin/artsd) Running - No Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf8ef8000 irq 22 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1b.0 0403: 8086:293e (rev 03) Subsystem: 1462:7220 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 snd-hda-intel: model=6stack-dig probe_mask=1 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 model : 6stack-dig,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC1200 Address: 0 Vendor Id: 0x10ec0888 Subsystem Id: 0x14627220 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x1b 0x1b] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x17 0x17] [0x09 0x09] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x18 0x18] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x18 0x18] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x18 0x18] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x18 0x18] Connection: 2 0x05 0x0b Node 0x10 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x11 [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x10 Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01214410: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000036: IN OUT Detect Trigger Pin Default 0x99130120: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000036: IN OUT Detect Trigger Pin Default 0x01012050: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19860: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x6, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x99a30961: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x6, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0181306f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x6, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000373e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x99130130: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x01454140: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Green DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=17 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x25 0x0b --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 8 Apr 20 09:10 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 7 Apr 20 09:13 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 6 Apr 20 09:51 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 5 Apr 20 09:10 /dev/snd/pcmC0D1c crw-rw----+ 1 root audio 116, 4 Apr 20 09:10 /dev/snd/pcmC0D1p crw-rw----+ 1 root audio 116, 3 Apr 20 09:10 /dev/snd/pcmC0D2c crw-rw----+ 1 root audio 116, 2 Apr 20 09:10 /dev/snd/timer !!ALSA configuration files !!------------------------ !!User specific config file (~/.asoundrc) # ALSA library configuration file # Include settings that are under the control of asoundconf(1). # (To disable these settings, comment out this line.) </home/tom/.asoundrc.asoundconf> !!asoundconf-generated config file # ALSA library configuration file managed by asoundconf(1). # # MANUAL CHANGES TO THIS FILE WILL BE OVERWRITTEN! # # Manual changes to the ALSA library configuration should be implemented # by editing the ~/.asoundrc file, not by editing this file. !defaults.pcm.card Intel defaults.ctl.card Intel defaults.pcm.device 0 defaults.pcm.subdevice -1 defaults.pcm.nonblock 1 defaults.pcm.ipc_key 5678293 defaults.pcm.ipc_gid audio defaults.pcm.ipc_perm 0660 defaults.pcm.dmix.max_periods 0 defaults.pcm.dmix.rate 48000 defaults.pcm.dmix.format "unchanged" defaults.pcm.dmix.card defaults.pcm.card defaults.pcm.dmix.device defaults.pcm.device defaults.pcm.dsnoop.card defaults.pcm.card defaults.pcm.dsnoop.device defaults.pcm.device defaults.pcm.front.card defaults.pcm.card defaults.pcm.front.device defaults.pcm.device defaults.pcm.rear.card defaults.pcm.card defaults.pcm.rear.device defaults.pcm.device defaults.pcm.center_lfe.card defaults.pcm.card defaults.pcm.center_lfe.device defaults.pcm.device defaults.pcm.side.card defaults.pcm.card defaults.pcm.side.device defaults.pcm.device defaults.pcm.surround40.card defaults.pcm.card defaults.pcm.surround40.device defaults.pcm.device defaults.pcm.surround41.card defaults.pcm.card defaults.pcm.surround41.device defaults.pcm.device defaults.pcm.surround50.card defaults.pcm.card defaults.pcm.surround50.device defaults.pcm.device defaults.pcm.surround51.card defaults.pcm.card defaults.pcm.surround51.device defaults.pcm.device defaults.pcm.surround71.card defaults.pcm.card defaults.pcm.surround71.device defaults.pcm.device defaults.pcm.iec958.card defaults.pcm.card defaults.pcm.iec958.device defaults.pcm.device defaults.pcm.modem.card defaults.pcm.card defaults.pcm.modem.device defaults.pcm.device defaults.pcm.file_format "raw" defaults.pcm.file_truncate true defaults.rawmidi.card 0 defaults.rawmidi.device 0 defaults.rawmidi.subdevice -1 defaults.hwdep.card 0 defaults.hwdep.device 0 defaults.timer.class 2 defaults.timer.sclass 0 defaults.timer.card 0 defaults.timer.device 0 defaults.timer.subdevice 0 defaults.namehint.showall off defaults.namehint.basic on defaults.namehint.extended off !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC1200 Analog [ALC1200 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC1200 Digital [ALC1200 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC1200 Analog [ALC1200 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC1200 Digital [ALC1200 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 2: ALC1200 Analog [ALC1200 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Card hw:0 'Intel'/'HDA Intel at 0xf8ef8000 irq 22' Mixer name : 'Realtek ALC1200' Components : 'HDA:10ec0888,14627220,00100101' Controls : 40 Simple ctrls : 22 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 24 [77%] [-10.50dB] [on] Simple mixer control 'Headphone',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Front Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] Front Right: 0 [0%] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Side',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [on] Front Right: Playback 0 [0%] [-34.50dB] [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 23 [74%] [0.00dB] [on] Front Right: Playback 23 [74%] [0.00dB] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] Front Right: 0 [0%] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined cswitch cswitch-joined Playback channels: Mono Capture channels: Mono Mono: Playback [off] Capture [off] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'PC Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 9 [29%] [-21.00dB] [on] Front Right: Playback 9 [29%] [-21.00dB] [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 27 [87%] [24.00dB] [on] Front Right: Capture 27 [87%] [24.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-16.50dB] [off] Front Right: Capture 0 [0%] [-16.50dB] [off] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '6ch' '8ch' Item0: '8ch' Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' !!Alsactl output !!------------- --startcollapse-- state.Intel { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Front Playback Volume' value.0 31 value.1 31 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Front Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Surround Playback Volume' value.0 31 value.1 31 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Surround Playback Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Center Playback Volume' value 31 } control.6 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'LFE Playback Volume' value 31 } control.7 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Center Playback Switch' value true } control.8 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'LFE Playback Switch' value true } control.9 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Side Playback Volume' value.0 31 value.1 31 } control.10 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Side Playback Switch' value.0 true value.1 true } control.11 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true } control.12 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'CD Playback Volume' value.0 23 value.1 23 } control.13 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'CD Playback Switch' value.0 true value.1 true } control.14 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Line Playback Volume' value.0 0 value.1 0 } control.15 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Line Playback Switch' value.0 true value.1 true } control.16 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Mic Playback Volume' value.0 0 value.1 0 } control.17 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 3' comment.dbmin 0 comment.dbmax 3000 iface MIXER name 'Mic Boost' value.0 0 value.1 0 } control.18 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Mic Playback Switch' value.0 false value.1 false } control.19 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Front Mic Playback Volume' value.0 0 value.1 0 } control.20 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 3' comment.dbmin 0 comment.dbmax 3000 iface MIXER name 'Front Mic Boost' value.0 0 value.1 0 } control.21 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Front Mic Playback Switch' value.0 false value.1 false } control.22 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'PC Speaker Playback Volume' value.0 9 value.1 9 } control.23 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'PC Speaker Playback Switch' value.0 true value.1 true } control.24 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -1650 comment.dbmax 3000 iface MIXER name 'Capture Volume' value.0 27 value.1 27 } control.25 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true } control.26 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -1650 comment.dbmax 3000 iface MIXER name 'Capture Volume' index 1 value.0 0 value.1 0 } control.27 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false } control.28 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Front Mic' comment.item.2 Line comment.item.3 CD iface MIXER name 'Input Source' value Mic } control.29 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Front Mic' comment.item.2 Line comment.item.3 CD iface MIXER name 'Input Source' index 1 value Mic } control.30 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 '6ch' comment.item.1 '8ch' iface MIXER name 'Channel Mode' value '8ch' } control.31 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.32 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.33 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.34 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value false } control.35 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Default PCM Playback Switch' value true } control.36 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Capture Switch' value false } control.37 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Capture Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.38 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value 24 } control.39 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value true } control.40 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' comment.dbmin -5100 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 } } --endcollapse-- !!All Loaded Modules !!------------------ Module binfmt_misc ppdev bridge stp bnep vboxnetflt vboxdrv input_polldev sbp2 lp parport arc4 ecb snd_hda_intel iwlagn snd_pcm_oss iwlcore snd_mixer_oss led_class snd_pcm uvcvideo mac80211 snd_timer joydev compat_ioctl32 snd videodev sdhci_pci soundcore video psmouse v4l1_compat sdhci snd_page_alloc iTCO_wdt iTCO_vendor_support output pcspkr serio_raw wacom intel_agp cfg80211 nvidia usbhid ohci1394 ieee1394 r8169 mii fbcon tileblit font bitblit softcursor

1 0

[alsa-devel] [PATCH 1/1] ASoC: TWL4030: Add support Voice DAI
by Joonyoung Shim 20 Apr '09

20 Apr '09

Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz (voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. If the system master clock is not 26MHz, the voice PCM interface is not available. The PCM voice interface has two modes - PCM mode1 : This uses the rising edge of the clock signal - PCM mode2 : This uses the falling edge of the clock signal PCM mode1 and mode2 have a look of DSP_A and DSP_B, so we used DSP_A and DSP_B. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 135 +++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/twl4030.h | 16 +++++- sound/soc/omap/omap2evm.c | 2 +- sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/omap3pandora.c | 4 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/sdp3430.c | 2 +- 7 files changed, 150 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index db7fecf..13461d1 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1437,6 +1437,107 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int twl4030_voice_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u8 infreq; + + /* If the system master clock is not 26MHz, the voice PCM interface is + * not avilable. + */ + infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL) + & TWL4030_APLL_INFREQ; + + if (infreq != TWL4030_APLL_INFREQ_26000KHZ) + return -EPERM; + + return 0; +} + +static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u8 old_mode, mode; + + /* bit rate */ + old_mode = twl4030_read_reg_cache(codec, + TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; + mode = old_mode; + + switch (params_rate(params)) { + case 8000: + mode &= ~(TWL4030_SEL_16K); + break; + case 16000: + mode |= TWL4030_SEL_16K; + break; + default: + printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n", + params_rate(params)); + return -EINVAL; + } + + if (mode != old_mode) { + /* change rate and set CODECPDZ */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_codec_enable(codec, 1); + } + + return 0; +} + +static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 old_format, format; + + /* get format */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF); + format = old_format; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFM: + format &= ~(TWL4030_VIF_SLAVE_EN); + break; + case SND_SOC_DAIFMT_CBS_CFS: + format |= TWL4030_VIF_SLAVE_EN; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + format &= ~(TWL4030_VIF_FORMAT); + break; + case SND_SOC_DAIFMT_DSP_B: + format |= TWL4030_VIF_FORMAT; + break; + default: + return -EINVAL; + } + + if (format != old_format) { + /* change format and set CODECPDZ */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_VOICE_IF, format); + twl4030_codec_enable(codec, 1); + } + + return 0; +} + #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) @@ -1448,8 +1549,14 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { .set_fmt = twl4030_set_dai_fmt, }; -struct snd_soc_dai twl4030_dai = { - .name = "twl4030", +static struct snd_soc_dai_ops twl4030_dai_voice_ops = { + .startup = twl4030_voice_startup, + .hw_params = twl4030_voice_hw_params, + .set_fmt = twl4030_voice_set_dai_fmt, +}; + +struct snd_soc_dai twl4030_dai[] = { +{ .name = "twl4030", .playback = { .stream_name = "Playback", .channels_min = 2, @@ -1463,6 +1570,22 @@ struct snd_soc_dai twl4030_dai = { .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, .ops = &twl4030_dai_ops, +}, +{ .name = "twl4030 Voice", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &twl4030_dai_voice_ops, +}, }; EXPORT_SYMBOL_GPL(twl4030_dai); @@ -1503,8 +1626,8 @@ static int twl4030_init(struct snd_soc_device *socdev) codec->read = twl4030_read_reg_cache; codec->write = twl4030_write; codec->set_bias_level = twl4030_set_bias_level; - codec->dai = &twl4030_dai; - codec->num_dai = 1; + codec->dai = twl4030_dai; + codec->num_dai = ARRAY_SIZE(twl4030_dai), codec->reg_cache_size = sizeof(twl4030_reg); codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), GFP_KERNEL); @@ -1598,13 +1721,13 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); static int __init twl4030_modinit(void) { - return snd_soc_register_dai(&twl4030_dai); + return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); } module_init(twl4030_modinit); static void __exit twl4030_exit(void) { - snd_soc_unregister_dai(&twl4030_dai); + snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); } module_exit(twl4030_exit); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index cb63765..305b8ba 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -171,6 +171,17 @@ #define TWL4030_CLK256FS_EN 0x02 #define TWL4030_AIF_EN 0x01 +/* VOICE_IF (0x0F) Fields */ + +#define TWL4030_VIF_SLAVE_EN 0x80 +#define TWL4030_VIF_DIN_EN 0x40 +#define TWL4030_VIF_DOUT_EN 0x20 +#define TWL4030_VIF_SWAP 0x10 +#define TWL4030_VIF_FORMAT 0x08 +#define TWL4030_VIF_TRI_EN 0x04 +#define TWL4030_VIF_SUB_EN 0x02 +#define TWL4030_VIF_EN 0x01 + /* EAR_CTL (0x21) */ #define TWL4030_EAR_GAIN 0x30 @@ -236,7 +247,10 @@ #define TWL4030_SMOOTH_ANAVOL_EN 0x02 #define TWL4030_DIGMIC_LR_SWAP_EN 0x01 -extern struct snd_soc_dai twl4030_dai; +#define TWL4030_DAI_HIFI 0 +#define TWL4030_DAI_VOICE 1 + +extern struct snd_soc_dai twl4030_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_twl4030; #endif /* End of __TWL4030_AUDIO_H__ */ diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c index 0c2322d..027e1a4 100644 --- a/sound/soc/omap/omap2evm.c +++ b/sound/soc/omap/omap2evm.c @@ -86,7 +86,7 @@ static struct snd_soc_dai_link omap2evm_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap2evm_ops, }; diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index fd24a4a..6aa428e 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link omap3beagle_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3beagle_ops, }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index fe282d4..ad219aa 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -228,14 +228,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { .name = "PCM1773", .stream_name = "HiFi Out", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3pandora_out_ops, .init = omap3pandora_out_init, }, { .name = "TWL4030", .stream_name = "Line/Mic In", .cpu_dai = &omap_mcbsp_dai[1], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3pandora_in_ops, .init = omap3pandora_in_init, } diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index a72dc4e..ec4f8fd 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link overo_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &overo_ops, }; diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 10f1c86..1c79741 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -197,7 +197,7 @@ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, }; -- 1.5.6.3

3 7

Re: [alsa-devel] [PATCH v2 2/6] ARM: DaVinci: ASoC: Add the platform devices for ASP
by naresh 20 Apr '09

20 Apr '09

Dave, Thanks for the review. I will re-submit the patch after fixing the comments. Regards, Naresh > >On Thursday 16 April 2009, Medisetty, Naresh wrote: >> arch/arm/mach-davinci/dm355.c | 29 ++++++++++++- >> arch/arm/mach-davinci/dm644x.c | 27 +++++++++++- >> arch/arm/mach-davinci/dm646x.c | 67 >++++++++++++++++++++++++++ >> arch/arm/mach-davinci/include/mach/asp.h | 39 ++++++++++++++++- > >Note the merge dependency: most of those aren't yet in mainline. >I think Kevin sent the dm644x.c changes to the ARM list for their >first review cycle recently. > >My comments below are for the dm355.c changes, but they apply just >as much to the other two SoC-specific setup files. > > >> +static struct resource dm355_evm_snd_resources[] = { >> + { >> + .start = DAVINCI_ASP1_BASE, >> + .end = DAVINCI_ASP1_BASE + SZ_8K - 1, >> + .flags = IORESOURCE_MEM, >> + }, >> +}; >> + >> +static struct evm_snd_platform_data dm355_evm_snd_data = { >> + .clk_name = "asp1", >> + .tx_dma_ch = DAVINCI_DMA_ASP1_TX, >> + .rx_dma_ch = DAVINCI_DMA_ASP1_RX, >> +}; > >If you're going to move to proper driver model support >(good!) ... the DMA channels should be IORESOURCE_DMA >resources, and the clock should be associated directly >with that platform device. > >That might seem to leave that current platform data >struct as useless ... not so! It should tell about how >the audio is configured: whether RX and TX are both >wired, if they're stereo or what, how *audio* clocks >are set up, and so forth. All that is specific to the >specific *board* in use, not to that SoC, so it should >be provided by board setup code. A lot of it is now >encoded in the sound/soc/davinci/davinci-evm.c file. > >Example: dm6446 evm supports six different audio clocks, >but right now one choice is fixed at board setup time. > >ASoC seems to have ways to let drivers make that choice. >That could be handled by board-specific callbacks and data >tables, and would be good to package in platform data. >A different dm6446 board may use a different audio clock >setup, with different choices. > > >> + >> +static struct platform_device dm355_asoc_device = { >> + .name = "davinci-asoc", >> + .id = -1, >> + .dev.platform_data = &dm355_evm_snd_data, >> + .num_resources = ARRAY_SIZE(dm355_evm_snd_resources), >> + .resource = dm355_evm_snd_resources, > >Looks like the alignment goofage there (for "=") is more than >what can be explained by patch mangling from email... > > >> +}; > > >> @@ -555,6 +579,7 @@ static int __init dm355_init_devices(void) >> >> davinci_cfg_reg(DM355_INT_EDMA_CC); >> platform_device_register(&dm355_edma_device); >> + platform_device_register(&dm355_asoc_device); > >This does *NOT* belong in the generic dm355 setup code like that. >Not every DM355 board will even have audio! Or have it wired >exactly like Spectrum's EVM boards. > >It'd be appropriate to add dm355_init_asoc(asp_num, platform_data) >or something similar, so that different boards wouldn't need to >repeat the won't-change parts. That routine might need to set up >some pin, IRQ, and EDMA muxing. Boards with no audio support would >not call that routine. Ones with two audio interfaces might need >to call it once for each interface. > > >> return 0; >> } >> postcore_initcall(dm355_init_devices); > > >Similar comments for the two other SoC chips touched by this patch; >although obviously the dm6446 wouldn't need an asp_num, etc. > >- dave

1 0

[alsa-devel] Via 1708S no S/PDIF out
by Gus Zernial 19 Apr '09

19 Apr '09

I posted this on alsa-user and got no response. I have an ASUS M4A78T-E motherboard system, which has 7.1 channel HD Audio out using Via VT1708S chip, (plus ATI HDMI out). I'm running Kubuntu Jaunty 9.04 with a custom 2.6.28.9 kernel. I am getting sound out the Line Out/Speaker Out jacks, but no sound through the Toslink S/PDIF Out jack. The S/PDIF out channel does not show in aplay -L command output, and there is no S/PDIF out slider in amixer. This is a dual-boot system and everything, including S/PDIF out, works fine with WinXP. In linux, I'm running alsa version 1.0.19. I can post more information if needed, but I think I have the right modules loaded, and I've tried various "model=" options for snd_hda_intel, so far none provide S/PDIF out. My questions are .... can someone tell me if S/PDIF out is supposed to work with this audio chip, and if so what do I need to do configuration-wise to get it to work? Thankyou ...

1 0

[alsa-devel] [PATCH] ASoC: pxa-ssp: Don't use SSCR0_SerClkDiv and SSCR0_SCR
by Philipp Zabel 19 Apr '09

19 Apr '09

Those macros are just screwed as soon as CONFIG_PXA25x is enabled. This patch - changes ssp_set_scr to take an ssp_dev pointer instead of ssp_device - adds a corresponding ssp_get_scr function. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- To me, using ssp_dev seems to be cleaner, as all the places where ssp_set_scr is called, we already have an ssp_dev *ssp = priv->dev.ssp set up, which allows us to call ssp_set_scr(ssp, ...) instead of ssp_set_scr(&priv->dev, ...). Same for ssp_get_scr. --- sound/soc/pxa/pxa-ssp.c | 36 ++++++++++++++++++++++++++++-------- 1 files changed, 28 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 308a657..d979c5c 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -280,12 +280,33 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) * ssp_set_clkdiv - set SSP clock divider * @div: serial clock rate divider */ -static void ssp_set_scr(struct ssp_dev *dev, u32 div) +static void ssp_set_scr(struct ssp_device *ssp, u32 div) { - struct ssp_device *ssp = dev->ssp; - u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR; + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + sscr0 &= ~0x0000ff00; + sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ + } else { + sscr0 &= ~0x000fff00; + sscr0 |= (div - 1) << 8; /* 1..4096 */ + } + ssp_write_reg(ssp, SSCR0, sscr0); +} + +/** + * ssp_get_clkdiv - get SSP clock divider + */ +static u32 ssp_get_scr(struct ssp_device *ssp) +{ + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 div; - ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div))); + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + div = ((sscr0 >> 8) & 0xff) * 2 + 2; + else + div = ((sscr0 >> 8) & 0xfff) + 1; + return div; } /* @@ -326,7 +347,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, break; case PXA_SSP_CLK_AUDIO: priv->sysclk = 0; - ssp_set_scr(&priv->dev, 1); + ssp_set_scr(ssp, 1); sscr0 |= SSCR0_ACS; break; default: @@ -387,7 +408,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, ssp_write_reg(ssp, SSACD, val); break; case PXA_SSP_DIV_SCR: - ssp_set_scr(&priv->dev, div); + ssp_set_scr(ssp, div); break; default: return -ENODEV; @@ -674,8 +695,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_I2S: sspsp = ssp_read_reg(ssp, SSPSP); - if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) && - (width == 16)) { + if ((ssp_get_scr(ssp) == 4) && (width == 16)) { /* This is a special case where the bitclk is 64fs * and we're not dealing with 2*32 bits of audio * samples. -- 1.6.2.3

3 4

[alsa-devel] [PATCH] General fix for Palm27x aSoC driver
by Marek Vasut 19 Apr '09

19 Apr '09

Hi, Firstly, this patch makes the palm27x asoc driver a little more sane. Also, since all affected devices use GPIO95 as AC97_nRESET, this patch sets that properly. Affected are PalmT5, TX and LifeDrive. Please consider applying, it's against Russells tree. If Eric wants to merge it, I can remake it. Thanks.

4 19

[alsa-devel] No jack sense on Intel 82801I / IDT 92HD71B7X in a HP dv4
by Gustavo Vieira 19 Apr '09

19 Apr '09

Hi all, I've got sound problems in my HP Pavilion dv4-1125br. The sound card is a Intel 82801I with the IDT 92HD71B7X codec. Sound works ok, including the internal mic and external mic jack, but I still have a couple of problems. The main problem is that I don't have jack sense in the headphone jack. Headphones do not work and I'm unable even to make them work manually through the mixer. This machine got two headphone jacks in front and none of them works. I've tested it since alsa-driver 1.0.17 and it has never worked. The other problem is that the speaker volume is low. Even if I increase all available mixer controls (PCM and Master), the volume is still tiny. This is a regression. In alsa-driver 1.0.17 (Fedora 10, kernel 2.6.27.21) the volume is ok, the problem appeared sometime after that and alsa-driver 1.0.18a (Fedora 11 beta, kernel 2.6.29.1). One thing I noticed between these versions is the disappearance of the Front mixer control present in 1.0.17. Maybe that is the cause of the problem. I've tried all models available to my codec and none of them solved the issue. Actually, all of them seemed to created other problems. The no model version seems the best right now. I'm attaching the alsa-info output for the Fedora 10 and Fedora 11 beta configurations. I've tried to download and compile the alsa-driver snapshot, but the compilation failed early on with an a patch failure in my kernel 2.6.29.1 system. Abraços, Gustavo

2 7

[alsa-devel] how to find the last plugged in USB headset ?
by reetesh chandran 18 Apr '09

18 Apr '09

Hi all, Is there a way to find the card number associated with the USB headset that was plugged in last. I am listening to udev events and I can find the card number by comparing the VendorId product Id combination. But if there are multiple devices of same VendorId and productId, how do I find the card number associated with the device that was plugged in the last ? Thanks in advance ? Reetesh

1 0

[alsa-devel] SoC pxa2xx-ac97 + wm9705 + touchscreen suspend/resume
by Russell King - ARM Linux 18 Apr '09

18 Apr '09

Trying Linus' latest (as of this morning) out on a PXA255 platform with a WM9705 codec reveals a number of problems: 1. when suspend occurs, we turn the AC97 link off by setting the GCR_ACLINK_OFF bit, and stopping the functional units clock. On resume, we reconfigure the pins and re-enable the clock. Unfortunately, this results in GCR being left with zeros, and GSR remains all zeros, and the codec is inaccessible, with lots of warnings from the read/write functions scrolling by: pxa2xx_ac97_write: write error (ac97_reg=118 GSR=0x0) pxa2xx_ac97_read: read error (ac97_reg=118 GSR=0x0) pxa2xx_ac97_read: read error (ac97_reg=118 GSR=0x0) ... Setting GCR to '2' (to release cold reset) using devmem2 starts things moving again, shutting up this warning. 2. maybe as a result of the above problem, the wm9705 touchscreen driver doesn't reinitialize, causing loss of touchscreen. Unbinding and re-binding the driver restores the touchscreen, but with a very long lag between touching the screen and it being registered. I bring (2) up because I notice that the resume actions in (1) are deferred. Given that codecs have shared functions (such as touchscreens) need to access the codec from their own resume functions, how can this deferral be safe? Liam - this problem has been noticed on a NetbookPro (I'm sure you remember that platform...)

4 20

Re: [alsa-devel] hda Regression: jack doesn't mute speaker
by Takashi Iwai 17 Apr '09

17 Apr '09

At Thu, 16 Apr 2009 09:23:25 +0200, Ortwin Glück wrote: > > Takashi Iwai wrote: > >>>> Also the "Front" > >>>> and "Surround" outputs are now both labeled "Speaker" (see attached images of kmix). > >>> What I/O are they actually corresponding to? Are they speaker outputs > >>> or line-outs, or anything else? > >> They correspond to the built-in speakers. > > > > So they are really front/surround 4.0-speaker outputs? > > Probably. They seem to control two distinct speakers inside the laptop. More I > don't know. In OS-X there is only a single volume control (a sacrifice to > usability I think. who would want to control individual volumes of these on a > laptop anyway). The problem is likely because hp_detect field isn't set on your machine because the headphone shares the same audio-out widget with the speaker out. Could you try the patch below? thanks, Takashi --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ce30b45..917bc5d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3076,6 +3076,11 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, unsigned int wid_caps; for (i = 0; i < num_outs && i < ARRAY_SIZE(chname); i++) { + if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) { + wid_caps = get_wcaps(codec, pins[i]); + if (wid_caps & AC_WCAP_UNSOL_CAP) + spec->hp_detect = 1; + } nid = dac_nids[i]; if (!nid) continue; @@ -3119,11 +3124,6 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, err = create_controls_idx(codec, name, idx, nid, 3); if (err < 0) return err; - if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) { - wid_caps = get_wcaps(codec, pins[i]); - if (wid_caps & AC_WCAP_UNSOL_CAP) - spec->hp_detect = 1; - } } } return 0;

2 1

[alsa-devel] Cirrus Logic CS4207
by Peter-Jan Deweirdt 17 Apr '09

17 Apr '09

Hi, We are in the progress of developming a new platform and we are using the audio chip CS4207 from Cirrus Logic (very new). I've noticed that the audio is not working as it should. I'm using the moblin menlow kernel and I'm able to see 1 playback channel and 1 capture channel (which is not correct according to the chip). I have retreived from our vendor a patch for the kernel (2.6.27.7) but since the moblin kernel is based on the 2.6.29 I cannot use it, even not verify if it's working. Would it be possible to look to the patch and maybe even integrate it into a new build? Or would it be possible to guide me trough so that the audio is working properly? This is what I'm seeing when using the current kernel from moblin (2.6.29.rc7-40.1) >>> chunk from dmesg [ 5.166641] Advanced Linux Sound Architecture Driver Version 1.0.18a. [ 5.171792] HDA Intel 0000:00:1b.0: setting latency timer to 64 [ 5.216898] ALSA device list: [ 5.216916] #0: HDA Intel MID at 0xd0010000 irq 22 [ 5.217024] nf_conntrack version 0.5.0 (3947 buckets, 15788 max) <<<end chunk of dmesg [root@localhost ~]# cat /proc/asound/cards 0 [MID ]: HDA-Intel - HDA Intel MID HDA Intel MID at 0xd0010000 irq 22 [root@localhost ~]# ls -al /dev/snd/ total 0 drwxr-xr-x 2 root root 160 Mar 24 20:00 . drwxr-xr-x 15 root root 3360 Mar 24 20:00 .. crw-rw---- 1 root audio 116, 7 Mar 24 20:00 controlC0 crw-rw---- 1 root audio 116, 6 Mar 24 20:00 hwC0D0 crw-rw---- 1 root audio 116, 5 Mar 24 20:00 pcmC0D0c crw-rw---- 1 root audio 116, 4 Mar 24 20:00 pcmC0D0p crw-rw---- 1 root audio 116, 3 Mar 24 20:00 seq crw-rw---- 1 root audio 116, 2 Mar 24 20:00 timer [root@localhost ~]# [root@localhost ~]# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: MID [HDA Intel MID], device 0: HDA Generic [HDA Generic] Subdevices: 1/1 Subdevice #0: subdevice #0 [root@localhost ~]# [root@localhost ~]# aplay -L front:CARD=MID,DEV=0 HDA Intel MID, HDA Generic Front speakers surround40:CARD=MID,DEV=0 HDA Intel MID, HDA Generic 4.0 Surround output to Front and Rear speakers surround41:CARD=MID,DEV=0 HDA Intel MID, HDA Generic 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=MID,DEV=0 HDA Intel MID, HDA Generic 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=MID,DEV=0 HDA Intel MID, HDA Generic 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=MID,DEV=0 HDA Intel MID, HDA Generic 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server default Default [root@localhost ~]# [root@localhost ~]# alsamixer ALSA lib pulse.c:272:(pulse_connect) PulseAudio: Unable to connect: Connection refused alsamixer: function snd_ctl_open failed for default: Connection refused

2 2

[alsa-devel] [PATCH] us122l: add snd_us122l_free()
by Karsten Wiese 17 Apr '09

17 Apr '09

Use it to clean up snd_us122l_card_used[]. Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> Cc: stable(a)kernel.org --- sound/usb/usx2y/us122l.c | 16 ++++++++++++---- 1 files changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 012ff1f..e0385ce 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2007, 2008 Karsten Wiese <fzu(a)wemgehoertderstaat.de> + * Copyright (C) 2007-2009 Karsten Wiese <fzu(a)wemgehoertderstaat.de> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -26,7 +26,7 @@ #include "us122l.h" MODULE_AUTHOR("Karsten Wiese <fzu(a)wemgehoertderstaat.de>"); -MODULE_DESCRIPTION("TASCAM "NAME_ALLCAPS" Version 0.5"); +MODULE_DESCRIPTION("TASCAM "NAME_ALLCAPS" Version 0.6"); MODULE_LICENSE("GPL"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ @@ -474,6 +474,14 @@ static bool us122l_create_card(struct snd_card *card) return true; } +static void snd_us122l_free(struct snd_card *card) +{ + struct us122l *us122l = US122L(card); + int index = us122l->chip.index; + if (index >= 0 && index < SNDRV_CARDS) + snd_us122l_card_used[index] = 0; +} + static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) { int dev; @@ -490,7 +498,7 @@ static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) if (err < 0) return err; snd_us122l_card_used[US122L(card)->chip.index = dev] = 1; - + card->private_free = snd_us122l_free; US122L(card)->chip.dev = device; US122L(card)->chip.card = card; mutex_init(&US122L(card)->mutex); @@ -584,7 +592,7 @@ static void snd_us122l_disconnect(struct usb_interface *intf) } usb_put_intf(intf); - usb_put_dev(US122L(card)->chip.dev); + usb_put_dev(us122l->chip.dev); while (atomic_read(&us122l->mmap_count)) msleep(500); -- 1.6.0.6

2 3

[alsa-devel] [topic/asoc][PATCH] ASoC: OMAP: Update contact addresses
by Jarkko Nikula 17 Apr '09

17 Apr '09

My email address is going to expire soon so update it. Adding also Peter Ujfalusi <peter.ujfalusi(a)nokia.com> as a second contact to OMAP core drivers since I won't have anymore access to non-public OMAP documentation in the future and Peter is working with these drivers as well. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/omap/n810.c | 4 ++-- sound/soc/omap/omap-mcbsp.c | 5 +++-- sound/soc/omap/omap-mcbsp.h | 3 ++- sound/soc/omap/omap-pcm.c | 5 +++-- sound/soc/omap/omap-pcm.h | 3 ++- 5 files changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index e54e1c2..b60b1df 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula(a)nokia.com> + * Contact: Jarkko Nikula <jhnikula(a)gmail.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -416,6 +416,6 @@ static void __exit n810_soc_exit(void) module_init(n810_soc_init); module_exit(n810_soc_exit); -MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula(a)nokia.com>"); +MODULE_AUTHOR("Jarkko Nikula <jhnikula(a)gmail.com>"); MODULE_DESCRIPTION("ALSA SoC Nokia N810"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 4d6ef0b..495192a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula(a)nokia.com> + * Contact: Jarkko Nikula <jhnikula(a)gmail.com> + * Peter Ujfalusi <peter.ujfalusi(a)nokia.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -546,6 +547,6 @@ static void __exit snd_omap_mcbsp_exit(void) } module_exit(snd_omap_mcbsp_exit); -MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula(a)nokia.com>"); +MODULE_AUTHOR("Jarkko Nikula <jhnikula(a)gmail.com>"); MODULE_DESCRIPTION("OMAP I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index df7ad13..c8147aa 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula(a)nokia.com> + * Contact: Jarkko Nikula <jhnikula(a)gmail.com> + * Peter Ujfalusi <peter.ujfalusi(a)nokia.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 1bdbb04..07cf7f4 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula(a)nokia.com> + * Contact: Jarkko Nikula <jhnikula(a)gmail.com> + * Peter Ujfalusi <peter.ujfalusi(a)nokia.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -367,6 +368,6 @@ static void __exit omap_soc_platform_exit(void) } module_exit(omap_soc_platform_exit); -MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula(a)nokia.com>"); +MODULE_AUTHOR("Jarkko Nikula <jhnikula(a)gmail.com>"); MODULE_DESCRIPTION("OMAP PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index e4369bd..8d9d269 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula(a)nokia.com> + * Contact: Jarkko Nikula <jhnikula(a)gmail.com> + * Peter Ujfalusi <peter.ujfalusi(a)nokia.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License -- 1.6.2.1

2 1

[alsa-devel] [PATCH] us122l: Fix signedness in comparisions
by Karsten Wiese 17 Apr '09

17 Apr '09

Within 2.6.30's mergewindow, struct urb's transfer_buffer_length has become unsigned. This changed an "int > int" comparision to an "unsigned > int" one in snd_usb_122l. Fix this by using a local int variable instead of urb->transfer_buffer_length in comparisions. Shorten playback_prep_freqn() a bit and tweak error-paths in usb_stream_prepare_playback(). Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> Cc: stable(a)kernel.org --- sound/usb/usx2y/usb_stream.c | 67 +++++++++++++++++++----------------------- 1 files changed, 30 insertions(+), 37 deletions(-) diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index 24393da..12ae034 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -33,32 +33,26 @@ static unsigned usb_stream_next_packet_size(struct usb_stream_kernel *sk) static void playback_prep_freqn(struct usb_stream_kernel *sk, struct urb *urb) { struct usb_stream *s = sk->s; - unsigned l = 0; - int pack; - - urb->iso_frame_desc[0].offset = 0; - urb->iso_frame_desc[0].length = usb_stream_next_packet_size(sk); - sk->out_phase = sk->out_phase_peeked; - urb->transfer_buffer_length = urb->iso_frame_desc[0].length; - - for (pack = 1; pack < sk->n_o_ps; pack++) { - l = usb_stream_next_packet_size(sk); - if (s->idle_outsize + urb->transfer_buffer_length + l > - s->period_size) + int pack, lb = 0; + + for (pack = 0; pack < sk->n_o_ps; pack++) { + int l = usb_stream_next_packet_size(sk); + if (s->idle_outsize + lb + l > s->period_size) goto check; sk->out_phase = sk->out_phase_peeked; - urb->iso_frame_desc[pack].offset = urb->transfer_buffer_length; + urb->iso_frame_desc[pack].offset = lb; urb->iso_frame_desc[pack].length = l; - urb->transfer_buffer_length += l; + lb += l; } - snd_printdd(KERN_DEBUG "%i\n", urb->transfer_buffer_length); + snd_printdd(KERN_DEBUG "%i\n", lb); check: urb->number_of_packets = pack; - s->idle_outsize += urb->transfer_buffer_length - s->period_size; + urb->transfer_buffer_length = lb; + s->idle_outsize += lb - s->period_size; snd_printdd(KERN_DEBUG "idle=%i ul=%i ps=%i\n", s->idle_outsize, - urb->transfer_buffer_length, s->period_size); + lb, s->period_size); } static void init_pipe_urbs(struct usb_stream_kernel *sk, unsigned use_packsize, @@ -282,21 +276,20 @@ static int usb_stream_prepare_playback(struct usb_stream_kernel *sk, struct usb_stream *s = sk->s; struct urb *io; struct usb_iso_packet_descriptor *id, *od; - int p, l = 0; + int p = 0, lb = 0, l = 0; io = sk->idle_outurb; od = io->iso_frame_desc; - io->transfer_buffer_length = 0; - for (p = 0; s->sync_packet < 0; ++p, ++s->sync_packet) { + for (; s->sync_packet < 0; ++p, ++s->sync_packet) { struct urb *ii = sk->completed_inurb; id = ii->iso_frame_desc + ii->number_of_packets + s->sync_packet; l = id->actual_length; od[p].length = l; - od[p].offset = io->transfer_buffer_length; - io->transfer_buffer_length += l; + od[p].offset = lb; + lb += l; } for (; @@ -304,38 +297,38 @@ static int usb_stream_prepare_playback(struct usb_stream_kernel *sk, ++p, ++s->sync_packet) { l = inurb->iso_frame_desc[s->sync_packet].actual_length; - if (s->idle_outsize + io->transfer_buffer_length + l > - s->period_size) + if (s->idle_outsize + lb + l > s->period_size) goto check_ok; od[p].length = l; - od[p].offset = io->transfer_buffer_length; - io->transfer_buffer_length += l; + od[p].offset = lb; + lb += l; } check_ok: s->sync_packet -= inurb->number_of_packets; - if (s->sync_packet < -2 || s->sync_packet > 0) { + if (unlikely(s->sync_packet < -2 || s->sync_packet > 0)) { snd_printk(KERN_WARNING "invalid sync_packet = %i;" " p=%i nop=%i %i %x %x %x > %x\n", s->sync_packet, p, inurb->number_of_packets, - s->idle_outsize + io->transfer_buffer_length + l, - s->idle_outsize, io->transfer_buffer_length, l, + s->idle_outsize + lb + l, + s->idle_outsize, lb, l, s->period_size); return -1; } - if (io->transfer_buffer_length % s->cfg.frame_size) { + if (unlikely(lb % s->cfg.frame_size)) { snd_printk(KERN_WARNING"invalid outsize = %i\n", - io->transfer_buffer_length); + lb); return -1; } - s->idle_outsize += io->transfer_buffer_length - s->period_size; + s->idle_outsize += lb - s->period_size; io->number_of_packets = p; - if (s->idle_outsize > 0) { - snd_printk(KERN_WARNING "idle=%i\n", s->idle_outsize); - return -1; - } - return 0; + io->transfer_buffer_length = lb; + if (s->idle_outsize <= 0) + return 0; + + snd_printk(KERN_WARNING "idle=%i\n", s->idle_outsize); + return -1; } static void prepare_inurb(int number_of_packets, struct urb *iu) -- 1.6.0.6

2 2

[alsa-devel] [PATCH] MAINTAINERS - Update SOUND
by Joe Perches 17 Apr '09

17 Apr '09

Signed-off-by: Joe Perches <joe(a)perches.com> diff --git a/MAINTAINERS b/MAINTAINERS index 0cb20d8..c81792f 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5184,13 +5184,17 @@ F: drivers/memstick/host/tifm_ms.c SOUND P: Jaroslav Kysela M: perex(a)perex.cz P: Takashi Iwai M: tiwai(a)suse.de L: alsa-devel(a)alsa-project.org (subscribers-only) +W: http://www.alsa-project.org/main/index.php/Main_Page +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git S: Maintained +F: Documentation/sound/ +F: include/sound/ F: sound/ SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) P: Liam Girdwood M: lrg(a)slimlogic.co.uk P: Mark Brown

2 1

[alsa-devel] [PATCH] ALSA: off by 1 in snd_emu10k1_wait()
by Roel Kluin 17 Apr '09

17 Apr '09

Is this useful? It was in Linus' git tree. ----------------------------->8-----------------8<------------------------------ With `while (count++ < 16384)' count reaches 16385. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index 4bfc31d..c1a5aa1 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -490,7 +490,7 @@ void snd_emu10k1_wait(struct snd_emu10k1 *emu, unsigned int wait) if (newtime != curtime) break; } - if (count >= 16384) + if (count > 16384) break; curtime = newtime; }

2 1

[alsa-devel] [PATCH v2 0/6] ARM: DaVinci: ASoC: Add mcasp support for DM646x
by Naresh Medisetty 17 Apr '09

17 Apr '09

From: Naresh.Medisetty <naresh(a)ti.com> Add audio support for DM646x Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch set adds audio support for DM646x This series is dependant on the following 3 patches submitted to the DaVinci git: [PATCH v2 1/2] ARM: DaVinci: EDMA: Move queue related mappings to dm<soc>.c [PATCH v2 2/2] ARM: DaVinci: EDMA: Enable EDMA support for dm646x [PATCH] ARM: DaVinci: Fix the machine_is_davinci_dm6467_evm() error Patch 1: Adds driver support for the two instances of McASP on TI's DM646x Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 2: Adds McASP clock support for the two instances of mcasp casp0,mcasp1) This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 3: Registers the platform devices for ASP on dm355, dm644x and dm646x This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 4: Add dummy codec support Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 5: Add machine driver support for DM646x Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 6: Introduces platfrom driver model for dm644x and dm355 Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git These patches are based on the work done by Steve Chen <schen(a)mvista.com> and Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> on Monstavista 2.6.18 kernel.

3 2

[alsa-devel] [2.6.30-rc2][PATCH 0/2] Yet another DSP format fix into OMAP McBSP DAI
by Jarkko Nikula 17 Apr '09

17 Apr '09

Hopefully this is the last fourth DSP format fix into McBSP DAI... Tested with N810 modified to use DSP_B and with AIC33 codec. Also with Beagle by using McBSP3 routed to expansion pin header and using scope. Luckily TLV320AIC23 and TLV320AIC33 codec drivers seems to be correct but machine driver for OSK5912 must be corrected to not use inverted FS polarity since AIC23 doesn't have support for it. -- Jarkko

6 25

[alsa-devel] [ALSA - patch_sigmatel.c for HP dv4-1222nr laptop 9/9Patch to patch_sigmatel.c for HP dv4-1222nr laptop
by renidragsemaj＠yahoo.com 17 Apr '09

17 Apr '09

From: James Gardiner <renidragsemaj(a)yahoo.com> This patch to patch_sigmatel.c is for the HP dv4-1222nr laptop. It activates the included microphone(s) and the mute status change light for this computer. I don't know if this will work for any other dv4 laptop. The commit isn't perfect, for it doesn't seem to work too well with pulseaudio - the mic boost doesn't work with that audioserver. Otherwise, it works well with esound, even across reboots and suspend/hibernate/resume. The built-in microphone sits on node 0x18 and feeds through node 0x1c. The gpio for the mute light is GPIO #0. I borrowed the code from the HP HDX. Occasionally if you press the mute button too fast in succession it lowers the actual volume to 0 and you'll have to adjust it again...But if you are doing that you probably are getting a kick out of making me feel bad about the behavior. :) No autodetect of the laptop seems to work for me, but "model=hp-dv4-1222nr" as an option to modprobe will make the changes. Signed-off-by: James Gardiner <renidragsemaj(a)yahoo.com> diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 61996a2..4286155 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -100,6 +100,7 @@ enum { STAC_HP_M4, STAC_HP_DV5, STAC_HP_HDX, + STAC_HP_DV4_1222NR, STAC_92HD71BXX_MODELS }; @@ -1847,6 +1848,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_HP_M4] = "hp-m4", [STAC_HP_DV5] = "hp-dv5", [STAC_HP_HDX] = "hp-hdx", + [STAC_HP_DV4_1222NR] = "hp-dv4-1222nr", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { @@ -4534,6 +4536,28 @@ static int stac92xx_hp_hdx_check_power_status(struct hda_codec *codec, } #endif +static int stac92xx_hp_dv4_1222nr_check_power_status(struct hda_codec *codec, + hda_nid_t nid) +{ + /* Shamelessly stolen from the function stac92xx_hp_hdx_check_power_status */ + /* by James Gardiner */ + struct sigmatel_spec *spec = codec->spec; + + if (nid == 0x10) { + if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & + HDA_AMP_MUTE) + spec->gpio_data &= ~0x01; /* orange */ + else + spec->gpio_data |= 0x01; /* white */ + + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, + spec->gpio_data); + } + + return 0; +} + static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) { struct sigmatel_spec *spec = codec->spec; @@ -5202,6 +5226,19 @@ again: spec->num_smuxes = 0; spec->num_dmuxes = 1; break; + case STAC_HP_DV4_1222NR: /* James Gardiner */ + spec->num_dmics = 1; + spec->num_dmuxes = 2; + stac92xx_auto_set_pinctl(codec, 0x1c, 2); + stac92xx_auto_set_pinctl(codec, 0x18, AC_PINCTL_IN_EN); +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->gpio_mask |= 0x01; + spec->gpio_dir |= 0x01; + spec->gpio_data |= 0x01; + codec->patch_ops.check_power_status = + stac92xx_hp_dv4_1222nr_check_power_status; +#endif + /* fallthrough */ case STAC_HP_DV5: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); -- 1.5.6.3

3 3

[alsa-devel] [PATCH v2 3/6] ARM: DaVinci: ASoC: Add the platform devices for ASP
by Naresh Medisetty 17 Apr '09

17 Apr '09

1)Registers the platform devices for ASP on dm355, dm644x and dm646x so that the machine driver can probe to get ASP related platform data. 2)Move towards definition of the asp clocks using phyisical name(for dm355 and dm644x) Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git arch/arm/mach-davinci/dm355.c | 29 ++++++++++++- arch/arm/mach-davinci/dm644x.c | 27 +++++++++++- arch/arm/mach-davinci/dm646x.c | 67 ++++++++++++++++++++++++++++++ arch/arm/mach-davinci/include/mach/asp.h | 39 ++++++++++++++++- 4 files changed, 157 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/dm355.c b/arch/arm/mach-davinci/dm355.c index c20211c..222c3f3 100755 --- a/arch/arm/mach-davinci/dm355.c +++ b/arch/arm/mach-davinci/dm355.c @@ -23,6 +23,7 @@ #include <mach/psc.h> #include <mach/mux.h> #include <mach/irqs.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -351,8 +352,8 @@ static struct davinci_clk dm355_clks[] = { CLK(NULL, "uart1", &uart1_clk), CLK(NULL, "uart2", &uart2_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), - CLK("soc-audio.0", NULL, &asp0_clk), - CLK("soc-audio.1", NULL, &asp1_clk), + CLK(NULL, "asp0", &asp0_clk), + CLK(NULL, "asp1", &asp1_clk), CLK("davinci_mmc.0", NULL, &mmcsd0_clk), CLK("davinci_mmc.1", NULL, &mmcsd1_clk), CLK(NULL, "spi0", &spi0_clk), @@ -540,6 +541,29 @@ static struct platform_device dm355_edma_device = { .resource = edma_resources, }; +/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ +static struct resource dm355_evm_snd_resources[] = { + { + .start = DAVINCI_ASP1_BASE, + .end = DAVINCI_ASP1_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data dm355_evm_snd_data = { + .clk_name = "asp1", + .tx_dma_ch = DAVINCI_DMA_ASP1_TX, + .rx_dma_ch = DAVINCI_DMA_ASP1_RX, +}; + +static struct platform_device dm355_asoc_device = { + .name = "davinci-asoc", + .id = -1, + .dev.platform_data = &dm355_evm_snd_data, + .num_resources = ARRAY_SIZE(dm355_evm_snd_resources), + .resource = dm355_evm_snd_resources, +}; + /*----------------------------------------------------------------------*/ void __init dm355_init(void) @@ -555,6 +579,7 @@ static int __init dm355_init_devices(void) davinci_cfg_reg(DM355_INT_EDMA_CC); platform_device_register(&dm355_edma_device); + platform_device_register(&dm355_asoc_device); return 0; } postcore_initcall(dm355_init_devices); diff --git a/arch/arm/mach-davinci/dm644x.c b/arch/arm/mach-davinci/dm644x.c index 9878c17..01112ca 100755 --- a/arch/arm/mach-davinci/dm644x.c +++ b/arch/arm/mach-davinci/dm644x.c @@ -20,6 +20,7 @@ #include <mach/irqs.h> #include <mach/psc.h> #include <mach/mux.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -296,7 +297,7 @@ struct davinci_clk dm644x_clks[] = { CLK("davinci_emac.1", NULL, &emac_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), CLK("palm_bk3710", NULL, &ide_clk), - CLK("soc-audio.0", NULL, &asp_clk), + CLK(NULL, "asp0", &asp_clk), CLK("davinci_mmc.0", NULL, &mmcsd_clk), CLK(NULL, "spi", &spi_clk), CLK(NULL, "gpio", &gpio_clk), @@ -463,6 +464,29 @@ static struct platform_device dm644x_edma_device = { .resource = edma_resources, }; +/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ +static struct resource dm644x_evm_snd_resources[] = { + { + .start = DAVINCI_ASP0_BASE, + .end = DAVINCI_ASP0_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data dm644x_evm_snd_data = { + .clk_name = "asp0", + .tx_dma_ch = DAVINCI_DMA_ASP0_TX, + .rx_dma_ch = DAVINCI_DMA_ASP0_RX, +}; + +static struct platform_device dm644x_asoc_device = { + .name = "davinci-asoc", + .id = -1, + .dev.platform_data = &dm644x_evm_snd_data, + .num_resources = ARRAY_SIZE(dm644x_evm_snd_resources), + .resource = dm644x_evm_snd_resources, +}; + /*----------------------------------------------------------------------*/ #if defined(CONFIG_TI_DAVINCI_EMAC) || defined(CONFIG_TI_DAVINCI_EMAC_MODULE) @@ -495,6 +519,7 @@ static int __init dm644x_init_devices(void) return 0; platform_device_register(&dm644x_edma_device); + platform_device_register(&dm644x_asoc_device); return 0; } postcore_initcall(dm644x_init_devices); diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c index 39ffd17..bb8c65a 100755 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -20,6 +20,7 @@ #include <mach/irqs.h> #include <mach/psc.h> #include <mach/mux.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -499,6 +500,70 @@ static struct platform_device dm646x_edma_device = { .resource = edma_resources, }; +static u8 dm6467_iis_serializer_direction[] = { + TX_MODE, RX_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static u8 dm6467_dit_serializer_direction[] = { + TX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static struct evm_snd_platform_data dm6467_evm_snd_data[] = { + { + .clk_name = "mcasp0", + .tx_dma_ch = DAVINCI_DM646X_DMA_MCASP0_AXEVT0, + .rx_dma_ch = DAVINCI_DM646X_DMA_MCASP0_AREVT0, + .tx_dma_offset = 0x400, + .rx_dma_offset = 0x400, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = 4, + .tdm_slots = 2, + .serial_dir = dm6467_iis_serializer_direction, + .eventq_no = EVENTQ_0, + }, + { + .clk_name = "mcasp1", + .tx_dma_ch = DAVINCI_DM646X_DMA_MCASP1_AXEVT1, + .rx_dma_ch = -1, + .tx_dma_offset = 0x400, + .rx_dma_offset = 0, + .op_mode = DAVINCI_MCASP_DIT_MODE, + .num_serializer = 4, + .tdm_slots = 32, + .serial_dir = dm6467_dit_serializer_direction, + .eventq_no = EVENTQ_0, + }, +}; + +static struct resource dm6467_evm_snd_resources[] = { + { + .name = "mcasp0", + .start = DAVINCI_DM646X_MCASP0_REG_BASE, + .end = DAVINCI_DM646X_MCASP0_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, + { + .name = "mcasp1", + .start = DAVINCI_DM646X_MCASP1_REG_BASE, + .end = DAVINCI_DM646X_MCASP1_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct platform_device dm646x_asoc_device = { + .name = "davinci-asoc", + .id = -1, + .dev.platform_data = dm6467_evm_snd_data, + .num_resources = ARRAY_SIZE(dm6467_evm_snd_resources), + .resource = dm6467_evm_snd_resources, +}; + +static struct platform_device dm646x_dit_device = { + .name = "davinci-dit", + .id = -1, + +}; + /*----------------------------------------------------------------------*/ #if defined(CONFIG_TI_DAVINCI_EMAC) || defined(CONFIG_TI_DAVINCI_EMAC_MODULE) @@ -532,6 +597,8 @@ static int __init dm646x_init_devices(void) return 0; platform_device_register(&dm646x_edma_device); + platform_device_register(&dm646x_asoc_device); + platform_device_register(&dm646x_dit_device); return 0; } postcore_initcall(dm646x_init_devices); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index e0abc43..1bb1f0b 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -5,21 +5,56 @@ #define __ASM_ARCH_DAVINCI_ASP_H #include <mach/irqs.h> +#include <mach/edma.h> -/* Bases of register banks */ +/* Bases of dm644x and dm355 register banks */ #define DAVINCI_ASP0_BASE 0x01E02000 #define DAVINCI_ASP1_BASE 0x01E04000 -/* EDMA channels */ + +/* Bases of dm646x register banks */ +#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 +#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 + +/* EDMA channels of dm644x and dm355 */ #define DAVINCI_DMA_ASP0_TX 2 #define DAVINCI_DMA_ASP0_RX 3 #define DAVINCI_DMA_ASP1_TX 8 #define DAVINCI_DMA_ASP1_RX 9 + +/* EDMA channels of dm646x */ +#define DAVINCI_DM646X_DMA_MCASP0_AXEVT0 6 +#define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 +#define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 /* Interrupts */ #define DAVINCI_ASP0_RX_INT IRQ_MBRINT #define DAVINCI_ASP0_TX_INT IRQ_MBXINT #define DAVINCI_ASP1_RX_INT IRQ_MBRINT #define DAVINCI_ASP1_TX_INT IRQ_MBXINT +struct evm_snd_platform_data { + char *clk_name; + int tx_dma_ch; + int rx_dma_ch; + u32 tx_dma_offset; + u32 rx_dma_offset; + enum dma_event_q eventq_no; /* event queue number */ + unsigned int codec_fmt; + + /* McASP specific fields */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + +#define INACTIVE_MODE 0 +#define TX_MODE 1 +#define RX_MODE 2 + +#define DAVINCI_MCASP_IIS_MODE 0 +#define DAVINCI_MCASP_DIT_MODE 1 + + #endif /* __ASM_ARCH_DAVINCI_ASP_H */ -- 1.5.6

1 0

[alsa-devel] [PATCH v2 2/6] ARM: DaVinci: ASoC: Adds McASP clock support for TI DM646X processor
by Naresh Medisetty 17 Apr '09

17 Apr '09

Adds McASP clock support for the two instances of mcasp (mcasp0,mcasp1). This patch is part of the audio support for dm646x series. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git arch/arm/mach-davinci/dm646x.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c index f4a71da..39ffd17 100755 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -220,6 +220,18 @@ static struct clk gpio_clk = { .lpsc = DM646X_LPSC_GPIO, }; +static struct clk mcasp0_clk = { + .name = "mcasp0", + .parent = &pll1_sysclk3, + .lpsc = DM646X_LPSC_McASP0, +}; + +static struct clk mcasp1_clk = { + .name = "mcasp1", + .parent = &pll1_sysclk3, + .lpsc = DM646X_LPSC_McASP1, +}; + static struct clk aemif_clk = { .name = "aemif", .parent = &pll1_sysclk3, @@ -307,6 +319,8 @@ struct davinci_clk dm646x_clks[] = { CLK(NULL, "uart2", &uart2_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), CLK(NULL, "gpio", &gpio_clk), + CLK(NULL, "mcasp0", &mcasp0_clk), + CLK(NULL, "mcasp1", &mcasp1_clk), CLK(NULL, "aemif", &aemif_clk), CLK("davinci_emac.1", NULL, &emac_clk), CLK(NULL, "pwm0", &pwm0_clk), -- 1.5.6

1 0

[alsa-devel] [PATCH v2 1/6] ARM: DaVinci: ASoC: Add mcasp support for DM646x
by Naresh Medisetty 17 Apr '09

17 Apr '09

From: Naresh.Medisetty <naresh(a)ti.com> Adds driver support for the two instances of McASP on TI's DM646x. The multichannel audio serial port (McASP) functions as a general-purpose audio serial port optimized for the needs of multichannel audio application. (http://www.ti.com/litv/pdf/spruer1b). There are two instances of McASP on DM646x. The McASP0 module includes up to 4 serializers that can be individually enabled to either transmit or receive in different modes. The McASP1 module is limited with only 1 pinned-out serializer that can be enabled to only transmit in DIT mode (neither receiving in any mode nor transmitting in either Burst or TDM mode is supported). The codec on the DM646x EVM is AIC32 which is connected to Serializer0 and Serializer1 of McASP0. The McASP0 consists of transmit and receive sections that may operate synchronized, or completely independently with separate master clocks, bit clocks, and frame syncs, and using different transmit modes with different bit-stream formats. Signed-off-by: Steve Chen <schen(a)mvista.com> Signed-off-by: Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/Kconfig | 15 +- sound/soc/davinci/Makefile | 3 + sound/soc/davinci/davinci-i2s-mcasp.c | 890 +++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-i2s-mcasp.h | 56 ++ sound/soc/davinci/davinci-pcm.h | 17 +- 5 files changed, 972 insertions(+), 9 deletions(-) create mode 100644 sound/soc/davinci/davinci-i2s-mcasp.c create mode 100644 sound/soc/davinci/davinci-i2s-mcasp.h diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index bd7392c..0ae84d4 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -9,6 +9,9 @@ config SND_DAVINCI_SOC config SND_DAVINCI_SOC_I2S tristate +config SND_DAVINCI_SOC_I2S_MCASP + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM @@ -16,7 +19,17 @@ config SND_DAVINCI_SOC_EVM select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on TI - DaVinci EVM platform. + DaVinci DM6446 or DM355 EVM platforms. + +config SND_DM6467_SOC_EVM + tristate "SoC Audio support for DaVinci DM6467 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM + select SND_DAVINCI_SOC_I2S_MCASP + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on TI + DaVinci DM6467 EVM platform. + config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index ca8bae1..152d59c 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,13 +1,16 @@ # DAVINCI Platform Support snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o +snd-soc-davinci-i2s-mcasp-objs:= davinci-i2s-mcasp.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o +obj-$(CONFIG_SND_DAVINCI_SOC_I2S_MCASP) += snd-soc-davinci-i2s-mcasp.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-i2s-mcasp.c b/sound/soc/davinci/davinci-i2s-mcasp.c new file mode 100644 index 0000000..5e9ba49 --- /dev/null +++ b/sound/soc/davinci/davinci-i2s-mcasp.c @@ -0,0 +1,890 @@ +/* + * ALSA SoC I2S (McASP) Audio Layer for TI DAVINCI processor + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen, <schen@.mvista.com> + * + * Copyright: (C) 2009 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2009 Texas Instruments, India + * + * 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. + */ + + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s-mcasp.h" + +/* + * McASP register definitions + */ +#define DAVINCI_MCASP_PID_REG 0x00 +#define DAVINCI_MCASP_PWREMUMGT_REG 0x04 + +#define DAVINCI_MCASP_PFUNC_REG 0x10 +#define DAVINCI_MCASP_PDIR_REG 0x14 +#define DAVINCI_MCASP_PDOUT_REG 0x18 +#define DAVINCI_MCASP_PDSET_REG 0x1c + +#define DAVINCI_MCASP_PDCLR_REG 0x20 + +#define DAVINCI_MCASP_TLGC_REG 0x30 +#define DAVINCI_MCASP_TLMR_REG 0x34 + +#define DAVINCI_MCASP_GBLCTL_REG 0x44 +#define DAVINCI_MCASP_AMUTE_REG 0x48 +#define DAVINCI_MCASP_LBCTL_REG 0x4c + +#define DAVINCI_MCASP_TXDITCTL_REG 0x50 + +#define DAVINCI_MCASP_GBLCTLR_REG 0x60 +#define DAVINCI_MCASP_RXMASK_REG 0x64 +#define DAVINCI_MCASP_RXFMT_REG 0x68 +#define DAVINCI_MCASP_RXFMCTL_REG 0x6c + +#define DAVINCI_MCASP_ACLKRCTL_REG 0x70 +#define DAVINCI_MCASP_AHCLKRCTL_REG 0x74 +#define DAVINCI_MCASP_RXTDM_REG 0x78 +#define DAVINCI_MCASP_EVTCTLR_REG 0x7c + +#define DAVINCI_MCASP_RXSTAT_REG 0x80 +#define DAVINCI_MCASP_RXTDMSLOT_REG 0x84 +#define DAVINCI_MCASP_RXCLKCHK_REG 0x88 +#define DAVINCI_MCASP_REVTCTL_REG 0x8c + +#define DAVINCI_MCASP_GBLCTLX_REG 0xa0 +#define DAVINCI_MCASP_TXMASK_REG 0xa4 +#define DAVINCI_MCASP_TXFMT_REG 0xa8 +#define DAVINCI_MCASP_TXFMCTL_REG 0xac + +#define DAVINCI_MCASP_ACLKXCTL_REG 0xb0 +#define DAVINCI_MCASP_AHCLKXCTL_REG 0xb4 +#define DAVINCI_MCASP_TXTDM_REG 0xb8 +#define DAVINCI_MCASP_EVTCTLX_REG 0xbc + +#define DAVINCI_MCASP_TXSTAT_REG 0xc0 +#define DAVINCI_MCASP_TXTDMSLOT_REG 0xc4 +#define DAVINCI_MCASP_TXCLKCHK_REG 0xc8 +#define DAVINCI_MCASP_XEVTCTL_REG 0xcc + +/* Left (even TDM Slot) Channel Status Register File*/ +#define DAVINCI_MCASP_DITCSRA_REG 0x100 +/* Right (odd TDM slot) Channel Status RegisterFile*/ +#define DAVINCI_MCASP_DITCSRB_REG 0x118 +/* Left (even TDM slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRA_REG 0x130 +/* Right (odd TDM Slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRB_REG 0x148 + +/* Serializer n Control Register */ +#define DAVINCI_MCASP_XRSRCTL_BASE_REG 0x180 +#define DAVINCI_MCASP_XRSRCTL_REG(n) (DAVINCI_MCASP_XRSRCTL_BASE_REG + \ + (n << 2)) + +/* Transmit Buffer for Serializer n */ +#define DAVINCI_MCASP_TXBUF_REG 0x200 +/* Receive Buffer for Serializer n */ +#define DAVINCI_MCASP_RXBUF_REG 0x280 + + +/* + * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management + * Register Bits + */ +#define MCASP_FREE BIT(0) +#define MCASP_SOFT BIT(1) + +/* + * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits + */ +#define AXR(n) (1<<n) +#define PFUNC_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits + */ +#define AXR(n) (1<<n) +#define PDIR_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits + */ +#define DITEN BIT(0) /* Transmit DIT mode enable/disable */ +#define VA BIT(2) +#define VB BIT(3) + +/* + * DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits + */ +#define TXROT(val) (val) +#define TXSEL BIT(3) +#define TXSSZ(val) (val<<4) +#define TXPBIT(val) (val<<8) +#define TXPAD(val) (val<<13) +#define TXORD BIT(15) +#define FSXDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits + */ +#define RXROT(val) (val) +#define RXSEL BIT(3) +#define RXSSZ(val) (val<<4) +#define RXPBIT(val) (val<<8) +#define RXPAD(val) (val<<13) +#define RXORD BIT(15) +#define FSRDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_TXFMCTL_REG - Transmit Frame Control Register Bits + */ +#define FSXPOL BIT(0) +#define AFSXE BIT(1) +#define FSXDUR BIT(4) +#define FSXMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits + */ +#define FSRPOL BIT(0) +#define AFSRE BIT(1) +#define FSRDUR BIT(4) +#define FSRMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits + */ +#define ACLKXDIV(val) (val) +#define ACLKXE BIT(5) +#define TX_ASYNC BIT(6) +#define ACLKXPOL BIT(7) + +/* + * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits + */ +#define ACLKRDIV(val) (val) +#define ACLKRE BIT(5) +#define RX_ASYNC BIT(6) +#define ACLKRPOL BIT(7) + +/* + * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control + * Register Bits + */ +#define AHCLKXDIV(val) (val) +#define AHCLKXPOL BIT(14) +#define AHCLKXE BIT(15) + +/* + * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control + * Register Bits + */ +#define AHCLKRDIV(val) (val) +#define AHCLKRPOL BIT(14) +#define AHCLKRE BIT(15) + +/* + * DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits + */ +#define MODE(val) (val) +#define DISMOD(val) (val<<2) +#define TXSTATE BIT(4) +#define RXSTATE BIT(5) + +/* + * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits + */ +#define LBEN BIT(0) +#define LBORD BIT(1) +#define LBGENMODE(val) (val<<2) + +/* + * DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration + */ +#define TXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration + */ +#define RXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_GBLCTL_REG - Global Control Register Bits + */ +#define RXCLKRST BIT(0) /* Receiver Clock Divider Reset */ +#define RXHCLKRST BIT(1) /* Receiver High Frequency Clock Divider */ +#define RXSERCLR BIT(2) /* Receiver Serializer Clear */ +#define RXSMRST BIT(3) /* Receiver State Machine Reset */ +#define RXFSRST BIT(4) /* Frame Sync Generator Reset */ +#define TXCLKRST BIT(8) /* Transmitter Clock Divider Reset */ +#define TXHCLKRST BIT(9) /* Transmitter High Frequency Clock Divider*/ +#define TXSERCLR BIT(10) /* Transmit Serializer Clear */ +#define TXSMRST BIT(11) /* Transmitter State Machine Reset */ +#define TXFSRST BIT(12) /* Frame Sync Generator Reset */ + +/* + * DAVINCI_MCASP_AMUTE_REG - Mute Control Register Bits + */ +#define MUTENA(val) (val) +#define MUTEINPOL BIT(2) +#define MUTEINENA BIT(3) +#define MUTEIN BIT(4) +#define MUTER BIT(5) +#define MUTEX BIT(6) +#define MUTEFSR BIT(7) +#define MUTEFSX BIT(8) +#define MUTEBADCLKR BIT(9) +#define MUTEBADCLKX BIT(10) +#define MUTERXDMAERR BIT(11) +#define MUTETXDMAERR BIT(12) + +/* + * DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits + */ +#define RXDATADMADIS BIT(0) + +/* + * DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits + */ +#define TXDATADMADIS BIT(0) + + + +#define DAVINCI_MCASP_NUM_SERIALIZER 16 + + +static inline void mcasp_set_bits(void __iomem *reg, u32 val) +{ + __raw_writel(__raw_readl(reg) | val, reg); +} + +static inline void mcasp_clr_bits(void __iomem *reg, u32 val) +{ + __raw_writel((__raw_readl(reg) & ~(val)), reg); +} + +static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask) +{ + __raw_writel((__raw_readl(reg) & ~mask) | val, reg); +} + +static inline void mcasp_set_reg(void __iomem *reg, u32 val) +{ + __raw_writel(val, reg); +} + +static inline u32 mcasp_get_reg(void __iomem *reg) +{ + return (unsigned int)__raw_readl(reg); +} + +static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +{ + int i = 0; + mcasp_set_bits(regs, val); + + /* programming GBLCTL needs to read back from GBLCTL and verfiy */ + /* loop count is to avoid the lock-up */ + for (i = 0; i < 1000; i++) { + if ((mcasp_get_reg(regs) & val) == val) + break; + } + if (i == 1000 && ((mcasp_get_reg(regs) & val) != val)) + printk(KERN_ERR "GBLCTL write error\n"); +} + +static int davinci_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + cpu_dai->dma_data = dev->dma_params[substream->stream]; + return 0; +} + +static void mcasp_start_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); +} + +static void mcasp_start_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); +} + +static void davinci_mcasp_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_start_tx(dev); + else + mcasp_start_rx(dev); +} + +static void mcasp_stop_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); +} + +static void mcasp_stop_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); +} + +static void davinci_mcasp_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_stop_tx(dev); + else + mcasp_stop_rx(dev); +} + +static int davinci_i2s_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + void __iomem *base = dev->base; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* codec is clock and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* codec is clock and frame master */ + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26)); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_NB_IF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_IB_IF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_NB_NF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int davinci_config_channel_size(struct snd_pcm_substream *substream, + int channel_size) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + u32 fmt = 0; + + switch (channel_size) { + case DAVINCI_AUDIO_WORD_8: + fmt = 0x03; + break; + + case DAVINCI_AUDIO_WORD_12: + fmt = 0x05; + break; + + case DAVINCI_AUDIO_WORD_16: + fmt = 0x07; + break; + + case DAVINCI_AUDIO_WORD_20: + fmt = 0x09; + break; + + case DAVINCI_AUDIO_WORD_24: + fmt = 0x0B; + break; + + case DAVINCI_AUDIO_WORD_28: + fmt = 0x0D; + break; + + case DAVINCI_AUDIO_WORD_32: + fmt = 0x0F; + break; + + default: + return -1; + } + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + return 0; +} + +static void davinci_hw_common_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + int i; + + /* Default configuration */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); + + /* All PINS as McASP */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, + TXDATADMADIS); + } else { + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG, + RXDATADMADIS); + } + + for (i = 0; i < dev->num_serializer; i++) { + mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + dev->serial_dir[i]); + if (dev->serial_dir[i] == TX_MODE) + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + else if (dev->serial_dir[i] == RX_MODE) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + } +} + +static void davinci_hw_param(struct snd_pcm_substream *substream) +{ + int i, active_slots; + u32 mask = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots; + for (i = 0; i < active_slots; i++) + mask |= (1 << i); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* bit stream is MSB first with no delay */ + /* DSP_B mode */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(dev->tdm_slots), FSXMOD(0x1FF)); + else + printk(KERN_ERR "playback tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + + } else { + /* bit stream is MSB first with no delay */ + /* DSP_B mode */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXORD); + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, + AHCLKRE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(dev->tdm_slots), FSRMOD(0x1FF)); + else + printk(KERN_ERR "capture tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + } +} + +/* S/PDIF */ +static void davinci_hw_dit_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + /* Set the PDIR for Serialiser as output */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); + + /* TXMASK for 24 bits */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); + + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 + and LSB first */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(6) | TXSSZ(15)); + + /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + AFSXE | FSXMOD(0x180)); + + /* Set the TX tdm : for all the slots */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); + + /* Set the TX clock controls : div = 1 and internal */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE | TX_ASYNC); + + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + + /* Only 44100 and 48000 are valid, both have the same setting */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXDIV(3)); + + /* Enable the DIT */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); +} + +static int davinci_i2s_mcasp_hw_params( + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = + dev->dma_params[substream->stream]; + int word_length; + + davinci_hw_common_param(substream); + + if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) + davinci_hw_dit_param(substream); + else + davinci_hw_param(substream); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + word_length = DAVINCI_AUDIO_WORD_8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + word_length = DAVINCI_AUDIO_WORD_16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + dma_params->data_type = 4; + word_length = DAVINCI_AUDIO_WORD_32; + break; + default: + printk(KERN_WARNING "davinci-i2s: unsupported PCM format"); + return -EINVAL; + } + davinci_config_channel_size(substream, word_length); + + return 0; +} + +static int davinci_i2s_mcasp_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_mcasp_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcasp_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_mcasp_trigger, + .hw_params = davinci_i2s_mcasp_hw_params, + .set_fmt = davinci_i2s_mcasp_set_dai_fmt, + +}; + +struct snd_soc_dai davinci_iis_mcasp_dai[] = { + { + .name = "davinci-i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + + }, + .capture = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &davinci_i2s_dai_ops, + + }, + { + .name = "davinci-dit", + .id = 1, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &davinci_i2s_dai_ops, + }, + +}; +EXPORT_SYMBOL_GPL(davinci_iis_mcasp_dai); + +static int davinci_i2s_mcasp_probe(struct platform_device *pdev) +{ + int tmp, link_cnt, num_links = 2; + int count = 0; + struct davinci_audio_dev *dev; + struct resource *mem, *ioarea; + struct evm_snd_platform_data *parray = pdev->dev.platform_data; + struct evm_snd_platform_data *pdata; + struct davinci_pcm_dma_params *dma_data; + int ret = 0; + + dev = kzalloc(sizeof(struct davinci_audio_dev) * num_links, + GFP_KERNEL); + if (!dev) + return -ENOMEM; + + dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * + (num_links << 1), GFP_KERNEL); + + if (!dma_data) { + ret = -ENOMEM; + goto err_release_dev; + } + + for (link_cnt = 0; link_cnt < num_links; link_cnt++) { + mem = platform_get_resource(pdev, IORESOURCE_MEM, link_cnt); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + ret = -ENODEV; + goto err_release_data; + } + ioarea = request_mem_region(mem->start, + (mem->end - mem->start) + 1, pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "Audio region already claimed\n"); + ret = -EBUSY; + goto err_release_data; + } + + pdata = &parray[link_cnt]; + dev[link_cnt].clk = clk_get(&pdev->dev, pdata->clk_name); + + if (IS_ERR(dev[link_cnt].clk)) { + ret = -ENODEV; + goto err_release_region; + } + clk_enable(dev[link_cnt].clk); + + dev[link_cnt].base = (void __iomem *)IO_ADDRESS(mem->start); + dev[link_cnt].op_mode = pdata->op_mode; + dev[link_cnt].tdm_slots = pdata->tdm_slots; + dev[link_cnt].num_serializer = pdata->num_serializer; + dev[link_cnt].serial_dir = pdata->serial_dir; + dev[link_cnt].codec_fmt = pdata->codec_fmt; + + + dma_data[count].name = "I2S PCM Stereo out"; + dma_data[count].channel = pdata->tx_dma_ch; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset + + io_v2p(dev[link_cnt].base)); + dev[link_cnt].dma_params[SNDRV_PCM_STREAM_PLAYBACK] = + &dma_data[count]; + + + count++; + dma_data[count].name = "I2S PCM Stereo in"; + dma_data[count].channel = pdata->rx_dma_ch; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset + + io_v2p(dev[link_cnt].base)); + dev[link_cnt].dma_params[SNDRV_PCM_STREAM_CAPTURE] = + &dma_data[count]; + count++; + davinci_iis_mcasp_dai[link_cnt].private_data = &dev[link_cnt]; + } + ret = snd_soc_register_dais(davinci_iis_mcasp_dai, + ARRAY_SIZE(davinci_iis_mcasp_dai)); + if (ret != 0) + goto err_release_region; + return 0; + +err_release_region: + for (tmp = link_cnt; tmp >= 0; tmp--) { + mem = platform_get_resource(pdev, IORESOURCE_MEM, tmp); + release_mem_region(mem->start, + (mem->end - mem->start) + 1); + } +err_release_data: + kfree(dma_data); +err_release_dev: + kfree(dev); + + return ret; +} + +static int davinci_i2s_mcasp_remove(struct platform_device *pdev) +{ + int i, num_links = 2; + struct davinci_audio_dev *dev; + struct resource *mem; + struct davinci_pcm_dma_params *dma_data; + + snd_soc_unregister_dais(davinci_iis_mcasp_dai, + ARRAY_SIZE(davinci_iis_mcasp_dai)); + + for (i = 0; i < num_links; i++) { + dev = davinci_iis_mcasp_dai[i].private_data; + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, i); + release_mem_region(mem->start, (mem->end - mem->start) + 1); + } + + dev = davinci_iis_mcasp_dai[0].private_data; + dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + kfree(dma_data); + kfree(dev); + return 0; +} + +static struct platform_driver davinci_asoc_driver = { + .probe = davinci_i2s_mcasp_probe, + .remove = davinci_i2s_mcasp_remove, + .driver = { + .name = "davinci-asoc", + .owner = THIS_MODULE, + }, +}; + +static int __init davinci_i2s_mcasp_init(void) +{ + + return platform_driver_register(&davinci_asoc_driver); + +} +module_init(davinci_i2s_mcasp_init); + +static void __exit davinci_i2s_mcasp_exit(void) +{ + + platform_driver_unregister(&davinci_asoc_driver); + +} +module_exit(davinci_i2s_mcasp_exit); + + + +MODULE_AUTHOR("Steve Chen"); +MODULE_DESCRIPTION("TI DAVINCI I2S (McASP) SoC Interface"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/davinci/davinci-i2s-mcasp.h b/sound/soc/davinci/davinci-i2s-mcasp.h new file mode 100644 index 0000000..6469e7d --- /dev/null +++ b/sound/soc/davinci/davinci-i2s-mcasp.h @@ -0,0 +1,56 @@ +/* + * ALSA SoC I2S (McASP) Audio Layer for TI DAVINCI processor + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen, <schen@.mvista.com> + * + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2008 Texas Instruments, India + * + * 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. + */ + + +#ifndef DAVINCI_I2S_MCASP_H +#define DAVINCI_I2S_MCASP_H + +#include <linux/io.h> +#include <mach/asp.h> +#include "davinci-pcm.h" + +extern struct snd_soc_dai davinci_iis_mcasp_dai[]; +extern struct snd_soc_dai davinci_dit_mcasp_dai; + + +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +enum { + DAVINCI_AUDIO_WORD_8 = 0, + DAVINCI_AUDIO_WORD_12, + DAVINCI_AUDIO_WORD_16, + DAVINCI_AUDIO_WORD_20, + DAVINCI_AUDIO_WORD_24, + DAVINCI_AUDIO_WORD_32, + DAVINCI_AUDIO_WORD_28, /* This is only valid for McASP */ +}; + +struct davinci_audio_dev { + void __iomem *base; + int sample_rate; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; + unsigned int codec_fmt; + + /* McASP specific data */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + + + +#endif /* DAVINCI_I2S_MCASP_H */ diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 62cb4eb..3cbe281 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -12,17 +12,18 @@ #ifndef _DAVINCI_PCM_H #define _DAVINCI_PCM_H +#include <mach/edma.h> +#include <mach/asp.h> + + struct davinci_pcm_dma_params { - char *name; /* stream identifier */ - int channel; /* sync dma channel ID */ - dma_addr_t dma_addr; /* device physical address for DMA */ - unsigned int data_type; /* xfer data type */ + char *name; /* stream identifier */ + int channel; /* sync dma channel ID */ + dma_addr_t dma_addr; /* device physical address for DMA */ + unsigned int data_type; /* xfer data type */ + enum dma_event_q eventq_no; /* event queue number */ }; -struct evm_snd_platform_data { - int tx_dma_ch; - int rx_dma_ch; -}; extern struct snd_soc_platform davinci_soc_platform; -- 1.5.6

1 0

[alsa-devel] /dev/snd/timer behavior
by Ron Cococcia 17 Apr '09

17 Apr '09

Hello, We have been attempting to debug an issue that we have been having for a few weeks related to audio playback. Specifically, over time, the playback of audio gets 'stuck' for a while (60 seconds or so), then quickly advances forward again (for about 60 seconds), then gets stuck again. It continues to repeat this until I either stop playback or pause/play. We have dubbed it 'skip-lock' as it skips ahead, then locks for a bit. It takes a while to reproduce this behavior, typically around 6 hours. We have about 30 of these systems setup, each with the same hardware configuration. They have a VIA C7 CPU (1.5GHz), 1GB RAM, and they are using the VIA 8233A AC97 controller with an ALC655 codec. The system is SATA based (using sata_via), and an RTL8110 gigabit ethernet controller (using rtl8169). For the OS, we are using Debian Etch with the 2.6.29 kernel, and we have tested a number of these units with 2.6.24-etchnhalf as well. ALSA lib 1.0.13 (from the install) is still in use, but I am trying out 1.0.19 on a few of the units to see if it helps. The player we use is called BlueTune, which we have used in older systems for years without trouble. I plan on trying to set up another player app to do the same testing, but haven't had the time to do that yet. On most of the systems, the player is using the default alsa output. I have configured a few systems to use plughw:0,0,0 for playback (mainly so that /dev/snd/timer is not used), and they have not had any playback issues so far (about 24 hours so far). After a bunch of other tests, we started to run strace against the player. We saw that it was repeatedly getting EAGAIN when reading from /dev/snd/timer during the 'stuck' phase. I have started to poke around a bit more in the timer code and see why EAGAIN is repeatedly being generated (in snd_timer_user_read, tu->qused is stuck at 0 for a while). I figured that if anybody had some things to try or places to look, it might save a bit of time in finding the cause of this issue. Thanks! Ron PS: I apologize if this is the wrong place to post this. I had posted a not-terribly-detailed message to alsa-user a few weeks ago, but have gained more data that I felt fit better here.

1 0

[alsa-devel] /proc/asound/* shows signs of life, but alsactl see's nothing
by Justin Mattock 16 Apr '09

16 Apr '09

This is confusing to me (with or without the imac9,1 patch for the imac) there is still entries in /proc/asound lrwxrwxrwx 1 root root 5 Apr 15 22:49 NVidia -> card0 dr-xr-xr-x 7 root root 0 Apr 15 22:49 card0 -r--r--r-- 1 root root 0 Apr 15 22:49 cards -r--r--r-- 1 root root 0 Apr 15 22:49 devices -r--r--r-- 1 root root 0 Apr 15 22:49 hwdep -r--r--r-- 1 root root 0 Apr 15 22:49 pcm -r--r--r-- 1 root root 0 Apr 15 22:49 timers -r--r--r-- 1 root root 0 Apr 15 22:49 version but when I do: sudo /usr/sbin/alsactl store /usr/sbin/alsactl: save_state:1502: No soundcards found... I this a software issue, or is this because the kernel is missing something important in patch_realtek.c ? I've moved different things around in patch_realtek.c but still no speaker sound or headphone, not even the sound during boot when the module is being loaded. Also what is the best way to find PCI SSID? under codec#0:(I have) Codec: Realtek ALC889A Address: 0 Vendor Id: 0x10ec0885 Subsystem Id: 0x106b4900 Revision Id: 0x100103 No Modem Function Group found but with lspci -nnv 00:08.0 0403: 10de:0ac0 (rev b1) Subsystem: 10de:cb79 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 (500ns min, 1250ns max) Interrupt: pin A routed to IRQ 17 Region 0: Memory at d3480000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel the subsystem is 10de:cb79 but is Subsystem Id: 0x106b4900 regards, -- Justin P. Mattock

2 4

[alsa-devel] [PATCH] hdspm: native float
by Florian Faber 16 Apr '09

16 Apr '09

Adds native float support to the driver. The interface has hardware support for CoreAudio[tm], why shouldn't jack users benefit from it. diff a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c 13a14,16 > * Modified 2009-04-14 for native float support by Florian Faber > * <faber(a)faberman.de> > * 39a43 > #include <sound/pcm_params.h> 198a203 > #define HDSPe_FLOAT_FORMAT 0x2000000 3726a3732,3747 > > > /* Switch to native float format if requested */ > if (SNDRV_PCM_FORMAT_FLOAT_LE == params_format(params)) { > if (!(hdspm->control_register & HDSPe_FLOAT_FORMAT)) { > snd_printk(KERN_INFO "hdspm: Switching to native 32bit LE float format.\n"); > } > hdspm->control_register |= HDSPe_FLOAT_FORMAT; > } else if (SNDRV_PCM_FORMAT_S32_LE == params_format(params)) { > if (hdspm->control_register & HDSPe_FLOAT_FORMAT) { > snd_printk(KERN_INFO "hdspm: Switching to native 32bit LE integer format.\n"); > } > hdspm->control_register &= ~HDSPe_FLOAT_FORMAT; > } > hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register); > 3870c3891 < .formats = SNDRV_PCM_FMTBIT_S32_LE, --- > .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_FLOAT_LE, 3895c3916 < .formats = SNDRV_PCM_FMTBIT_S32_LE, --- > .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_FLOAT_LE, Flo -- Machines can do the work, so people have time to think. public key 6C002249 x-hkp://wwwkeys.eu.pgp.net

6 8

[alsa-devel] [PATCH v2 2/6] ARM: DaVinci: ASoC: Add the platform devices for ASP
by Naresh Medisetty 16 Apr '09

16 Apr '09

1)Registers the platform devices for ASP on dm355, dm644x and dm646x so that the machine driver can probe to get ASP related platform data. 2)Move towards definition of the asp clocks using phyisical name(for dm355 and dm644x) Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git arch/arm/mach-davinci/dm355.c | 29 ++++++++++++- arch/arm/mach-davinci/dm644x.c | 27 +++++++++++- arch/arm/mach-davinci/dm646x.c | 67 ++++++++++++++++++++++++++++++ arch/arm/mach-davinci/include/mach/asp.h | 39 ++++++++++++++++- 4 files changed, 157 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/dm355.c b/arch/arm/mach-davinci/dm355.c index c20211c..222c3f3 100755 --- a/arch/arm/mach-davinci/dm355.c +++ b/arch/arm/mach-davinci/dm355.c @@ -23,6 +23,7 @@ #include <mach/psc.h> #include <mach/mux.h> #include <mach/irqs.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -351,8 +352,8 @@ static struct davinci_clk dm355_clks[] = { CLK(NULL, "uart1", &uart1_clk), CLK(NULL, "uart2", &uart2_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), - CLK("soc-audio.0", NULL, &asp0_clk), - CLK("soc-audio.1", NULL, &asp1_clk), + CLK(NULL, "asp0", &asp0_clk), + CLK(NULL, "asp1", &asp1_clk), CLK("davinci_mmc.0", NULL, &mmcsd0_clk), CLK("davinci_mmc.1", NULL, &mmcsd1_clk), CLK(NULL, "spi0", &spi0_clk), @@ -540,6 +541,29 @@ static struct platform_device dm355_edma_device = { .resource = edma_resources, }; +/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ +static struct resource dm355_evm_snd_resources[] = { + { + .start = DAVINCI_ASP1_BASE, + .end = DAVINCI_ASP1_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data dm355_evm_snd_data = { + .clk_name = "asp1", + .tx_dma_ch = DAVINCI_DMA_ASP1_TX, + .rx_dma_ch = DAVINCI_DMA_ASP1_RX, +}; + +static struct platform_device dm355_asoc_device = { + .name = "davinci-asoc", + .id = -1, + .dev.platform_data = &dm355_evm_snd_data, + .num_resources = ARRAY_SIZE(dm355_evm_snd_resources), + .resource = dm355_evm_snd_resources, +}; + /*----------------------------------------------------------------------*/ void __init dm355_init(void) @@ -555,6 +579,7 @@ static int __init dm355_init_devices(void) davinci_cfg_reg(DM355_INT_EDMA_CC); platform_device_register(&dm355_edma_device); + platform_device_register(&dm355_asoc_device); return 0; } postcore_initcall(dm355_init_devices); diff --git a/arch/arm/mach-davinci/dm644x.c b/arch/arm/mach-davinci/dm644x.c index 9878c17..01112ca 100755 --- a/arch/arm/mach-davinci/dm644x.c +++ b/arch/arm/mach-davinci/dm644x.c @@ -20,6 +20,7 @@ #include <mach/irqs.h> #include <mach/psc.h> #include <mach/mux.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -296,7 +297,7 @@ struct davinci_clk dm644x_clks[] = { CLK("davinci_emac.1", NULL, &emac_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), CLK("palm_bk3710", NULL, &ide_clk), - CLK("soc-audio.0", NULL, &asp_clk), + CLK(NULL, "asp0", &asp_clk), CLK("davinci_mmc.0", NULL, &mmcsd_clk), CLK(NULL, "spi", &spi_clk), CLK(NULL, "gpio", &gpio_clk), @@ -463,6 +464,29 @@ static struct platform_device dm644x_edma_device = { .resource = edma_resources, }; +/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ +static struct resource dm644x_evm_snd_resources[] = { + { + .start = DAVINCI_ASP0_BASE, + .end = DAVINCI_ASP0_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data dm644x_evm_snd_data = { + .clk_name = "asp0", + .tx_dma_ch = DAVINCI_DMA_ASP0_TX, + .rx_dma_ch = DAVINCI_DMA_ASP0_RX, +}; + +static struct platform_device dm644x_asoc_device = { + .name = "davinci-asoc", + .id = -1, + .dev.platform_data = &dm644x_evm_snd_data, + .num_resources = ARRAY_SIZE(dm644x_evm_snd_resources), + .resource = dm644x_evm_snd_resources, +}; + /*----------------------------------------------------------------------*/ #if defined(CONFIG_TI_DAVINCI_EMAC) || defined(CONFIG_TI_DAVINCI_EMAC_MODULE) @@ -495,6 +519,7 @@ static int __init dm644x_init_devices(void) return 0; platform_device_register(&dm644x_edma_device); + platform_device_register(&dm644x_asoc_device); return 0; } postcore_initcall(dm644x_init_devices); diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c index 39ffd17..bb8c65a 100755 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -20,6 +20,7 @@ #include <mach/irqs.h> #include <mach/psc.h> #include <mach/mux.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -499,6 +500,70 @@ static struct platform_device dm646x_edma_device = { .resource = edma_resources, }; +static u8 dm6467_iis_serializer_direction[] = { + TX_MODE, RX_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static u8 dm6467_dit_serializer_direction[] = { + TX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static struct evm_snd_platform_data dm6467_evm_snd_data[] = { + { + .clk_name = "mcasp0", + .tx_dma_ch = DAVINCI_DM646X_DMA_MCASP0_AXEVT0, + .rx_dma_ch = DAVINCI_DM646X_DMA_MCASP0_AREVT0, + .tx_dma_offset = 0x400, + .rx_dma_offset = 0x400, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = 4, + .tdm_slots = 2, + .serial_dir = dm6467_iis_serializer_direction, + .eventq_no = EVENTQ_0, + }, + { + .clk_name = "mcasp1", + .tx_dma_ch = DAVINCI_DM646X_DMA_MCASP1_AXEVT1, + .rx_dma_ch = -1, + .tx_dma_offset = 0x400, + .rx_dma_offset = 0, + .op_mode = DAVINCI_MCASP_DIT_MODE, + .num_serializer = 4, + .tdm_slots = 32, + .serial_dir = dm6467_dit_serializer_direction, + .eventq_no = EVENTQ_0, + }, +}; + +static struct resource dm6467_evm_snd_resources[] = { + { + .name = "mcasp0", + .start = DAVINCI_DM646X_MCASP0_REG_BASE, + .end = DAVINCI_DM646X_MCASP0_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, + { + .name = "mcasp1", + .start = DAVINCI_DM646X_MCASP1_REG_BASE, + .end = DAVINCI_DM646X_MCASP1_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct platform_device dm646x_asoc_device = { + .name = "davinci-asoc", + .id = -1, + .dev.platform_data = dm6467_evm_snd_data, + .num_resources = ARRAY_SIZE(dm6467_evm_snd_resources), + .resource = dm6467_evm_snd_resources, +}; + +static struct platform_device dm646x_dit_device = { + .name = "davinci-dit", + .id = -1, + +}; + /*----------------------------------------------------------------------*/ #if defined(CONFIG_TI_DAVINCI_EMAC) || defined(CONFIG_TI_DAVINCI_EMAC_MODULE) @@ -532,6 +597,8 @@ static int __init dm646x_init_devices(void) return 0; platform_device_register(&dm646x_edma_device); + platform_device_register(&dm646x_asoc_device); + platform_device_register(&dm646x_dit_device); return 0; } postcore_initcall(dm646x_init_devices); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index e0abc43..1bb1f0b 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -5,21 +5,56 @@ #define __ASM_ARCH_DAVINCI_ASP_H #include <mach/irqs.h> +#include <mach/edma.h> -/* Bases of register banks */ +/* Bases of dm644x and dm355 register banks */ #define DAVINCI_ASP0_BASE 0x01E02000 #define DAVINCI_ASP1_BASE 0x01E04000 -/* EDMA channels */ + +/* Bases of dm646x register banks */ +#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 +#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 + +/* EDMA channels of dm644x and dm355 */ #define DAVINCI_DMA_ASP0_TX 2 #define DAVINCI_DMA_ASP0_RX 3 #define DAVINCI_DMA_ASP1_TX 8 #define DAVINCI_DMA_ASP1_RX 9 + +/* EDMA channels of dm646x */ +#define DAVINCI_DM646X_DMA_MCASP0_AXEVT0 6 +#define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 +#define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 /* Interrupts */ #define DAVINCI_ASP0_RX_INT IRQ_MBRINT #define DAVINCI_ASP0_TX_INT IRQ_MBXINT #define DAVINCI_ASP1_RX_INT IRQ_MBRINT #define DAVINCI_ASP1_TX_INT IRQ_MBXINT +struct evm_snd_platform_data { + char *clk_name; + int tx_dma_ch; + int rx_dma_ch; + u32 tx_dma_offset; + u32 rx_dma_offset; + enum dma_event_q eventq_no; /* event queue number */ + unsigned int codec_fmt; + + /* McASP specific fields */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + +#define INACTIVE_MODE 0 +#define TX_MODE 1 +#define RX_MODE 2 + +#define DAVINCI_MCASP_IIS_MODE 0 +#define DAVINCI_MCASP_DIT_MODE 1 + + #endif /* __ASM_ARCH_DAVINCI_ASP_H */ -- 1.5.6

3 2

Re: [alsa-devel] 15 April unstable does not handle wave-in
by Susan Cragin 16 Apr '09

16 Apr '09

>Works perfectly. Thank you. >Susan > > Whoops. Spoke too soon. Does not save alsa settings. They have to be re-set after each boot. Susan

1 0

Re: [alsa-devel] 15 April unstable does not handle wave-in
by Susan Cragin 16 Apr '09

16 Apr '09

>> Meanwhile, if ca0110 worked fine with unstable tree before, it means that the hda-ca0110 patch can go in to the sound git tree. This would be a better option for both of us. >> >> Now I merged hda-ca0110 patch to sound git tree. >> So alsa-driver snapshot (no unstable version) should work for your >> CA0110 board. Give it a try. >Takashi Works perfectly. Thank you. Susan

1 0

[alsa-devel] No sound with Creative Labs CA0106 Soundblaster
by Bernhard Posselt 16 Apr '09

16 Apr '09

Hi, some months ago i bought a new soundcard for my PC which costed 20€. This card is not working. Im using Kubuntu Jaunty Beta, its a fresh install from the beta iso. I also upgraded the alsadrivers with this package http://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sn… Still no sound Heres my bug report: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4490 I hope you can help me regards Ray -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger01

2 1

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 16 Apr '09

16 Apr '09

The following changes since commit 6967963d6d5cac40a091d075326f0e3ccb95c58a: Mark Brown (1): Merge branch 'for-2.6.30' into for-2.6.31 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Ben Dooks (4): ASoC: Fix jive_wm8750.c build problems ASoC: s3c-i2s-v2.c fix for s3c_i2sv2_iis_calc_rate ASoC: Fix s3c-i2s-v2.c snd_soc_dai changes ASoC: Fix include build error in s3c2412-i2s.c Daniel Mack (1): ASoC: pxa-ssp: allow setting of dai format 0 Jarkko Nikula (2): ASoC: OMAP: Fix DSP_B format in OMAP McBSP DAI driver ASoC: OMAP: Fix FS polarity in OSK5912 machine driver Mark Brown (4): ASoC: Check we have DAI ops when calling via accessor functions ASoC: Volume controls are never of boolean type ASoC: Request shared rates for WM8903 Merge branch 'for-2.6.30' into for-2.6.31 Peter Ujfalusi (2): ASoC: OMAP: Use single-phase for DSP mode ASoC: OMAP: Add DSP_A mode support for mcbsp sound/soc/codecs/wm8903.c | 1 + sound/soc/omap/omap-mcbsp.c | 45 ++++++++++++++++++++++++++------------ sound/soc/omap/osk5912.c | 4 +- sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/s3c24xx/jive_wm8750.c | 12 +++++----- sound/soc/s3c24xx/s3c-i2s-v2.c | 18 ++++++++------- sound/soc/s3c24xx/s3c2412-i2s.c | 2 +- sound/soc/soc-core.c | 20 ++++++++-------- 8 files changed, 62 insertions(+), 41 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 16 Apr '09

16 Apr '09

The following changes since commit 9dd175f7d2db1826c891855d3d150da3a5792e94: Takashi Iwai (1): Merge branch 'topic/hda' into for-linus are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Ben Dooks (4): ASoC: Fix jive_wm8750.c build problems ASoC: s3c-i2s-v2.c fix for s3c_i2sv2_iis_calc_rate ASoC: Fix s3c-i2s-v2.c snd_soc_dai changes ASoC: Fix include build error in s3c2412-i2s.c Daniel Mack (1): ASoC: pxa-ssp: allow setting of dai format 0 Jarkko Nikula (2): ASoC: OMAP: Fix DSP_B format in OMAP McBSP DAI driver ASoC: OMAP: Fix FS polarity in OSK5912 machine driver sound/soc/omap/omap-mcbsp.c | 7 +++++-- sound/soc/omap/osk5912.c | 4 ++-- sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/s3c24xx/jive_wm8750.c | 12 ++++++------ sound/soc/s3c24xx/s3c-i2s-v2.c | 18 ++++++++++-------- sound/soc/s3c24xx/s3c2412-i2s.c | 2 +- 6 files changed, 25 insertions(+), 19 deletions(-)

2 1

[alsa-devel] [PATCH v2 6/6] ARM: DaVinci: ASoC: Introduce platform driver model for dm644x, dm355
by Naresh Medisetty 16 Apr '09

16 Apr '09

From: Naresh.Medisetty <naresh(a)ti.com> Introduce the platform driver model to get platform data for dm355 and dm644x. Register platform driver and acquire the resources in the probe function Since the platform specific code had been moved from machine driver to dm<soc>.c Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/davinci-i2s.c | 97 +++++++++++++++++++++------------------ 1 files changed, 52 insertions(+), 45 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index ffdb943..f5cea59 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -416,16 +416,40 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_trigger, + .hw_params = davinci_i2s_hw_params, + .set_fmt = davinci_i2s_set_dai_fmt, + +}; + +struct snd_soc_dai davinci_i2s_dai = { + .name = "davinci-i2s", + .id = 0, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_i2s_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_i2s_dai); + +static int davinci_i2s_probe(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai; + int ret; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea; - struct evm_snd_platform_data *pdata; - int ret; + struct evm_snd_platform_data *pdata = pdev->dev.platform_data;; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { @@ -446,9 +470,7 @@ static int davinci_i2s_probe(struct platform_device *pdev, goto err_release_region; } - cpu_dai->private_data = dev; - - dev->clk = clk_get(&pdev->dev, "McBSPCLK"); + dev->clk = clk_get(&pdev->dev, pdata->clk_name); if (IS_ERR(dev->clk)) { ret = -ENODEV; goto err_free_mem; @@ -456,7 +478,6 @@ static int davinci_i2s_probe(struct platform_device *pdev, clk_enable(dev->clk); dev->base = (void __iomem *)IO_ADDRESS(mem->start); - pdata = pdev->dev.platform_data; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = pdata->tx_dma_ch; @@ -467,7 +488,10 @@ static int davinci_i2s_probe(struct platform_device *pdev, dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = pdata->rx_dma_ch; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG); - + davinci_i2s_dai.private_data = dev; + ret = snd_soc_register_dai(&davinci_i2s_dai); + if (ret != 0) + goto err_free_mem; return 0; err_free_mem: @@ -478,15 +502,13 @@ err_release_region: return ret; } -static void davinci_i2s_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int davinci_i2s_remove(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai; - struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = davinci_i2s_dai.private_data; struct resource *mem; + snd_soc_unregister_dai(&davinci_i2s_dai); + clk_disable(dev->clk); clk_put(dev->clk); dev->clk = NULL; @@ -495,45 +517,30 @@ static void davinci_i2s_remove(struct platform_device *pdev, mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); release_mem_region(mem->start, (mem->end - mem->start) + 1); + return 0; } -#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 -static struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, - .trigger = davinci_i2s_trigger, - .hw_params = davinci_i2s_hw_params, - .set_fmt = davinci_i2s_set_dai_fmt, +static struct platform_driver davinci_asoc_driver = { + .probe = davinci_i2s_probe, + .remove = davinci_i2s_remove, + .driver = { + .name = "davinci-asoc", + .owner = THIS_MODULE, + }, }; -struct snd_soc_dai davinci_i2s_dai = { - .name = "davinci-i2s", - .id = 0, - .probe = davinci_i2s_probe, - .remove = davinci_i2s_remove, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = DAVINCI_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = DAVINCI_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &davinci_i2s_dai_ops, -}; -EXPORT_SYMBOL_GPL(davinci_i2s_dai); - static int __init davinci_i2s_init(void) { - return snd_soc_register_dai(&davinci_i2s_dai); + return platform_driver_register(&davinci_asoc_driver); + } module_init(davinci_i2s_init); static void __exit davinci_i2s_exit(void) { - snd_soc_unregister_dai(&davinci_i2s_dai); + platform_driver_unregister(&davinci_asoc_driver); + } module_exit(davinci_i2s_exit); -- 1.5.6

2 1

[alsa-devel] [PATCH v2 5/6] ARM: DaVinci: ASoC: Add machine driver support for DM646x
by Naresh Medisetty 16 Apr '09

16 Apr '09

From: Naresh.Medisetty <naresh(a)ti.com> This patch does the following: (1) Add support for the DM646x machine (2)Modifications required to introduce the platform driver model to get platform data for all the machines including dm355 and dm644x. Signed-off-by: Steve Chen <schen(a)mvista.com> Signed-off-by: Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/davinci-evm.c | 99 ++++++++++++++++++++++++++++----------- 1 files changed, 72 insertions(+), 27 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 9b90b34..bb26abe 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -20,11 +20,19 @@ #include <sound/soc-dapm.h> #include <asm/dma.h> -#include <mach/hardware.h> +#include <asm/mach-types.h> + +#include <mach/asp.h> +#include <mach/edma.h> +#include <mach/mux.h> +#include <mach/cpu.h> + #include "../codecs/tlv320aic3x.h" +#include "../codecs/spdif_transciever.h" #include "davinci-pcm.h" #include "davinci-i2s.h" +#include "davinci-i2s-mcasp.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ @@ -39,7 +47,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -140,6 +148,25 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm6467_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = davinci_iis_mcasp_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "MCASP S/PDIF", + .stream_name = "spdif", + .cpu_dai = &davinci_iis_mcasp_dai[1], + .codec_dai = &dit_stub_dai, + .ops = &evm_ops, + }, +}; + + /* davinci-evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", @@ -148,12 +175,27 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm6467 evm audio machine driver */ +static struct snd_soc_card dm6467_snd_soc_card_evm = { + .name = "DaVinci DM6467 EVM", + .platform = &davinci_soc_platform, + .dai_link = dm6467_evm_dai, + .num_links = 2, +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { - .i2c_bus = 0, + .i2c_bus = 1, .i2c_address = 0x1b, }; +/* dm6467 evm audio private data */ +static struct aic3x_setup_data dm6467_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -161,51 +203,54 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; -static struct resource evm_snd_resources[] = { - { - .start = DAVINCI_MCBSP_BASE, - .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data evm_snd_data = { - .tx_dma_ch = DM644X_DMACH_MCBSP_TX, - .rx_dma_ch = DM644X_DMACH_MCBSP_RX, +/* evm audio subsystem */ +static struct snd_soc_device dm6467_evm_snd_devdata = { + .card = &dm6467_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &dm6467_evm_aic3x_setup, }; static struct platform_device *evm_snd_device; static int __init evm_init(void) { + struct snd_soc_device *evm_snd_dev_data; + int index; int ret; - evm_snd_device = platform_device_alloc("soc-audio", 0); + if (machine_is_davinci_evm()) { + davinci_cfg_reg(DM644X_MCBSP); + evm_snd_dev_data = &evm_snd_devdata; + index = 0; + } else if (machine_is_davinci_dm355_evm()) { + /* we don't use ASP1 IRQs, or we'd need to mux them ... */ + davinci_cfg_reg(DM355_EVT8_ASP1_TX); + davinci_cfg_reg(DM355_EVT9_ASP1_RX); + evm_snd_dev_data = &evm_snd_devdata; + index = 1; + } else if (machine_is_davinci_dm6467_evm()) { + evm_snd_dev_data = &dm6467_evm_snd_devdata; + index = 0; + } else + return -EINVAL; + evm_snd_device = platform_device_alloc("soc-audio", index); if (!evm_snd_device) return -ENOMEM; - platform_set_drvdata(evm_snd_device, &evm_snd_devdata); - evm_snd_devdata.dev = &evm_snd_device->dev; - platform_device_add_data(evm_snd_device, &evm_snd_data, - sizeof(evm_snd_data)); - - ret = platform_device_add_resources(evm_snd_device, evm_snd_resources, - ARRAY_SIZE(evm_snd_resources)); - if (ret) { - platform_device_put(evm_snd_device); - return ret; - } - + platform_set_drvdata(evm_snd_device, evm_snd_dev_data); + evm_snd_dev_data->dev = &evm_snd_device->dev; ret = platform_device_add(evm_snd_device); if (ret) platform_device_put(evm_snd_device); return ret; + } static void __exit evm_exit(void) { platform_device_unregister(evm_snd_device); + } module_init(evm_init); -- 1.5.6

2 1