March 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/3] ASoC: ak4671: Fix control-less DAPM routes
by Lars-Peter Clausen 05 Mar '15

05 Mar '15

Routes without a control must use NULL for the control name. The ak4671 driver uses "NULL" instead in a few places. Previous to commit 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") the DAPM core silently ignored non-NULL controls on non-mixer and non-mux routes. But starting with that commit it will complain and not add the route breaking the ak4671 driver in the process. This patch replaces the incorrect "NULL" control name with NULL to fix the issue. Fixes: 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ak4671.c | 44 ++++++++++++++++++++++---------------------- 1 file changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 632e89f..2a58b1d 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -343,25 +343,25 @@ static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = { }; static const struct snd_soc_dapm_route ak4671_intercon[] = { - {"DAC Left", "NULL", "PMPLL"}, - {"DAC Right", "NULL", "PMPLL"}, - {"ADC Left", "NULL", "PMPLL"}, - {"ADC Right", "NULL", "PMPLL"}, + {"DAC Left", NULL, "PMPLL"}, + {"DAC Right", NULL, "PMPLL"}, + {"ADC Left", NULL, "PMPLL"}, + {"ADC Right", NULL, "PMPLL"}, /* Outputs */ - {"LOUT1", "NULL", "LOUT1 Mixer"}, - {"ROUT1", "NULL", "ROUT1 Mixer"}, - {"LOUT2", "NULL", "LOUT2 Mix Amp"}, - {"ROUT2", "NULL", "ROUT2 Mix Amp"}, - {"LOUT3", "NULL", "LOUT3 Mixer"}, - {"ROUT3", "NULL", "ROUT3 Mixer"}, + {"LOUT1", NULL, "LOUT1 Mixer"}, + {"ROUT1", NULL, "ROUT1 Mixer"}, + {"LOUT2", NULL, "LOUT2 Mix Amp"}, + {"ROUT2", NULL, "ROUT2 Mix Amp"}, + {"LOUT3", NULL, "LOUT3 Mixer"}, + {"ROUT3", NULL, "ROUT3 Mixer"}, {"LOUT1 Mixer", "DACL", "DAC Left"}, {"ROUT1 Mixer", "DACR", "DAC Right"}, {"LOUT2 Mixer", "DACHL", "DAC Left"}, {"ROUT2 Mixer", "DACHR", "DAC Right"}, - {"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"}, - {"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"}, + {"LOUT2 Mix Amp", NULL, "LOUT2 Mixer"}, + {"ROUT2 Mix Amp", NULL, "ROUT2 Mixer"}, {"LOUT3 Mixer", "DACSL", "DAC Left"}, {"ROUT3 Mixer", "DACSR", "DAC Right"}, @@ -381,18 +381,18 @@ static const struct snd_soc_dapm_route ak4671_intercon[] = { {"LIN2", NULL, "Mic Bias"}, {"RIN2", NULL, "Mic Bias"}, - {"ADC Left", "NULL", "LIN MUX"}, - {"ADC Right", "NULL", "RIN MUX"}, + {"ADC Left", NULL, "LIN MUX"}, + {"ADC Right", NULL, "RIN MUX"}, /* Analog Loops */ - {"LIN1 Mixing Circuit", "NULL", "LIN1"}, - {"RIN1 Mixing Circuit", "NULL", "RIN1"}, - {"LIN2 Mixing Circuit", "NULL", "LIN2"}, - {"RIN2 Mixing Circuit", "NULL", "RIN2"}, - {"LIN3 Mixing Circuit", "NULL", "LIN3"}, - {"RIN3 Mixing Circuit", "NULL", "RIN3"}, - {"LIN4 Mixing Circuit", "NULL", "LIN4"}, - {"RIN4 Mixing Circuit", "NULL", "RIN4"}, + {"LIN1 Mixing Circuit", NULL, "LIN1"}, + {"RIN1 Mixing Circuit", NULL, "RIN1"}, + {"LIN2 Mixing Circuit", NULL, "LIN2"}, + {"RIN2 Mixing Circuit", NULL, "RIN2"}, + {"LIN3 Mixing Circuit", NULL, "LIN3"}, + {"RIN3 Mixing Circuit", NULL, "RIN3"}, + {"LIN4 Mixing Circuit", NULL, "LIN4"}, + {"RIN4 Mixing Circuit", NULL, "RIN4"}, {"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"}, {"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"}, -- 1.8.0

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[alsa-devel] [PATCH RESEND] ASoC: rt5670: Fix the speaker mono output issue
by Bard Liao 04 Mar '15

04 Mar '15

We need to set left/right control for the speaker amp to get stereo output on speaker. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- We need to set different value according to codec version. And the codec version is stored in in RT5670_VENDOR_ID register. --- sound/soc/codecs/rt5670.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 04ca247..ba8cc14 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2700,6 +2700,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, regmap_write(rt5670->regmap, RT5670_RESET, 0); + regmap_read(rt5670->regmap, RT5670_VENDOR_ID, &val); + if (val >= 4) + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0980); + else + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0d00); + ret = regmap_register_patch(rt5670->regmap, init_list, ARRAY_SIZE(init_list)); if (ret != 0) -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: omap-pcm: Allow only formats with 1, 2, and 4 byte physical size
by Jyri Sarha 04 Mar '15

04 Mar '15

sDMA support only transfer elements with 1, 2, and 4 byte physical size. Initialize the pcm driver accordingly. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/omap/omap-pcm.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index f4b05bc..e49ee23 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -39,7 +39,7 @@ #define pcm_omap1510() 0 #endif -static const struct snd_pcm_hardware omap_pcm_hardware = { +static struct snd_pcm_hardware omap_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | @@ -53,6 +53,24 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { .buffer_bytes_max = 128 * 1024, }; +/* sDMA supports only 1, 2, and 4 byte transfer elements. */ +static void omap_pcm_limit_supported_formats(void) +{ + int i; + + for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { + switch (snd_pcm_format_physical_width(i)) { + case 8: + case 16: + case 32: + omap_pcm_hardware.formats |= (1LL << i); + break; + default: + break; + } + } +} + /* this may get called several times by oss emulation */ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -235,6 +253,7 @@ static struct snd_soc_platform_driver omap_soc_platform = { int omap_pcm_platform_register(struct device *dev) { + omap_pcm_limit_supported_formats(); return devm_snd_soc_register_platform(dev, &omap_soc_platform); } EXPORT_SYMBOL_GPL(omap_pcm_platform_register); -- 1.9.1

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Re: [alsa-devel] [ASoC]fsl_ssi: fix of_property_read_u32_array return value check
by Mark Brown 04 Mar '15

04 Mar '15

On Sun, Feb 15, 2015 at 11:11:43PM +0100, Maciej S. Szmigiero wrote: > of_property_read_u32_array returns 0 on success, so the return value shouldn't be inverted twice, > first on assignment then in condition expression. This doesn't apply to current code, please check and resend. Please also keep your changelog under 80 columns as covered in SubmittingPatches.

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[alsa-devel] [PATCH] ASoC: omap-pcm: Correct dma mask
by Peter Ujfalusi 04 Mar '15

04 Mar '15

DMA_BIT_MASK of 64 is not valid dma address mask for OMAPs, it should be set to 32. The 64 was introduced by commit (in 2009): a152ff24b978 ASoC: OMAP: Make DMA 64 aligned But the dma_mask and coherent_dma_mask can not be used to specify alignment. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 54d59e4f8f5f..d52a6d1ae55d 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -227,7 +227,7 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) struct snd_pcm *pcm = rtd->pcm; int ret; - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(64)); + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); if (ret) return ret; -- 2.3.0

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[alsa-devel] [TINYCOMPRESS][PATCH] crec: support streaming output to stdout
by Richard Fitzgerald 04 Mar '15

04 Mar '15

From: Ammar Zahid Ali Syed <aali(a)opensource.wolfsonmicro.com> It's useful for testing to be able to stream the captured audio to stdout instead of having to go via a file. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Ammar Ali <aali(a)opensource.wolfsonmicro.com> --- crec.c | 73 ++++++++++++++++++++++++++++++++++++++++----------------------- 1 files changed, 46 insertions(+), 27 deletions(-) diff --git a/crec.c b/crec.c index e3f189f..258fade 100644 --- a/crec.c +++ b/crec.c @@ -77,6 +77,8 @@ static int verbose; static int file; +static FILE *finfo; +static bool streamed; static const unsigned int DEFAULT_CHANNELS = 1; static const unsigned int DEFAULT_RATE = 44100; @@ -151,7 +153,7 @@ static void size_wave_header(struct wave_header *header, uint32_t size) static void usage(void) { - fprintf(stderr, "usage: crec [OPTIONS] filename\n" + fprintf(stderr, "usage: crec [OPTIONS] [filename]\n" "-c\tcard number\n" "-d\tdevice node\n" "-b\tbuffer size\n" @@ -162,6 +164,8 @@ static void usage(void) "-C\tSpecify the number of channels (default %u)\n" "-R\tSpecify the sample rate (default %u)\n" "-F\tSpecify the format: S16_LE, S32_LE (default S16_LE)\n\n" + "If filename is not given the output is\n" + "written to stdout\n\n" "Example:\n" "\tcrec -c 1 -d 2 test.wav\n" "\tcrec -f 5 test.wav\n", @@ -180,7 +184,7 @@ static int print_time(struct compress *compress) fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); return -1; } else { - printf("DSP recorded %jd.%jd\n", + fprintf(finfo, "DSP recorded %jd.%jd\n", (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); } return 0; @@ -195,6 +199,10 @@ static int finish_record() if (!file) return -ENOENT; + /* can't rewind if streaming to stdout */ + if (streamed) + return 0; + /* Get amount of data written to file */ ret = lseek(file, 0, SEEK_END); if (ret < 0) @@ -256,22 +264,27 @@ void capture_samples(char *name, unsigned int card, unsigned int device, length = length * rate * (samplebits / 8) * channels; if (verbose) - printf("%s: entry, reading %u bytes\n", __func__, length); - - file = open(name, O_RDWR | O_CREAT, S_IRUSR | S_IWUSR | S_IRGRP | S_IWGRP); - if (file == -1) { - fprintf(stderr, "Unable to open file '%s'\n", name); - exit(EXIT_FAILURE); - } + fprintf(finfo, "%s: entry, reading %u bytes\n", __func__, length); + if (!name) { + file = STDOUT_FILENO; + } else { + file = open(name, O_RDWR | O_CREAT, S_IRUSR | S_IWUSR | S_IRGRP | S_IWGRP); + if (file == -1) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + } /* Write a header, will update with size once record is complete */ - init_wave_header(&header, channels, rate, samplebits); - written = write(file, &header, sizeof(header)); - if (written != sizeof(header)) { + if (!streamed) { + init_wave_header(&header, channels, rate, samplebits); + written = write(file, &header, sizeof(header)); + if (written != sizeof(header)) { fprintf(stderr, "Error writing output file header: %s\n", strerror(errno)); goto file_exit; - } + } + } memset(&codec, 0, sizeof(codec)); memset(&config, 0, sizeof(config)); @@ -299,7 +312,7 @@ void capture_samples(char *name, unsigned int card, unsigned int device, }; if (verbose) - printf("%s: Opened compress device\n", __func__); + fprintf(finfo, "%s: Opened compress device\n", __func__); size = config.fragment_size; buffer = malloc(size * config.fragments); @@ -308,15 +321,15 @@ void capture_samples(char *name, unsigned int card, unsigned int device, goto comp_exit; } - printf("Recording file %s On Card %u device %u, with buffer of %lu bytes\n", + fprintf(finfo, "Recording file %s On Card %u device %u, with buffer of %lu bytes\n", name, card, device, buffer_size); - printf("Format %u Channels %u, %u Hz\n", - codec.id, codec.ch_out, rate); + fprintf(finfo, "Codec %u Format %u Channels %u, %u Hz\n", + codec.id, codec.format, codec.ch_out, rate); compress_start(compress); if (verbose) - printf("%s: Capturing audio NOW!!!\n", __func__); + fprintf(finfo, "%s: Capturing audio NOW!!!\n", __func__); do { if (length && size > length - total_read) @@ -344,7 +357,7 @@ void capture_samples(char *name, unsigned int card, unsigned int device, } if (verbose) { print_time(compress); - printf("%s: read %d\n", __func__, read); + fprintf(finfo, "%s: read %d\n", __func__, read); } } } while (!length || total_read < length); @@ -362,7 +375,7 @@ void capture_samples(char *name, unsigned int card, unsigned int device, } if (verbose) - printf("%s: exit success\n", __func__); + fprintf(finfo, "%s: exit success\n", __func__); free(buffer); close(file); @@ -379,7 +392,7 @@ file_exit: close(file); if (verbose) - printf("%s: exit failure\n", __func__); + fprintf(finfo, "%s: exit failure\n", __func__); exit(EXIT_FAILURE); } @@ -408,7 +421,7 @@ int main(int argc, char **argv) exit(EXIT_FAILURE); } - if (argc < 2) + if (argc < 1) usage(); verbose = 0; @@ -456,15 +469,21 @@ int main(int argc, char **argv) exit(EXIT_FAILURE); } } - if (optind >= argc) - usage(); - - file = argv[optind]; + if (optind >= argc) { + file = NULL; + finfo = fopen("/dev/null", "w"); + streamed = true; + } else { + file = argv[optind]; + finfo = stdout; + streamed = false; + } capture_samples(file, card, device, buffer_size, frag, length, rate, channels, format); - printf("Finish capturing... Close Normally\n"); + fprintf(finfo, "Finish capturing... Close Normally\n"); + exit(EXIT_SUCCESS); } -- 1.7.2.5

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[alsa-devel] [PATCH 0/4] Armada 38x and 370 audio
by Marcin Wojtas 04 Mar '15

04 Mar '15

Hi, I send a series of patches adding I2S - S/PDIF support for Marvell Armada 38x SoC. It is same controller as in Armada 370, however two quirks had to be taken into consideration: * new PLL frequency setting * exceptive pinout of I2S and S/PDIF formats Kirkwood-i2s driver was enhanced with obtaining mem resources by name. Sound interface was enabled for Armada 385 DB board, using simple-card DT binding for both I2S and S/PDIF. In addition to that minor fix was applied for Armada 370 audio DT description (stable-CC'ed). Any comments and remarks are welcome. Best regards, Marcin Marcin Wojtas (4): ASoC: kirkwood: enable Kirkwood driver for Armada 38x platforms ARM: mvebu: add audio I2S controller to Armada 38x Device Tree ARM: mvebu: add audio support to Armada 385 DB ARM: mvebu: fix routing for analog audio input of Armada 370 DB platform .../devicetree/bindings/sound/mvebu-audio.txt | 14 ++- arch/arm/boot/dts/armada-370-db.dts | 2 +- arch/arm/boot/dts/armada-388-db.dts | 69 +++++++++++ arch/arm/boot/dts/armada-38x.dtsi | 17 +++ sound/soc/kirkwood/kirkwood-i2s.c | 132 ++++++++++++++++++++- sound/soc/kirkwood/kirkwood.h | 2 + 6 files changed, 232 insertions(+), 4 deletions(-) -- 1.8.3.1

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[alsa-devel] [PATCH 1/2] ucm: document some standard values
by han.lu＠intel.com 04 Mar '15

04 Mar '15

From: "Lu, Han" <han.lu(a)intel.com> add standard value JackDev, JackControl and JackHWMute for speaker mute control on jack insertion. These values will be applied on pulseaudio. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/include/use-case.h b/include/use-case.h index 697377a..3728906 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -284,6 +284,10 @@ int snd_use_case_get_list(snd_use_case_mgr_t *uc_mgr, * PlaybackMixerID - mixer playback ID * CaptureMixer - name of capture mixer * CaptureMixerID - mixer capture ID + * JackDev - jack device name + * JackControl - jack control name + * JackHWMute - indicate if the HW mutes a device on jack insertion + * or not. */ int snd_use_case_get(snd_use_case_mgr_t *uc_mgr, const char *identifier, -- 2.1.0

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[alsa-devel] [PATCH] ALSA: Fix spelling typo in Documentation/DocBook/alsa-driver-api.xml
by Masanari Iida 04 Mar '15

04 Mar '15

This patch fix spelling typo found in alsa-driver-api.xml. It is because this file is generated from comments in source files, I have to fix source files. --- include/sound/compress_driver.h | 4 ++-- include/sound/control.h | 2 +- include/sound/soc.h | 2 +- include/uapi/sound/compress_offload.h | 2 +- sound/core/pcm_dmaengine.c | 4 ++-- 5 files changed, 7 insertions(+), 7 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f48089d..ffafda5b 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -70,7 +70,7 @@ struct snd_compr_runtime { * @device: device pointer * @direction: stream direction, playback/recording * @metadata_set: metadata set flag, true when set - * @next_track: has userspace signall next track transistion, true when set + * @next_track: has userspace signal next track transition, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -95,7 +95,7 @@ struct snd_compr_stream { * and the stream properties * @get_params: retrieve the codec parameters, mandatory * @set_metadata: Set the metadata values for a stream - * @get_metadata: retreives the requested metadata values from stream + * @get_metadata: retrieves the requested metadata values from stream * @trigger: Trigger operations like start, pause, resume, drain, stop. * This callback is mandatory * @pointer: Retrieve current h/w pointer information. Mandatory diff --git a/include/sound/control.h b/include/sound/control.h index 75f3054..95aad6d 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -227,7 +227,7 @@ snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) * Add a virtual slave control to the given master. * Unlike snd_ctl_add_slave(), the element added via this function * is supposed to have volatile values, and get callback is called - * at each time quried from the master. + * at each time queried from the master. * * When the control peeks the hardware values directly and the value * can be changed by other means than the put callback of the element, diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade1..cf0bb15 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1469,7 +1469,7 @@ static inline struct snd_soc_codec *snd_soc_kcontrol_codec( } /** - * snd_soc_kcontrol_platform() - Returns the platform that registerd the control + * snd_soc_kcontrol_platform() - Returns the platform that registered the control * @kcontrol: The control for which to get the platform * * Note: This function will only work correctly if the control has been diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 22ed8cb..e00d8cb 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -75,7 +75,7 @@ struct snd_compr_tstamp { /** * struct snd_compr_avail - avail descriptor * @avail: Number of bytes available in ring buffer for writing/reading - * @tstamp: timestamp infomation + * @tstamp: timestamp information */ struct snd_compr_avail { __u64 avail; diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 6542c40..fba365a 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -289,7 +289,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_request_channel); * * The function should usually be called from the pcm open callback. Note that * this function will use private_data field of the substream's runtime. So it - * is not availabe to your pcm driver implementation. + * is not available to your pcm driver implementation. */ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, struct dma_chan *chan) @@ -328,7 +328,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open); * This function will request a DMA channel using the passed filter function and * data. The function should usually be called from the pcm open callback. Note * that this function will use private_data field of the substream's runtime. So - * it is not availabe to your pcm driver implementation. + * it is not available to your pcm driver implementation. */ int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data) -- 2.3.1.167.g7f4ba4b

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[alsa-devel] [PATCH 00/10] initial clkdev cleanups
by Russell King - ARM Linux 04 Mar '15

04 Mar '15

Here's some initial clkdev cleanups. These are targetted for the next merge window, and while the initial patches can be merged independently, I'd prefer to keep the series together as further work on solving the problems which unique struct clk's has introduced is needed. The initial cleanups are more about using the correct clkdev function than anything else: there's no point interating over a array of clk_lookup structs, adding each one in turn when we have had a function which does this since forever. I'm also killing a chunk of seemingly unused code in the omap3isp driver. Lastly, I'm introducing a clkdev_create() helper, which combines the clkdev_alloc() + clkdev_add() pattern which keeps cropping up. Individual patches copied to appropriate people, but they will all appear on the mailing lists. arch/arm/mach-lpc32xx/clock.c | 5 +-- arch/arm/mach-omap2/clkt2xxx_virt_prcm_set.c | 12 ++----- arch/arm/mach-omap2/omap_device.c | 24 +++++-------- arch/arm/plat-orion/common.c | 6 +--- arch/sh/kernel/cpu/sh4a/clock-sh7734.c | 3 +- arch/sh/kernel/cpu/sh4a/clock-sh7757.c | 4 +-- arch/sh/kernel/cpu/sh4a/clock-sh7785.c | 4 +-- arch/sh/kernel/cpu/sh4a/clock-sh7786.c | 4 +-- arch/sh/kernel/cpu/sh4a/clock-shx3.c | 4 +-- drivers/clk/clk-s2mps11.c | 4 +-- drivers/clk/clkdev.c | 52 +++++++++++++++++++++------- drivers/clk/versatile/clk-impd1.c | 4 +-- drivers/media/platform/omap3isp/isp.c | 18 ---------- drivers/media/platform/omap3isp/isp.h | 1 - include/linux/clkdev.h | 3 ++ include/media/omap3isp.h | 6 ---- sound/soc/sh/migor.c | 3 +- 17 files changed, 68 insertions(+), 89 deletions(-) -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

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