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| author |   Linus Torvalds <torvalds@linux-foundation.org> | 2011-10-28 14:25:01 -0700 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2011-10-28 14:25:01 -0700 |
| commit | 68d99b2c8efcb6ed3807a55569300c53b5f88be5 (patch) | |
| tree | f189c8f2132d3668a2f0e503f5c3f8695b26a1c8 /sound/core/pcm_lib.c | |
| parent | Merge branch 'next-rebase' of git://git.kernel.org/pub/scm/linux/kernel/git/jbarnes/pci (diff) | |
| parent | Merge branch 'topic/hda' into for-linus (diff) | |
| download | linux-dev-68d99b2c8efcb6ed3807a55569300c53b5f88be5.tar.xz linux-dev-68d99b2c8efcb6ed3807a55569300c53b5f88be5.zip | |
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (549 commits)
ALSA: hda - Fix ADC input-amp handling for Cx20549 codec
ALSA: hda - Keep EAPD turned on for old Conexant chips
ALSA: hda/realtek - Fix missing volume controls with ALC260
ASoC: wm8940: Properly set codec->dapm.bias_level
ALSA: hda - Fix pin-config for ASUS W90V
ALSA: hda - Fix surround/CLFE headphone and speaker pins order
ALSA: hda - Fix typo
ALSA: Update the sound git tree URL
ALSA: HDA: Add new revision for ALC662
ASoC: max98095: Convert codec->hw_write to snd_soc_write
ASoC: keep pointer to resource so it can be freed
ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls
ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2
ASoC: da7210: Add support for line out and DAC
ASoC: da7210: Add support for DAPM
ALSA: hda/realtek - Fix DAC assignments of multiple speakers
ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value
ASoC: Set sgtl5000->ldo in ldo_regulator_register
ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture
ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture
...
Diffstat (limited to 'sound/core/pcm_lib.c')
| -rw-r--r-- | sound/core/pcm_lib.c | 26 |
1 files changed, 26 insertions, 0 deletions
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 62e90b862a0d..95d1e789715f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1399,6 +1399,32 @@ int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime, EXPORT_SYMBOL(snd_pcm_hw_constraint_pow2); +static int snd_pcm_hw_rule_noresample_func(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + unsigned int base_rate = (unsigned int)(uintptr_t)rule->private; + struct snd_interval *rate; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + return snd_interval_list(rate, 1, &base_rate, 0); +} + +/** + * snd_pcm_hw_rule_noresample - add a rule to allow disabling hw resampling + * @runtime: PCM runtime instance + * @base_rate: the rate at which the hardware does not resample + */ +int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime, + unsigned int base_rate) +{ + return snd_pcm_hw_rule_add(runtime, SNDRV_PCM_HW_PARAMS_NORESAMPLE, + SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_rule_noresample_func, + (void *)(uintptr_t)base_rate, + SNDRV_PCM_HW_PARAM_RATE, -1); +} +EXPORT_SYMBOL(snd_pcm_hw_rule_noresample); + static void _snd_pcm_hw_param_any(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { |