From 322e4095c9261d4cf0326f10d8e398d05e66521c Mon Sep 17 00:00:00 2001 From: Hans Verkuil Date: Tue, 22 Jul 2008 16:25:35 -0300 Subject: V4L/DVB (8484): videodev: missed two more usages of the removed 'owner' field. Signed-off-by: Hans Verkuil Signed-off-by: Mauro Carvalho Chehab --- sound/i2c/other/tea575x-tuner.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 87e3aefeddc..187c9527725 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -189,7 +189,6 @@ void snd_tea575x_init(struct snd_tea575x *tea) } memset(&tea->vd, 0, sizeof(tea->vd)); - tea->vd.owner = tea->card->module; strcpy(tea->vd.name, tea->tea5759 ? "TEA5759 radio" : "TEA5757 radio"); tea->vd.type = VID_TYPE_TUNER; tea->vd.release = snd_tea575x_release; -- cgit v1.2.3 From 0ea6bc8d43c9ee3c5384bea184eab020927a5b2c Mon Sep 17 00:00:00 2001 From: Hans Verkuil Date: Sat, 26 Jul 2008 08:26:43 -0300 Subject: V4L/DVB (8523): v4l2-dev: remove unused type and type2 field from video_device The type and type2 fields were unused and so could be removed. Instead add a vfl_type field that contains the type of the video device. Signed-off-by: Hans Verkuil Signed-off-by: Mauro Carvalho Chehab --- sound/i2c/other/tea575x-tuner.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 187c9527725..83e90057270 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -190,7 +190,6 @@ void snd_tea575x_init(struct snd_tea575x *tea) memset(&tea->vd, 0, sizeof(tea->vd)); strcpy(tea->vd.name, tea->tea5759 ? "TEA5759 radio" : "TEA5757 radio"); - tea->vd.type = VID_TYPE_TUNER; tea->vd.release = snd_tea575x_release; video_set_drvdata(&tea->vd, tea); tea->vd.fops = &tea->fops; -- cgit v1.2.3 From f15cbe6f1a4b4d9df59142fc8e4abb973302cf44 Mon Sep 17 00:00:00 2001 From: Paul Mundt Date: Tue, 29 Jul 2008 08:09:44 +0900 Subject: sh: migrate to arch/sh/include/ This follows the sparc changes a439fe51a1f8eb087c22dd24d69cebae4a3addac. Most of the moving about was done with Sam's directions at: http://marc.info/?l=linux-sh&m=121724823706062&w=2 with subsequent hacking and fixups entirely my fault. Signed-off-by: Sam Ravnborg Signed-off-by: Paul Mundt --- sound/sh/aica.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 9ca11332614..54df8baf916 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -42,7 +42,7 @@ #include #include #include -#include +#include #include "aica.h" MODULE_AUTHOR("Adrian McMenamin "); -- cgit v1.2.3 From a7b815169aae65072017efb1fba9dcecc82ba7c1 Mon Sep 17 00:00:00 2001 From: Huang Weiyi Date: Sat, 26 Jul 2008 20:43:01 +0800 Subject: ALSA: sound/soc/pxa/tosa.c: removed duplicated include Removed duplicated include in sound/soc/pxa/tosa.c. Signed-off-by: Huang Weiyi Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/pxa/tosa.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index fe6cca9c9e7..22971a0f040 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -33,7 +33,6 @@ #include #include #include -#include #include "../codecs/wm9712.h" #include "pxa2xx-pcm.h" -- cgit v1.2.3 From be41e941d5f1a48bde7f44d09d56e8d2605f98e1 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Mon, 28 Jul 2008 17:04:39 -0500 Subject: ALSA: asoc: restrict sample rate and size in Freescale MPC8610 sound drivers The Freescale MPC8610 SSI device has the option of using one clock for both transmit and receive (synchronous mode), or independent clocks (asynchronous). The SSI driver, however, programs the SSI into synchronous mode and then tries to program the clock registers independently. The result is that the wrong sample size is usually generated during recording. This patch fixes the discrepancy by restricting the sample rate and sample size of the playback and capture streams. The SSI driver remembers which stream is opened first. When a second stream is opened, that stream is constrained to the same sample rate and size as the first stream. A future version of this driver will lift the sample size restriction. Supporting independent sample rates is more difficult, because only certain codecs provide dual independent clocks. Signed-off-by: Timur Tabi Signed-off-by: Takashi Iwai --- sound/soc/fsl/fsl_dma.c | 7 ++++- sound/soc/fsl/fsl_ssi.c | 74 ++++++++++++++++++++++++++++++++++++++++++------- 2 files changed, 70 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index da2bc590286..7ceea2bba1f 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -132,12 +132,17 @@ struct fsl_dma_private { * Since each link descriptor has a 32-bit byte count field, we set * period_bytes_max to the largest 32-bit number. We also have no maximum * number of periods. + * + * Note that we specify SNDRV_PCM_INFO_JOINT_DUPLEX here, but only because a + * limitation in the SSI driver requires the sample rates for playback and + * capture to be the same. */ static const struct snd_pcm_hardware fsl_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_JOINT_DUPLEX, .formats = FSLDMA_PCM_FORMATS, .rates = FSLDMA_PCM_RATES, .rate_min = 5512, diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 71bff33f552..157a7895ffa 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -67,6 +67,8 @@ * @ssi: pointer to the SSI's registers * @ssi_phys: physical address of the SSI registers * @irq: IRQ of this SSI + * @first_stream: pointer to the stream that was opened first + * @second_stream: pointer to second stream * @dev: struct device pointer * @playback: the number of playback streams opened * @capture: the number of capture streams opened @@ -79,6 +81,8 @@ struct fsl_ssi_private { struct ccsr_ssi __iomem *ssi; dma_addr_t ssi_phys; unsigned int irq; + struct snd_pcm_substream *first_stream; + struct snd_pcm_substream *second_stream; struct device *dev; unsigned int playback; unsigned int capture; @@ -342,6 +346,49 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream) */ } + if (!ssi_private->first_stream) + ssi_private->first_stream = substream; + else { + /* This is the second stream open, so we need to impose sample + * rate and maybe sample size constraints. Note that this can + * cause a race condition if the second stream is opened before + * the first stream is fully initialized. + * + * We provide some protection by checking to make sure the first + * stream is initialized, but it's not perfect. ALSA sometimes + * re-initializes the driver with a different sample rate or + * size. If the second stream is opened before the first stream + * has received its final parameters, then the second stream may + * be constrained to the wrong sample rate or size. + * + * FIXME: This code does not handle opening and closing streams + * repeatedly. If you open two streams and then close the first + * one, you may not be able to open another stream until you + * close the second one as well. + */ + struct snd_pcm_runtime *first_runtime = + ssi_private->first_stream->runtime; + + if (!first_runtime->rate || !first_runtime->sample_bits) { + dev_err(substream->pcm->card->dev, + "set sample rate and size in %s stream first\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "capture" : "playback"); + return -EAGAIN; + } + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + first_runtime->rate, first_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + first_runtime->sample_bits, + first_runtime->sample_bits); + + ssi_private->second_stream = substream; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ssi_private->playback++; @@ -371,18 +418,16 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream) struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u32 wl; - wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); + if (substream == ssi_private->first_stream) { + u32 wl; - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + /* The SSI should always be disabled at this points (SSIEN=0) */ + wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + /* In synchronous mode, the SSI uses STCCR for capture */ clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); - else - clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); - - setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + } return 0; } @@ -407,9 +452,13 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - setbits32(&ssi->scr, CCSR_SSI_SCR_TE); + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); } else { - setbits32(&ssi->scr, CCSR_SSI_SCR_RE); + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); /* * I think we need this delay to allow time for the SSI @@ -452,6 +501,11 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) ssi_private->capture--; + if (ssi_private->first_stream == substream) + ssi_private->first_stream = ssi_private->second_stream; + + ssi_private->second_stream = NULL; + /* * If this is the last active substream, disable the SSI and release * the IRQ. -- cgit v1.2.3 From 877db3c1af24a65f78ae865b1fb642165e065a8b Mon Sep 17 00:00:00 2001 From: Dmitry Baryshkov Date: Tue, 29 Jul 2008 11:42:22 +0100 Subject: ALSA: ASoC: Update Poodle to current ASoC API Signed-off-by: Dmitry Baryshkov Cc: Richard Purdie Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/pxa/poodle.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 65a4e9a8c39..d968cf71b56 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -85,17 +85,13 @@ static int poodle_startup(struct snd_pcm_substream *substream) } /* we need to unmute the HP at shutdown as the mute burns power on poodle */ -static int poodle_shutdown(struct snd_pcm_substream *substream) +static void poodle_shutdown(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; - /* set = unmute headphone */ locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_L, 1); locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 1); - return 0; } static int poodle_hw_params(struct snd_pcm_substream *substream, @@ -232,7 +228,7 @@ static const struct soc_enum poodle_enum[] = { SOC_ENUM_SINGLE_EXT(2, spk_function), }; -static const snd_kcontrol_new_t wm8731_poodle_controls[] = { +static const struct snd_kcontrol_new wm8731_poodle_controls[] = { SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack, poodle_set_jack), SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk, -- cgit v1.2.3 From 11589418a1c4cf68be9367f802898d35e07809c4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 29 Jul 2008 11:42:23 +0100 Subject: ALSA: ASoC: Export dapm_reg_event() fully dapm_reg_event() is used by devices using SND_SOC_DAPM_REG() so needs to be exported to support building them as modules and prototyped to avoid sparse warnings and potential build issues. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 820347c9ae4..f9d100bc847 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -470,6 +470,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, return 0; } +EXPORT_SYMBOL_GPL(dapm_reg_event); /* * Scan each dapm widget for complete audio path. -- cgit v1.2.3 From 82e68f7ffec3800425f2391c8c86277606860442 Mon Sep 17 00:00:00 2001 From: Willy Tarreau Date: Sat, 2 Aug 2008 18:25:16 +0200 Subject: sound: ensure device number is valid in snd_seq_oss_synth_make_info snd_seq_oss_synth_make_info() incorrectly reports information to userspace without first checking for the validity of the device number, leading to possible information leak (CVE-2008-3272). Reported-By: Tobias Klein Acked-and-tested-by: Takashi Iwai Cc: stable@kernel.org Signed-off-by: Willy Tarreau Signed-off-by: Linus Torvalds --- sound/core/seq/oss/seq_oss_synth.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 558dadbf45f..e024e4588b8 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -604,6 +604,9 @@ snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_in { struct seq_oss_synth *rec; + if (dev < 0 || dev >= dp->max_synthdev) + return -ENXIO; + if (dp->synths[dev].is_midi) { struct midi_info minf; snd_seq_oss_midi_make_info(dp, dp->synths[dev].midi_mapped, &minf); -- cgit v1.2.3