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author | Linus Torvalds <torvalds@linux-foundation.org> | 2014-08-06 20:07:24 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2014-08-06 20:07:24 -0700 |
commit | 930e0312bcdc96d15f02ed6812d4a6c947855a2d (patch) | |
tree | d2d620c06359510562b25987cf329c77e41b7c11 /sound/core | |
parent | ec6c0a77786524e44003e70ea69651ad7fb35aec (diff) | |
parent | a509574e5ea7b617268943526773ebf7e2d20a9b (diff) | |
download | talos-op-linux-930e0312bcdc96d15f02ed6812d4a6c947855a2d.tar.gz talos-op-linux-930e0312bcdc96d15f02ed6812d4a6c947855a2d.zip |
Merge tag 'sound-3.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"There've been many updates in ASoC side at this time, especially the
framework enhancement for multiple CODECs on a single DAI and more
componentization works.
The only major change in ALSA core is the addition of timestamp type
in sw_params field. This should behave in backward compatible way.
Other than that, there are lots of small changes and new drivers in
wide range, including a large code cut in HD-audio driver for
deprecated static quirks. Some highlights are below:
ALSA Core:
- Add the new timestamp type field to sw_params to choose
MONOTONIC_RAW type
HD-audio:
- Continued conversion to standard printk macros, generic code
cleanups
- Removal of obsoleted static quirk codes for Conexant and C-Media
codecs
- Fixups for HP Envy TS, Dell XPS 15, HP and Dell mute/mic LED,
Gigabyte BXBT-2807 mobo
- Intel Braswell support
ASoC:
- Support for multiple CODECs attached to a single DAI, enabling
systems with for example multiple DAC/speaker drivers on a single
link, contributed by Benoit Cousson based on work from Misael Lopez
Cruz
- Support for byte controls larger than 256 bytes based on the use of
TLVs contributed by Omair Mohammed Abdullah
- More componentisation work from Lars-Peter Clausen
- The remainder of the conversions of CODEC drivers to params_width()
by Mark Brown
- Drivers for Cirrus Logic CS4265, Freescale i.MX ASRC blocks,
Realtek RT286 and RT5670, Rockchip RK3xxx I2S controllers and Texas
Instruments TAS2552
- Lots of updates and fixes, especially to the DaVinci, Intel,
Freescale, Realtek, and rcar drivers"
* tag 'sound-3.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (402 commits)
ALSA: usb-audio: Whitespace cleanups for sound/usb/midi.*
ALSA: usb-audio: Respond to suspend and resume callbacks for MIDI input
sound/oss/pss: Remove typedefs pss_mixerdata and pss_confdata
sound/oss/opl3: Remove typedef opl_devinfo
ALSA: fireworks: fix specifiers in format strings for propper output
ASoC: imx-audmux: Use uintptr_t for port numbers
ASoC: davinci: Enable menuconfig entry for McASP
ASoC: fsl_asrc: Don't access members of config before checking it
ASoC: fsl_sarc_dma: Check pair before using it
ASoC: adau1977: Fix truncation warning on 64 bit architectures
ALSA: virtuoso: add Xonar Essence STX II support
ALSA: riptide: fix %d confusingly prefixed with 0x in format strings
ALSA: fireworks: fix %d confusingly prefixed with 0x in format strings
ALSA: hda - add codec ID for Braswell display audio codec
ALSA: hda - add PCI IDs for Intel Braswell
ALSA: usb-audio: Adjust Gamecom 780 volume level
ALSA: usb-audio: improve dmesg source grepability
ASoC: rt5670: Fix duplicate const warnings
ASoC: rt5670: Staticise non-exported symbols
ASoC: Intel: update stream only on stream IPC msgs
...
Diffstat (limited to 'sound/core')
-rw-r--r-- | sound/core/compress_offload.c | 2 | ||||
-rw-r--r-- | sound/core/control.c | 6 | ||||
-rw-r--r-- | sound/core/pcm_compat.c | 8 | ||||
-rw-r--r-- | sound/core/pcm_dmaengine.c | 4 | ||||
-rw-r--r-- | sound/core/pcm_native.c | 9 | ||||
-rw-r--r-- | sound/core/seq/seq_memory.c | 4 |
6 files changed, 21 insertions, 12 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 7403f348ed14..89028fab64fd 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -491,7 +491,7 @@ static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ if (params->buffer.fragment_size == 0 || - params->buffer.fragments > SIZE_MAX / params->buffer.fragment_size) + params->buffer.fragments > INT_MAX / params->buffer.fragment_size) return -EINVAL; /* now codec parameters */ diff --git a/sound/core/control.c b/sound/core/control.c index f0b0e14497a5..b9611344ff9e 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1406,11 +1406,11 @@ static long snd_ctl_ioctl(struct file *file, unsigned int cmd, unsigned long arg case SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS: return snd_ctl_subscribe_events(ctl, ip); case SNDRV_CTL_IOCTL_TLV_READ: - return snd_ctl_tlv_ioctl(ctl, argp, 0); + return snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_READ); case SNDRV_CTL_IOCTL_TLV_WRITE: - return snd_ctl_tlv_ioctl(ctl, argp, 1); + return snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_WRITE); case SNDRV_CTL_IOCTL_TLV_COMMAND: - return snd_ctl_tlv_ioctl(ctl, argp, -1); + return snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_CMD); case SNDRV_CTL_IOCTL_POWER: return -ENOPROTOOPT; case SNDRV_CTL_IOCTL_POWER_STATE: diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index af49721ba0e3..102e8fd1d450 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -101,7 +101,9 @@ struct snd_pcm_sw_params32 { u32 silence_threshold; u32 silence_size; u32 boundary; - unsigned char reserved[64]; + u32 proto; + u32 tstamp_type; + unsigned char reserved[56]; }; /* recalcuate the boundary within 32bit */ @@ -133,7 +135,9 @@ static int snd_pcm_ioctl_sw_params_compat(struct snd_pcm_substream *substream, get_user(params.start_threshold, &src->start_threshold) || get_user(params.stop_threshold, &src->stop_threshold) || get_user(params.silence_threshold, &src->silence_threshold) || - get_user(params.silence_size, &src->silence_size)) + get_user(params.silence_size, &src->silence_size) || + get_user(params.tstamp_type, &src->tstamp_type) || + get_user(params.proto, &src->proto)) return -EFAULT; /* * Check silent_size parameter. Since we have 64bit boundary, diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 76cbb9ec953a..6542c4083594 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -65,13 +65,15 @@ int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, enum dma_slave_buswidth buswidth; int bits; - bits = snd_pcm_format_physical_width(params_format(params)); + bits = params_physical_width(params); if (bits < 8 || bits > 64) return -EINVAL; else if (bits == 8) buswidth = DMA_SLAVE_BUSWIDTH_1_BYTE; else if (bits == 16) buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + else if (bits == 24) + buswidth = DMA_SLAVE_BUSWIDTH_3_BYTES; else if (bits <= 32) buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; else diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index b653ab001fba..8cd2f930ad0b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -543,6 +543,9 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, if (params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST) return -EINVAL; + if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12) && + params->tstamp_type > SNDRV_PCM_TSTAMP_TYPE_LAST) + return -EINVAL; if (params->avail_min == 0) return -EINVAL; if (params->silence_size >= runtime->boundary) { @@ -557,6 +560,8 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, err = 0; snd_pcm_stream_lock_irq(substream); runtime->tstamp_mode = params->tstamp_mode; + if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12)) + runtime->tstamp_type = params->tstamp_type; runtime->period_step = params->period_step; runtime->control->avail_min = params->avail_min; runtime->start_threshold = params->start_threshold; @@ -2540,9 +2545,7 @@ static int snd_pcm_tstamp(struct snd_pcm_substream *substream, int __user *_arg) return -EFAULT; if (arg < 0 || arg > SNDRV_PCM_TSTAMP_TYPE_LAST) return -EINVAL; - runtime->tstamp_type = SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY; - if (arg == SNDRV_PCM_TSTAMP_TYPE_MONOTONIC) - runtime->tstamp_type = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC; + runtime->tstamp_type = arg; return 0; } diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index 1e206de0c2dd..ba8e4a64e13e 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -101,9 +101,9 @@ int snd_seq_dump_var_event(const struct snd_seq_event *event, len -= size; } return 0; - } if (! (event->data.ext.len & SNDRV_SEQ_EXT_CHAINED)) { - return func(private_data, event->data.ext.ptr, len); } + if (!(event->data.ext.len & SNDRV_SEQ_EXT_CHAINED)) + return func(private_data, event->data.ext.ptr, len); cell = (struct snd_seq_event_cell *)event->data.ext.ptr; for (; len > 0 && cell; cell = cell->next) { |