October 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCHv2 5/7] ASoC: TWL6030: Add support for low-power mode
by Lopez Cruz, Misael 14 Oct '09

14 Oct '09

TWL6030 codec supports two power modes: low-power and high performance. In low-power mode, headset downlink must be the only path enabled and components in that path (headset DAC and driver) should be in that mode too. In this mode, codec can stream audio at 44.1 and 48 kHz if sys clock from CLK32K is configured to 17.64 and 19.2 MHz using low-power PLL, respectively. In high-performance mode, codec can only work at 19.2 MHz from high-performance PLL. All paths supported in the codec can be used but audio can be streamed only at 48 kHz. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 214 ++++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/twl6030.h | 17 ++++ 2 files changed, 202 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index f1e333f..032619d 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -46,6 +46,7 @@ struct twl6030_data { struct snd_soc_codec codec; int codec_powered; + unsigned int sysclk; }; /* @@ -305,6 +306,29 @@ static void twl6030_power_down(struct snd_soc_codec *codec) mdelay(10); } +/* set headset dac and driver power mode */ +static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) +{ + int hslctl, hsrctl; + int mask = TWL6030_HSDRVMODEL | TWL6030_HSDACMODEL; + + hslctl = twl6030_read_reg_cache(codec, TWL6030_REG_HSLCTL); + hsrctl = twl6030_read_reg_cache(codec, TWL6030_REG_HSRCTL); + + if (high_perf) { + hslctl &= ~mask; + hsrctl &= ~mask; + } else { + hslctl |= mask; + hsrctl |= mask; + } + + twl6030_write(codec, TWL6030_REG_HSLCTL, hslctl); + twl6030_write(codec, TWL6030_REG_HSRCTL, hsrctl); + + return 0; +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -598,6 +622,7 @@ static int twl6030_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct twl6030_data *priv = codec->private_data; int rate, format; /* hardware dai (McPDM) requires bit stream of twice @@ -607,7 +632,20 @@ static int twl6030_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); switch (rate) { case 44100: + if (priv->sysclk != 17640000) { + dev_err(codec->dev, + "rate %d not supported at current sysclk %d\n", + rate, priv->sysclk); + return -EINVAL; + } + break; case 48000: + if (priv->sysclk != 19200000) { + dev_err(codec->dev, + "rate %d not supported at current sysclk %d\n", + rate, priv->sysclk); + return -EINVAL; + } break; default: dev_err(codec->dev, "unknown rate %d\n", rate); @@ -630,46 +668,163 @@ static int twl6030_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 twl6030_data *priv = codec->private_data; u8 hppll, lppll; + lppll = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); hppll = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); - hppll &= TWL6030_HPLLRST; - - switch (freq) { - case 12000000: - /* MCLK input, PLL enabled */ - hppll = TWL6030_MCLK_12000KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLENA; - break; - case 19200000: - /* MCLK input, PLL disabled */ - hppll = TWL6030_MCLK_19200KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLBP; + + switch (clk_id) { + case TWL6030_SYSCLK_SEL_LPPLL: + if (freq != 32768) { + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* CLK32K input requires low-power PLL */ + lppll |= TWL6030_LPLLENA | TWL6030_LPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + mdelay(5); + lppll &= ~TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* headset dac and driver must be in low-power mode */ + headset_power_mode(codec, 0); + + hppll &= ~TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); break; - case 26000000: - /* MCLK input, PLL enabled */ - hppll = TWL6030_MCLK_26000KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLENA; + case TWL6030_SYSCLK_SEL_HPPLL: + switch (freq) { + case 12000000: + hppll = TWL6030_MCLK_12000KHZ; + break; + case 26000000: + hppll = TWL6030_MCLK_26000KHZ; + break; + default: + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* 12 and 26 MHz freqs require high-performance PLL */ + hppll |= TWL6030_HPLLSQRBP | TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + udelay(500); + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + lppll |= TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + lppll &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* only 19.2 MHz can be generated by HPPLL */ + priv->sysclk = 19200000; break; - case 38400000: - /* clk slicer input, PLL disabled */ - hppll = TWL6030_MCLK_38400KHZ - | TWL6030_HPLLSQRENA - | TWL6030_HPLLBP; + case TWL6030_SYSCLK_SEL_MCLK: + switch (freq) { + case 19200000: + hppll = TWL6030_MCLK_19200KHZ | TWL6030_HPLLSQRBP; + break; + case 38400000: + hppll = TWL6030_MCLK_38400KHZ; + break; + default: + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* 19.2 and 38.4 MHz freqs don't require PLL */ + hppll |= TWL6030_HPLLBP; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + udelay(500); + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + lppll |= TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + lppll &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* only 19.2 MHz can be generated by MCLK */ + priv->sysclk = 19200000; break; default: - dev_err(codec->dev, "unknown sysclk rate %d\n", freq); + dev_err(codec->dev, "unknown clk_id %d\n", clk_id); return -EINVAL; } - twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + return 0; +} + +static int twl6030_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; + struct twl6030_data *priv = codec->private_data; + int div; + u8 ldoctl, lppllctl, hppllctl; + + ldoctl = twl6030_read_reg_cache(codec, TWL6030_REG_LDOCTL); + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + hppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); + + if (freq_in && freq_out) { + /* div = round(freq_out / freq_in) */ + div = (freq_out + (freq_in >> 2)) / freq_in; + + switch (pll_id) { + case TWL6030_LPPLL_ID: + if (div < 512) { + dev_err(codec->dev, "invalid pll div %d\n", + div); + return -EINVAL; + } + twl6030_write(codec, TWL6030_REG_LPPLLDIV, div - 512); + lppllctl |= TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + + priv->sysclk = freq_out; + break; + case TWL6030_HPPLL_ID: + hppllctl |= TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); + break; + default: + dev_err(codec->dev, "unknown pll id %d\n", pll_id); + return -EINVAL; + } - /* Disable LPPLL and select HPPLL */ - lppll = TWL6030_HPLLSEL; - twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + /* switch to pll */ + ldoctl &= ~TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + } else { + /* switch to internal oscillator */ + ldoctl |= TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + + /* turn-off pll */ + switch (pll_id) { + case TWL6030_HPPLL_ID: + hppllctl &= ~TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); + break; + case TWL6030_LPPLL_ID: + lppllctl &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + break; + default: + dev_err(codec->dev, "unknown pll id %d\n", pll_id); + return -EINVAL; + } + + priv->sysclk = 0; + } return 0; } @@ -689,6 +844,7 @@ static int twl6030_set_dai_fmt(struct snd_soc_dai *codec_dai, static struct snd_soc_dai_ops twl6030_dai_ops = { .hw_params = twl6030_hw_params, .set_sysclk = twl6030_set_dai_sysclk, + .set_pll = twl6030_set_dai_pll, .set_fmt = twl6030_set_dai_fmt, }; diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 7375ae8..15d3e1b 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -92,6 +92,10 @@ #define TWL6030_MCLK_38400KHZ (3 << 5) #define TWL6030_MCLK_MSK 0x60 +#define TWL6030_SYSCLK_SEL_LPPLL 1 +#define TWL6030_SYSCLK_SEL_HPPLL 2 +#define TWL6030_SYSCLK_SEL_MCLK 3 + /* LPPLLCTL (0x08) fields */ #define TWL6030_LPLLENA 0x01 @@ -100,6 +104,19 @@ #define TWL6030_LPLLFIN 0x08 #define TWL6030_HPLLSEL 0x10 +#define TWL6030_HPPLL_ID 1 +#define TWL6030_LPPLL_ID 2 + +/* HSLCTL (0x10) fields */ + +#define TWL6030_HSDACMODEL 0x02 +#define TWL6030_HSDRVMODEL 0x08 + +/* HSRCTL (0x11) fields */ + +#define TWL6030_HSDACMODER 0x02 +#define TWL6030_HSDRVMODER 0x08 + /* ACCCTL (0x2D) fields */ #define TWL6030_RESETSPLIT 0x04 -- 1.5.4.3

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[alsa-devel] [PATCH 09/10] [PATCH 04/10] ASoC: finally enable support for eXeda and CM-X300
by Mike Rapoport 14 Oct '09

14 Oct '09

From: Igor Grinberg <grinberg(a)compulab.co.il> Signed-off-by: Igor Grinberg <grinberg(a)compulab.co.il> Signed-off-by: Mike Rapoport <mike(a)compulab.co.il> CC: Mark Brown <broonie(a)opensource.wolfsonmicro.com> CC: alsa-devel(a)alsa-project.org --- sound/soc/pxa/Kconfig | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index dcb3181..d4f4031 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -90,7 +90,8 @@ config SND_PXA2XX_SOC_E800 config SND_PXA2XX_SOC_EM_X270 tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" - depends on SND_PXA2XX_SOC && MACH_EM_X270 + depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ + MACH_CM_X300) select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help -- 1.6.0.6

2 1

[alsa-devel] [PATCH 1/1] ALSA: HDA VIA: Remove 48k sample rate limit for S/PDIF
by Logan Li 14 Oct '09

14 Oct '09

From: Logan Li <loganli(a)viatech.com.cn> Subject: ALSA: HDA VIA: Remove 48k sample rate limit for S/PDIF 48 kHz limit is for slightly better stability, and sample rates other than 48k (like 96k/192k) are for better sound quality. We choose better quality, so remove the 48k limit. Signed-off-by: Logan Li <loganli(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 4 ---- 1 file changed, 4 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -4626,7 +4626,6 @@ .substreams = 2, .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -5124,7 +5123,6 @@ .substreams = 2, .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -5561,7 +5559,6 @@ .substreams = 1, .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, @@ -5914,7 +5911,6 @@ .substreams = 1, .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open,

2 1

[alsa-devel] Which transfer mode to use
by Till Crueger 14 Oct '09

14 Oct '09

Hi, First of all, I am fairly new to this list, as well as sound development. So if this list is the wrong place for this question, it would be nice if you could point me in the right direction. I am trying to use ALSA in my OCaml [1] programms. As far as I could see there are no well documented bindings to ALSA for this language, so I am stuck with writing my own bindings. I have gotten to the point where I can open a pcm device with a minimal configuration space on my desktop as well as on my laptop. However I am not sure which transfer mode to use, since it is hard to see for me, what the advantages and disadvantages would be. Here is what I have found out so far. 1. Asynchronous callback: This method seems to be the best explained by all the tutorials I could find, so I am guessing this is the best choice for audio applications. However for my purpose this seems hard to do, since the thread which contains the callback must never enter the OCaml side of the wrapper. This constraint is due to the OCaml garbage collector, which AFAIK would be messed up severely. So to use this method I would have to write an extra buffer in the C side, to which I write the data from OCaml. Depending on the way this buffer is set up -- I was thinking of using a linked list to contain all the single packets -- I would have to keep track what part of the usage of this buffer to be able to synchronise anything to the sound. Not the best choice I as far as I can see. 2. read/write: So far this seems the best choice. This way I could probably rewrite the asynchronous callback method on the OCaml side. For this however I would have to figure out after each polling, how much data can be written. I had a look at the API documentation and it seems ALSA provides function to get this information. However when I had a look at the example sine-wave generator [2], those functions were only used by the mmap method. So the question would be if this information could still be provided when using the read/write method. 3. mmap: During the research I did for this project I found some posts on the portaudio list, that said this method could not be used with all soundcards. Also I would have to figure out if the memory layout of the data passed from the OCaml side is at all compatible to this method. If it is incompatible the use of mmap does not seem to make sense at all. So far this are the points I have gathered to help me make a choice. As you can see I already have a clear preference, but at this point I am still not quite sure. I also found some links that hinted at other issues that would have to be taken into consideration. For example I found some concerns about differences in lattency between these three methods. So what is your opinion on this. Do you think reintroducing the async. callback on the OCaml side would be a good idea, or would you rather use the second buffer that is accesible from the OCaml side. As I said, I haven't done any real sound development, so I am really new to this. Thanks for your help, Till [1] OCaml is a functional language developed by Inria. http://caml.inria.fr/ [2] http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html

2 1

[alsa-devel] Mic/Line-in record is not working with wm9713 AC97
by Sreekanth W 14 Oct '09

14 Oct '09

Hi, I am trying to record a wav file using mic/line-in with wm9713 driver. The Wav file is created, only header is created but almost there is no data. This is seen using CoolEdit/Vim editor. If I try to play the recorded file, there is no sound. But, playing works for other existing files, able to using aplay. I am issuing the following command to record # ./arecord –t wav –c 1 –r 8000 –d 10 –f S16_LE test.wav The below is the dump of /sys/devices/platform/soc-audio/dapm_widget, while "arecord" is running (using mic): # cat /sys/devices/platform/soc-audio/dapm_widget Mic Bias: Off Mic B Pre Amp: Off Mic A Pre Amp: On Mic B PGA: Off Mic A PGA: Off Mono In: Off Right Line In: Off Left Line In: Off Mono Out: Off Out 4: Off Out 3: Off Right Speaker: Off Left Speaker: Off Right Headphone: Off Left Headphone: Off Right Voice ADC: Off Left Voice ADC: Off Right HiFi ADC: On Left HiFi ADC: On Right ADC: On Left ADC: On Aux DAC: Off Voice DAC: Off Capture Mixer: Off Line Mixer: Off HP Mixer: Off AC97 Mixer: Off Right DAC: Off Left DAC: Off Speaker Mixer: Off Mono Mixer: Off Right HP Mixer: Off Left HP Mixer: Off PM State: On I'm using DMA for data transfer, and getting the callback function after DMA completed. Also I'm printing the number of Bytes transfered. Any idea why record is not working? Anything wrong with these settings? Regards, Sreekanth

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[alsa-devel] Channel conversion in plughw not working for my driver... confused about snd_pcm_plug_hw_params()
by David Huggins-Daines 14 Oct '09

14 Oct '09

Hi, I'm trying to make mono output work on the audio driver for the Cirrus EDB9302 evaluation board. For some reason, plughw is refusing to do any channel conversion. When I try to play a mono file, I simply get: aplay: set_params:1022: Unable to install hw params: The driver's struct snd_pcm_hardware contains: .channels_min = 2, .channels_max = 2, which I believe is the correct way to tell userspace "this driver can only do stereo output" - at least, this is how intel8x0.c does it, and that driver really ought to work by now! I've traced this to the exact point of failure, and what I see is that in snd_pcm_plug_hw_params(), despite figuring out exactly what the set of hardware parameters will work for the slave device, it goes and calls _snd_pcm_hw_params() with the actual parameters requested by the client, which obviously fails, and leads to the error shown above. I added some calls to snd_pcm_hw_params_dump() which give this output. The first is sparams, which is the refined set of parameters which I assume are supposed to be set in the slave device. The second is params, which is the set of parameters requested by the client (aplay in this case). sparams are: ACCESS: RW_INTERLEAVED FORMAT: U8 SUBFORMAT: STD SAMPLE_BITS: 8 FRAME_BITS: 16 CHANNELS: 2 RATE: 11025 PERIOD_TIME: (124988 124989) PERIOD_SIZE: 1378 PERIOD_BYTES: 2756 PERIODS: (4 5) BUFFER_TIME: (500045 500046) BUFFER_SIZE: 5513 BUFFER_BYTES: 11026 TICK_TIME: 0 params are: ACCESS: RW_INTERLEAVED FORMAT: U8 SUBFORMAT: STD SAMPLE_BITS: 8 FRAME_BITS: 8 CHANNELS: 1 RATE: 11025 PERIOD_TIME: (124988 124989) PERIOD_SIZE: 1378 PERIOD_BYTES: 1378 PERIODS: (4 5) BUFFER_TIME: (500045 500046) BUFFER_SIZE: 5513 BUFFER_BYTES: 5513 TICK_TIME: 0 inserting plugins whatever that means in snd_pcm_plug_change_channels! So, yeah, that's great, it figured it all out, and realized that it had to do channel conversion, which presumably is done in snd_pcm_plug_change_channels(). But now, we see that it goes ahead and tries to set these parameters in the slave device: calling _snd_pcm_hw_params with params: ACCESS: RW_INTERLEAVED FORMAT: U8 SUBFORMAT: STD SAMPLE_BITS: 8 FRAME_BITS: 8 CHANNELS: 1 RATE: 11025 PERIOD_TIME: (124988 124989) PERIOD_SIZE: 1378 PERIOD_BYTES: 1378 PERIODS: (4 5) BUFFER_TIME: (500045 500046) BUFFER_SIZE: 5513 BUFFER_BYTES: 5513 TICK_TIME: 0 But that's wrong! The slave device can't do single-channel input! So what gives? This can't possibly be broken, or nobody's Intel on-board audio would ever work at all... so what is my driver doing wrong?

1 0

[alsa-devel] SPDIF with X-Fi
by Daniel L. Miller 14 Oct '09

14 Oct '09

Perhaps I'm going about this the wrong way (if so, please tell me), but... My X-Fi card is now working with the new driver. I am connected to my amplifier via a fiber optic cable. Sending standard two-channel audio to the "front" alsa plug gets directed to the spdif output and sounds great. But I want to have my full surround active - particularly for movies already encoded with dts and/or ac-3. Do I need to specially configure ALSA is any fashion to make this work? Do I need to use the iecset tool? I'm using Kubuntu Karmic, running a 2.6.31 kernel, if that makes any difference. -- Daniel

2 1

[alsa-devel] [PATCH 1/1] Echoaudio: fix broken Mia driver
by Jordan 14 Oct '09

14 Oct '09

2 1

[alsa-devel] [PATCH] aaci: ARM1176 aaci-pl041 AC97 register read timeout
by Philby John 13 Oct '09

13 Oct '09

From: Philby John <pjohn(a)in.mvista.com> Date: Tue, 13 Oct 2009 16:30:22 +0530 Subject: [PATCH] aaci: ARM1176 aaci-pl041 AC97 register read timeout After a reboot on an ARM1176 which amounts to a softreset, it has been noted that the ALSA driver does not get registered and the probe fails with the error "aaci-pl041 fpga:04: ac97 read back fail". In the process of reading from a register the SL1TxBusy bit is set indicating that the transceiver is busy and remains so until the default timeout occurs. Set the Power down register 0x26 to an arbitrary value as specified in the PL041 manual (page: 3-18) so that AACISL1TX/AACISL2TX registers take their default state. Signed-off-by: Philby John <pjohn(a)in.mvista.com> --- sound/arm/aaci.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 89096e8..98b6ffe 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -937,6 +937,7 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci) struct snd_ac97 *ac97; int ret; + writel(0, aaci->base + AC97_POWERDOWN); /* * Assert AACIRESET for 2us */ -- 1.6.3.3.MVISTA

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[alsa-devel] S3C: Remove <plat/audio.h>
by Ben Dooks 13 Oct '09

13 Oct '09

Remove the <plat/audio.h> include from arch/arm/plat-s3c/include/plat/audio.h as it provides nothing to the current kernel and is not in any future plans for the system. Signed-off-by: Ben Dooks <ben(a)simtec.co.uk> Signed-off-by: Simtec Linux Team <linux(a)simtec.co.uk> --- arch/arm/plat-s3c/include/plat/audio.h | 45 --------------------------------- sound/soc/s3c24xx/neo1973_wm8753.c | 1 sound/soc/s3c24xx/s3c-i2s-v2.c | 1 sound/soc/s3c24xx/s3c2412-i2s.c | 1 sound/soc/s3c24xx/s3c2443-ac97.c | 1 sound/soc/s3c24xx/s3c24xx-i2s.c | 2 - sound/soc/s3c24xx/s3c24xx-pcm.c | 1 sound/soc/s3c24xx/s3c64xx-i2s.c | 1 8 files changed, 1 insertion(+), 52 deletions(-) Index: b/arch/arm/plat-s3c/include/plat/audio.h =================================================================== --- a/arch/arm/plat-s3c/include/plat/audio.h 2009-10-12 21:11:08.000000000 +0100 +++ /dev/null 1970-01-01 00:00:00.000000000 +0000 @@ -1,45 +0,0 @@ -/* arch/arm/mach-s3c2410/include/mach/audio.h - * - * Copyright (c) 2004-2005 Simtec Electronics - * http://www.simtec.co.uk/products/SWLINUX/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * S3C24XX - Audio platfrom_device info - * - * 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 __ASM_ARCH_AUDIO_H -#define __ASM_ARCH_AUDIO_H __FILE__ - -/* struct s3c24xx_iis_ops - * - * called from the s3c24xx audio core to deal with the architecture - * or the codec's setup and control. - * - * the pointer to itself is passed through in case the caller wants to - * embed this in an larger structure for easy reference to it's context. -*/ - -struct s3c24xx_iis_ops { - struct module *owner; - - int (*startup)(struct s3c24xx_iis_ops *me); - void (*shutdown)(struct s3c24xx_iis_ops *me); - int (*suspend)(struct s3c24xx_iis_ops *me); - int (*resume)(struct s3c24xx_iis_ops *me); - - int (*open)(struct s3c24xx_iis_ops *me, struct snd_pcm_substream *strm); - int (*close)(struct s3c24xx_iis_ops *me, struct snd_pcm_substream *strm); - int (*prepare)(struct s3c24xx_iis_ops *me, struct snd_pcm_substream *strm, struct snd_pcm_runtime *rt); -}; - -struct s3c24xx_platdata_iis { - const char *codec_clk; - struct s3c24xx_iis_ops *ops; - int (*match_dev)(struct device *dev); -}; - -#endif /* __ASM_ARCH_AUDIO_H */ Index: b/sound/soc/s3c24xx/s3c-i2s-v2.c =================================================================== --- a/sound/soc/s3c24xx/s3c-i2s-v2.c 2009-10-12 21:11:08.000000000 +0100 +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c 2009-10-12 21:11:39.000000000 +0100 @@ -32,7 +32,6 @@ #include <plat/regs-s3c2412-iis.h> -#include <plat/audio.h> #include <mach/dma.h> #include "s3c-i2s-v2.h" Index: b/sound/soc/s3c24xx/s3c2412-i2s.c =================================================================== --- a/sound/soc/s3c24xx/s3c2412-i2s.c 2009-10-12 21:11:08.000000000 +0100 +++ b/sound/soc/s3c24xx/s3c2412-i2s.c 2009-10-12 21:11:39.000000000 +0100 @@ -34,7 +34,6 @@ #include <plat/regs-s3c2412-iis.h> -#include <plat/audio.h> #include <mach/regs-gpio.h> #include <mach/dma.h> Index: b/sound/soc/s3c24xx/s3c2443-ac97.c =================================================================== --- a/sound/soc/s3c24xx/s3c2443-ac97.c 2009-10-12 21:11:08.000000000 +0100 +++ b/sound/soc/s3c24xx/s3c2443-ac97.c 2009-10-12 21:11:39.000000000 +0100 @@ -32,7 +32,6 @@ #include <plat/regs-ac97.h> #include <mach/regs-gpio.h> #include <mach/regs-clock.h> -#include <plat/audio.h> #include <asm/dma.h> #include <mach/dma.h> Index: b/sound/soc/s3c24xx/s3c24xx-i2s.c =================================================================== --- a/sound/soc/s3c24xx/s3c24xx-i2s.c 2009-10-12 21:11:08.000000000 +0100 +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c 2009-10-12 21:11:39.000000000 +0100 @@ -32,7 +32,7 @@ #include <mach/hardware.h> #include <mach/regs-gpio.h> #include <mach/regs-clock.h> -#include <plat/audio.h> + #include <asm/dma.h> #include <mach/dma.h> Index: b/sound/soc/s3c24xx/s3c24xx-pcm.c =================================================================== --- a/sound/soc/s3c24xx/s3c24xx-pcm.c 2009-10-12 21:11:08.000000000 +0100 +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c 2009-10-12 21:11:39.000000000 +0100 @@ -29,7 +29,6 @@ #include <asm/dma.h> #include <mach/hardware.h> #include <mach/dma.h> -#include <plat/audio.h> #include "s3c24xx-pcm.h" Index: b/sound/soc/s3c24xx/neo1973_wm8753.c =================================================================== --- a/sound/soc/s3c24xx/neo1973_wm8753.c 2009-10-12 21:13:18.000000000 +0100 +++ b/sound/soc/s3c24xx/neo1973_wm8753.c 2009-10-12 21:13:29.000000000 +0100 @@ -29,7 +29,6 @@ #include <mach/regs-clock.h> #include <mach/regs-gpio.h> #include <mach/hardware.h> -#include <plat/audio.h> #include <linux/io.h> #include <mach/spi-gpio.h> Index: b/sound/soc/s3c24xx/s3c64xx-i2s.c =================================================================== --- a/sound/soc/s3c24xx/s3c64xx-i2s.c 2009-10-12 21:13:35.000000000 +0100 +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c 2009-10-12 21:13:40.000000000 +0100 @@ -31,7 +31,6 @@ #include <plat/gpio-bank-d.h> #include <plat/gpio-bank-e.h> #include <plat/gpio-cfg.h> -#include <plat/audio.h> #include <mach/map.h> #include <mach/dma.h> -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

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