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- Re: [PATCH v2 00/11] ASoC: fsl-ssi: ac97-slave support
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- Re: M-Audio Delta 1010LT: S/PDIF clock source not working
- From: Alan Horstmann <gineera@xxxxxxxxxxxxxxx>
- Re: [PATCH v2 00/11] ASoC: fsl-ssi: ac97-slave support
- From: Markus Pargmann <mpa@xxxxxxxxxxxxxx>
- Re: [PATCH v2 00/11] ASoC: fsl-ssi: ac97-slave support
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- [PATCH v2 05/11] ASoC: fsl-ssi: Add support for imx-pcm-fiq
- From: Markus Pargmann <mpa@xxxxxxxxxxxxxx>
- [PATCH v2 01/11] ASoC: phycore-ac97: Add DT support
- From: Markus Pargmann <mpa@xxxxxxxxxxxxxx>
- [PATCH v2 08/11] ASoC: fsl-ssi: imx ac97 support
- From: Markus Pargmann <mpa@xxxxxxxxxxxxxx>
- [PATCH v2 00/11] ASoC: fsl-ssi: ac97-slave support
- From: Markus Pargmann <mpa@xxxxxxxxxxxxxx>
- [PATCH v2 03/11] ASoC: imx-pcm-fiq: Introduce pcm-fiq-params
- From: Markus Pargmann <mpa@xxxxxxxxxxxxxx>
- [PATCH v2 06/11] ASoC: fsl-ssi: Setup generic imx dma params
- From: Markus Pargmann <mpa@xxxxxxxxxxxxxx>
- [PATCH v2 11/11] ASoC: fsl: Update fsl-ssi binding doc
- From: Markus Pargmann <mpa@xxxxxxxxxxxxxx>
- [PATCH v2 09/11] ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97
- From: Markus Pargmann <mpa@xxxxxxxxxxxxxx>
- [PATCH v2 04/11] ASoC: fsl-ssi: Add SACNT definitions
- From: Markus Pargmann <mpa@xxxxxxxxxxxxxx>
- [PATCH v2 07/11] ARM: imx: Export ac97 reset functions
- From: Markus Pargmann <mpa@xxxxxxxxxxxxxx>
- [PATCH v2 10/11] ASoC: fsl: Move fsl-ssi binding doc to sound/
- From: Markus Pargmann <mpa@xxxxxxxxxxxxxx>
- [PATCH v2 02/11] ASoC: imx-pcm-dma: Add support for DMA init by device node
- From: Markus Pargmann <mpa@xxxxxxxxxxxxxx>
- Re: [PATCH 3/8] ASoC: ux500_pcm: Use the same snd_pcm_hardware for playback and capture
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: M-Audio Delta 1010LT: S/PDIF clock source not working
- From: "Gabriel M. Beddingfield" <gabrbedd@xxxxxxxxx>
- Samsung S5PV210 I2S DMA
- From: Mike Dyer <mike.dyer@xxxxxxxxxxxxx>
- Re: [PATCH] ALSA: hda/realtek - Add a quirk for AC700 Chromebook.
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_*
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] ALSA: hda - Apply codec delay to wallclock.
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 1/2] ALSA: add DSD formats
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: Handle large period size end-of-stream situation
- From: Patrick Lai <plai@xxxxxxxxxxxxxx>
- [PATCH v6 2/2] ALSA: ASoC: davinci-mcasp: don't overwrite DIT settings
- From: yegorslists@xxxxxxxxxxxxxx
- Re: BUG: unable to handle kernel NULL pointer dereference at 0000000000000010, IP: [<ffffffff819865fb>] snd_info_get_line+0x3b/0xf0
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 6/8] ASoC: tegra: Use common DAI DMA data struct
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- Re: [PATCH 6/8] ASoC: tegra: Use common DAI DMA data struct
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH v5 1/2] ALSA: ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode
- From: Yegor Yefremov <yegorslists@xxxxxxxxxxxxxx>
- [PATCH v6 1/2] ALSA: ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode
- From: yegorslists@xxxxxxxxxxxxxx
- Re: Asynchronous audio USB chips: choppy playback since 3.8-rc7
- From: Michael Trimarchi <michael@xxxxxxxxxxxxxxxxxxxx>
- Re: Asynchronous audio USB chips: choppy playback since 3.8-rc7
- From: Michael Trimarchi <michael@xxxxxxxxxxxxxxxxxxxx>
- [PATCH 2/2] ALSA: snd-usb: add quirks handler for DSD streams
- From: Daniel Mack <zonque@xxxxxxxxx>
- Re: [PATCH 1/2] ALSA: add DSD formats
- From: Daniel Mack <zonque@xxxxxxxxx>
- Re: Deadlock over semaphore issue with aplay while using dmix
- From: Jaroslav Kysela <perex@xxxxxxxx>
- Re: [PATCH] salsa-lib: Add seq_midi_event.h
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 3/4] ALSA: CA0132: Add latency monitoring
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH v6 1/2] ALSA: ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Audio capture on embedded system (via USB port)
- From: Israel Alejandro Arriaga Trejo <israel.alejandro.arriaga.trejo@xxxxxxxxxxx>
- Re: [PATCH 1/8] ASoC: ep93xx_pcm: Fix compile error
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 2/2] alsa-lib: bring pcm.h and pcm.c in sync with the kernel list
- From: Daniel Mack <zonque@xxxxxxxxx>
- Asynchronous audio USB chips: choppy playback since 3.8-rc7
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- Re: [PATCH] salsa-lib: Fix error on jack compilation with salsa
- From: Mihail Zenkov <mihail.zenkov@xxxxxxxxx>
- Re: [PATCH] ALSA: hda/ca0132 - Delay HP amp turnon.
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 2/2] alsa-lib: bring pcm.h and pcm.c in sync with the kernel list
- From: Daniel Mack <zonque@xxxxxxxxx>
- Re: [PATCH 7/8] ASoC: fsl: Use common DAI DMA data struct
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] sound: soc: tegra: do not provide PAUSE/RESUME in hardware info
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- Re: [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
- From: "Lin, Mengdong" <mengdong.lin@xxxxxxxxx>
- Re: [PATCH] ALSA: hda/cirrus - Add a quirk for Stumpy ChromeBox.
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH v2] ALSA: hda/ca0132 - Update latency based on DSP state.
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 2/2] alsa-lib: bring pcm.h and pcm.c in sync with the kernel list
- From: Jussi Laako <jussi@xxxxxxxxxxxxx>
- Re: [PATCH 2/8] ASoC: tegra: Don't claim to support PCM pause and resume
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH v2 1/2] ALSA: hda - bug fix on return value when getting HDMI ELD info
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- Re: [PATCH 2/2] alsa-lib: bring pcm.h and pcm.c in sync with the kernel list
- From: Jussi Laako <jussi@xxxxxxxxxxxxx>
- Re: [PATCH] pcm: fix typo, should be SND_PCM_STATE_PREPARED.
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH v2] ALSA: hda/ca0132 - Update latency based on DSP state.
- From: Pierre-Louis Bossart <pierre-louis.bossart@xxxxxxxxxxxxxxx>
- Re: [PATCH v2 2/2] ALSA: hda - bug fix on HDMI ELD debug message
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- ASOC snd_soc_dai_ops for clock mux control
- From: "Gellner, Christoph (ADITG/SW1)" <cgellner@xxxxxxxxxxxxxx>
- Re: [PATCH 2/3] ALSA: compress_core: Split copy into seperate read and write callbacks
- From: Pierre-Louis Bossart <pierre-louis.bossart@xxxxxxxxxxxxxxx>
- Re: [PATCH 08/10] ALSA: usb-audio: UAC2: auto clock selection module param
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH v2] ALSA: hda/ca0132 - Update latency based on DSP state.
- From: Pierre-Louis Bossart <pierre-louis.bossart@xxxxxxxxxxxxxxx>
- [PATCH] ALSA: hda - Apply codec delay to wallclock.
- From: Dylan Reid <dgreid@xxxxxxxxxxxx>
- Re: M-Audio Delta 1010LT: S/PDIF clock source not working
- From: Jonas Petersen <jnsptrsn1@xxxxxxxxx>
- Re: [PATCH 07/10] ALSA: usb-audio: UAC2: try to find and switch to valid clock
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH 6/8] ASoC: tegra: Use common DAI DMA data struct
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- [PATCH 3/8] ASoC: ux500_pcm: Use the same snd_pcm_hardware for playback and capture
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- [PATCH] ALSA: hda - fix typo in proc output
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- Re: [PATCH v2 2/2] ALSA: hda - bug fix on HDMI ELD debug message
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: ASoC: BeagleBoard driver development (PCM3168)
- From: Wendelin Klimann <wklimann@xxxxxxxxx>
- [PATCH] ALSA: hda/cirrus - Add a quirk for Stumpy ChromeBox.
- From: Dylan Reid <dgreid@xxxxxxxxxxxx>
- Re: Asynchronous audio USB chips: choppy playback since 3.8-rc7
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH 5/8] ASoC: dmaengine-pcm: Add a common DAI DMA data struct
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- Re: [PATCH 4/8] ASoC: dmaengine_pcm: Setup device_fc in snd_hwparams_to_dma_slave_config
- From: Nicolas Ferre <nicolas.ferre@xxxxxxxxx>
- Re: [PATCH v5 1/2] ALSA: ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Handle large period size end-of-stream situation
- From: Patrick Lai <plai@xxxxxxxxxxxxxx>
- Re: Asynchronous audio USB chips: choppy playback since 3.8-rc7
- From: Michael Trimarchi <michael@xxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 2/2] alsa-lib: bring pcm.h and pcm.c in sync with the kernel list
- From: Jussi Laako <jussi@xxxxxxxxxxxxx>
- Re: [PATCH 3/4] ALSA: CA0132: Add latency monitoring
- From: Dylan Reid <dgreid@xxxxxxxxxxxx>
- [PATCH 7/8] ASoC: fsl: Use common DAI DMA data struct
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- Re: [PATCH 6/8] ASoC: tegra: Use common DAI DMA data struct
- From: Stephen Warren <swarren@xxxxxxxxxxxxx>
- Re: [PATCH 08/10] ALSA: usb-audio: UAC2: auto clock selection module param
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- Re: [PATCH v2 1/2] ALSA: hda - bug fix on return value when getting HDMI ELD info
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH v2 2/2] alsa-lib: bring pcm.h and pcm.c in sync with the kernel list
- From: Daniel Mack <zonque@xxxxxxxxx>
- Re: Asynchronous audio USB chips: choppy playback since 3.8-rc7
- From: Daniel Mack <zonque@xxxxxxxxx>
- Re: [PATCH 1/3] ALSA: compress_core: Update calc_avail to use cumulative values
- From: Pierre-Louis Bossart <pierre-louis.bossart@xxxxxxxxxxxxxxx>
- Re: Asynchronous audio USB chips: choppy playback since 3.8-rc7
- From: Daniel Mack <zonque@xxxxxxxxx>
- Re: [PATCH v5 2/2] ALSA: ASoC: davinci-mcasp: don't overwrite DIT settings
- From: Yegor Yefremov <yegorslists@xxxxxxxxxxxxxx>
- [PATCH 2/8] ASoC: tegra: Don't claim to support PCM pause and resume
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- [PATCHv2 10/10] ALSA: usb-audio: UAC2: support read-only freq control
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- [PATCH 1/8] ASoC: ep93xx_pcm: Fix compile error
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- Re: [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 2/3] ALSA: compress_core: Split copy into seperate read and write callbacks
- From: Charles Keepax <ckeepax@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH v5 1/2] ALSA: ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode
- From: Yegor Yefremov <yegorslists@xxxxxxxxxxxxxx>
- Re: [PATCH v5] ALSA: usb: Work around CM6631 sample rate change bug
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH 0/8] More dmaengine PCM cleanups
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- [PATCH 3/3] ASoC: soc-compress: Split copy into seperate read and write callbacks
- From: Charles Keepax <ckeepax@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 07/10] ALSA: usb-audio: UAC2: try to find and switch to valid clock
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- [PATCH v2 1/2] alsa-lib: pcm.c: fix indentation
- From: Daniel Mack <zonque@xxxxxxxxx>
- [PATCH 2/2] ALSA: snd-usb: add quirks handler for DSD streams
- From: Daniel Mack <zonque@xxxxxxxxx>
- Re: [PATCH 7/8] ASoC: fsl: Use common DAI DMA data struct
- From: Shawn Guo <shawn.guo@xxxxxxxxxx>
- Re: [PATCH v2] ALSA: hda/ca0132 - Update latency based on DSP state.
- From: Dylan Reid <dgreid@xxxxxxxxxxxx>
- Re: [PATCH v2] ALSA: hda/ca0132 - Update latency based on DSP state.
- From: Dylan Reid <dgreid@xxxxxxxxxxxx>
- Re: [PATCH 1/3] ALSA: compress_core: Update calc_avail to use cumulative values
- From: Charles Keepax <ckeepax@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] salsa-lib: Fix error on jack compilation with salsa
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 2/2] ALSA: snd-usb: add quirks handler for DSD streams
- From: Daniel Mack <zonque@xxxxxxxxx>
- Re: ASoC updates for v3.9
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH 1/2] ALSA: add DSD formats
- From: Daniel Mack <zonque@xxxxxxxxx>
- [PATCH] Enabling Realtek ALC 671 codec
- From: Rainer Koenig <Rainer.Koenig@xxxxxxxxxxxxxx>
- Re: [PATCH v5] ALSA: usb: Work around CM6631 sample rate change bug
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: [PATCH 2/2] alsa-lib: bring pcm.h and pcm.c in sync with the kernel list
- From: Daniel Mack <zonque@xxxxxxxxx>
- Re: Deadlock over semaphore issue with aplay while using dmix
- From: mateen <abdulmateen.shaikh@xxxxxxxxx>
- Re: [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- Re: [PATCH v2] ALSA: hda/ca0132 - Update latency based on DSP state.
- From: Dylan Reid <dgreid@xxxxxxxxxxxx>
- Re: [PATCH 1/2] ALSA: hda - Add fixed codec delay to runtime delay.
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 8/8] ASoC: omap: Use common DAI DMA data
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: Deadlock over semaphore issue with aplay while using dmix
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCHv2 00/10] UAC2: Automatic clock switching
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- Re: [PATCH] salsa-lib: Add seq_midi_event.h
- From: Mihail Zenkov <mihail.zenkov@xxxxxxxxx>
- Re: [PATCH] salsa-lib: Add seq_midi_event.h
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] Enable Realtek ALC671 Codec
- From: Takashi Iwai <tiwai@xxxxxxx>
- Deadlock over semaphore issue with aplay while using dmix
- From: mateen <abdulmateen.shaikh@xxxxxxxxx>
- [PATCHv2 01/10] ALSA: usb-audio: convert list_for_each to entry variant
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- Re: Deadlock over semaphore issue with aplay while using dmix
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: Handle large period size end-of-stream situation
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: ASOC snd_soc_dai_ops for clock mux control
- From: "Gabriel M. Beddingfield" <gabrbedd@xxxxxxxxx>
- Re: [PATCH 5/8] ASoC: dmaengine-pcm: Add a common DAI DMA data struct
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxx>
- Re: [PATCH 4/8] ASoC: dmaengine_pcm: Setup device_fc in snd_hwparams_to_dma_slave_config
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxx>
- Re: [PATCH 5/8] ASoC: dmaengine-pcm: Add a common DAI DMA data struct
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: Deadlock over semaphore issue with aplay while using dmix
- From: Jaroslav Kysela <perex@xxxxxxxx>
- Re: [PATCH 8/8] ASoC: omap: Use common DAI DMA data
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxx>
- Re: How to create and verify verb tables for coreboot?
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 4/8] ASoC: dmaengine_pcm: Setup device_fc in snd_hwparams_to_dma_slave_config
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH v2] ALSA: hda/ca0132 - Update latency based on DSP state.
- From: Pierre-Louis Bossart <pierre-louis.bossart@xxxxxxxxxxxxxxx>
- [PATCH 2/3] ALSA: compress_core: Split copy into seperate read and write callbacks
- From: Charles Keepax <ckeepax@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] ALSA: hda - fix typo in proc output
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 1/2] ALSA: hda - Add fixed codec delay to runtime delay.
- From: Dylan Reid <dgreid@xxxxxxxxxxxx>
- Re: [PATCH] Enable Realtek ALC671 Codec
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- Re: [PATCH 1/2] ALSA: add DSD formats
- From: Jussi Laako <jussi@xxxxxxxxxxxxx>
- [PATCH] ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_*
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- Re: [PATCH 3/4] ALSA: CA0132: Add latency monitoring
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: Handle large period size end-of-stream situation
- From: "Gabriel M. Beddingfield" <gabrbedd@xxxxxxxxx>
- Re: [PATCH] sound: oss: sb_common: Used kmemdup instead of kmalloc and memcpy
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH 1/3] ALSA: compress_core: Update calc_avail to use cumulative values
- From: Charles Keepax <ckeepax@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 1/2] ALSA: hda - Add fixed codec delay to runtime delay.
- From: Takashi Iwai <tiwai@xxxxxxx>
- Can DAPM paths span multiple codecs/drivers?
- From: Mike Looijmans <mike.looijmans@xxxxxxxx>
- Re: [PATCH 1/3] ALSA: compress_core: Update calc_avail to use cumulative values
- From: Charles Keepax <ckeepax@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 1/2] ASoC: Samsung: return error if drvdata is not set
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- M-Audio Delta 1010LT: S/PDIF clock source not working
- From: Jonas Petersen <jnsptrsn1@xxxxxxxxx>
- Re: [PATCH 1/2] ALSA: add DSD formats
- From: Jussi Laako <jussi@xxxxxxxxxxxxx>
- Re: [PATCH 1/2] ALSA: add DSD formats
- From: Jussi Laako <jussi@xxxxxxxxxxxxx>
- Re: [PATCH 05/10] ALSA: usb-audio: use endianness macros
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- [PATCH] ALSA: hda/realtek - Add a quirk for AC700 Chromebook.
- From: Dylan Reid <dgreid@xxxxxxxxxxxx>
- PROBLEM: very loud sound of electric discharge on shutdown and restart with ALC882M with all kernels I tried up to 3.9 rc5
- From: Jenya Y <jy.gerstmaier@xxxxxxxxx>
- Re: [PATCH v2] ALSA: hda/ca0132 - Update latency based on DSP state.
- From: Pierre-Louis Bossart <pierre-louis.bossart@xxxxxxxxxxxxxxx>
- Re: [PATCHv2 00/10] UAC2: Automatic clock switching
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] Enable Realtek ALC671 Codec
- From: Rainer Koenig <Rainer.Koenig@xxxxxxxxxxxxxx>
- Re: Deadlock over semaphore issue with aplay while using dmix
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 2/8] ASoC: tegra: Don't claim to support PCM pause and resume
- From: Stephen Warren <swarren@xxxxxxxxxxxxx>
- Re: [PATCH] Enabling Realtek ALC 671 codec
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: How to use implicit feedback with full duplex?
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- [PATCH 8/8] ASoC: omap: Use common DAI DMA data
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- [PATCH v2] ALSA: hda/ca0132 - Update latency based on DSP state.
- From: Dylan Reid <dgreid@xxxxxxxxxxxx>
- Re: Deadlock over semaphore issue with aplay while using dmix
- From: Jaroslav Kysela <perex@xxxxxxxx>
- [PATCH 4/8] ASoC: dmaengine_pcm: Setup device_fc in snd_hwparams_to_dma_slave_config
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- Re: [PATCH] sound: oss: uart401: Used kmemdup instead of kmalloc and memcpy
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: Handle large period size end-of-stream situation
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
- From: "Lin, Mengdong" <mengdong.lin@xxxxxxxxx>
- Re: [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
- From: "Lin, Mengdong" <mengdong.lin@xxxxxxxxx>
- Re: [PATCH] USB-audio: Add "HP Digital Stereo Headset" to SPDIF blacklist
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 8/8] ASoC: omap: Use common DAI DMA data
- From: Jarkko Nikula <jarkko.nikula@xxxxxxxxxx>
- Re: [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 1/2] dmaengine: add const for name parameter
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 1/2] ALSA: hda - Add fixed codec delay to runtime delay.
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: Asynchronous audio USB chips: choppy playback since 3.8-rc7
- From: Daniel Mack <zonque@xxxxxxxxx>
- [PATCH] Enable Realtek ALC671 Codec
- From: Rainer Koenig <Rainer.Koenig@xxxxxxxxxxxxxx>
- Re: Deadlock over semaphore issue with aplay while using dmix
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: Handle large period size end-of-stream situation
- From: "Gabriel M. Beddingfield" <gabrbedd@xxxxxxxxx>
- Re: ASoC: BeagleBoard driver development (PCM3168)
- From: Daniel Mack <zonque@xxxxxxxxx>
- Re: [PATCH 2/2] alsa-lib: bring pcm.h and pcm.c in sync with the kernel list
- From: Jussi Laako <jussi@xxxxxxxxxxxxx>
- Re: [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH v5] ALSA: usb: Work around CM6631 sample rate change bug
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH v5] ALSA: usb: Work around CM6631 sample rate change bug
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- Re: Deadlock over semaphore issue with aplay while using dmix
- From: mateen <abdulmateen.shaikh@xxxxxxxxx>
- [PATCHv2 03/10] ALSA: usb-audio: neaten EXPORT_SYMBOLS placement
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- Re: Handle large period size end-of-stream situation
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- [PATCHv2 07/10] ALSA: usb-audio: UAC2: try to find and switch to valid clock
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- Re: [PATCH v2] ALSA: hda/ca0132 - Update latency based on DSP state.
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] salsa-lib: Add seq_midi_event.h
- From: Mihail Zenkov <mihail.zenkov@xxxxxxxxx>
- [PATCHv2 08/10] ALSA: usb-audio: UAC2: auto clock selection module param
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- Re: [PATCH v5 1/2] ALSA: ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCHv2 04/10] ALSA: usb-audio: spelling correction
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- [PATCH 1/2] ALSA: hda - Add fixed codec delay to runtime delay.
- From: Dylan Reid <dgreid@xxxxxxxxxxxx>
- [PATCH 2/2] ALSA: hda/ca0132 - Update latency based on DSP state.
- From: Dylan Reid <dgreid@xxxxxxxxxxxx>
- [PATCHv2 05/10] ALSA: usb-audio: use endianness macros
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- Re: [PATCH] ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_*
- From: Daniel Mack <zonque@xxxxxxxxx>
- Re: [PATCH 1/3] ALSA: compress_core: Update calc_avail to use cumulative values
- From: Pierre-Louis Bossart <pierre-louis.bossart@xxxxxxxxxxxxxxx>
- [PATCHv2 06/10] ALSA: usb-audio: UAC2: do clock validity check earlier
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- Re: [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCHv2 02/10] ALSA: usb-audio: neaten MODULE_DEVICE_TABLE placement
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- Re: [PATCH] Enable Realtek ALC671 Codec
- From: Rainer Koenig <Rainer.Koenig@xxxxxxxxxxxxxx>
- [PATCHv2 09/10] ALSA: usb-audio: show err in set_sample_rate_v2 debug
- From: Eldad Zack <eldad@xxxxxxxxxxxxxxx>
- got a segmentation fault when use multi-channel device
- From: "Zhang wei" <zhangwei@xxxxxxxxxxx>
- Re: [PATCH 1/2] dmaengine: add const for name parameter
- From: Vinod Koul <vinod.koul@xxxxxxxxx>
- Re: Audio capture on embedded system (via USB port)
- From: Michael Trimarchi <michael@xxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 06/10] ALSA: usb-audio: UAC2: do clock validity check earlier
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] Enable Realtek ALC671 Codec
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] salsa-lib: Fix error on jack compilation with salsa
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH] pcm: fix typo, should be SND_PCM_STATE_PREPARED.
- From: dsvensson@xxxxxxxxx
- [PATCH 52/53] ASoC: switch over to use snd_soc_register_component() on ux500 msp
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 24/54 v2] ASoC: switch over to use snd_soc_register_component() on sst
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH 2/2 resend] ASoC: cs4271: preserve "Master Playback Switch" setting
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH 27/54 v2] ASoC: switch over to use snd_soc_register_component() on omap mcbsp
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 1/2] ALSA: snd-usb: mixer: propagate errors up the call chain
- From: Daniel Mack <zonque@xxxxxxxxx>
- [PATCH 51/53] ASoC: switch over to use snd_soc_register_component() on txx9aclc ac97
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 2/2] ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()
- From: Daniel Mack <zonque@xxxxxxxxx>
- [PATCH 53/53] ASoC: snd_soc_[un]register_dai[s]() become non global function
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 50/53] ASoC: switch over to use snd_soc_register_component() on tegra20 i2s
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 25/54 v2] ASoC: switch over to use snd_soc_register_component() on mxs saif
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 04/54 v2] ASoC: switch over to use snd_soc_register_component() on au1x i2sc
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH v5] ASoC: soc-compress: Cancel delayed power down if needed
- From: Charles Keepax <ckeepax@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH 23/54 v2] ASoC: switch over to use snd_soc_register_component() on kirkwood i2s
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 22/54 v2] ASoC: switch over to use snd_soc_register_component() on jz4740 i2s
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH 50/54 v2] ASoC: switch over to use snd_soc_register_component() on tegra20 ac97
- From: Stephen Warren <swarren@xxxxxxxxxxxxx>
- Re: [PATCH 0/2] ASoC: dmaengine_pcm: support generic DMA binding users
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH] ALSA: hda - Introduce "Headset Mic" name
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- [PATCH 26/54 v2] ASoC: switch over to use snd_soc_register_component() on nuc900 ac97
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
- From: Daniel Mack <zonque@xxxxxxxxx>
- [PATCH 3/3] ALSA: snd-usb: add delay quirk for "Playback Design" products
- From: Daniel Mack <zonque@xxxxxxxxx>
- Re: [PATCH 1/2] ASoC: wm8903: Add the DAC boost control
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] ALSA: usb: Parse UAC2 extension unit like for UAC1
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH 15/54 v2] ASoC: switch over to use snd_soc_register_component() on davinci vcif
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 48/53] ASoC: switch over to use snd_soc_register_component() on tegra20 spdif
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: channel swapping issue on OMAP3/TWL4030 is back
- From: Peter Meerwald <pmeerw@xxxxxxxxxx>
- [PATCH 1/3] ALSA: snd-usb: handle the bmFormats field as unsigned int
- From: Daniel Mack <zonque@xxxxxxxxx>
- [PATCH 2/2] ASoC: tegra: add Tegra114 support to tegra_asoc_utils.c
- From: Stephen Warren <swarren@xxxxxxxxxxxxx>
- [PATCH 46/53] ASoC: switch over to use snd_soc_register_component() on spear spdif in
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 41/53] ASoC: switch over to use snd_soc_register_component() on goni_wm8994
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 2/3] ALSA: snd-usb: handle raw data format of UAC2 devices
- From: Daniel Mack <zonque@xxxxxxxxx>
- Re: [PATCH 0/2] ASoC: dmaengine_pcm: support generic DMA binding users
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- how to combine mutil-channel in playback or capture?
- From: "Zhang wei" <zhangwei@xxxxxxxxxxx>
- [PATCH 49/53] ASoC: switch over to use snd_soc_register_component() on tegra20 ac97
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 44/53] ASoC: switch over to use snd_soc_register_component() on sh4 hac
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH 2/2 resend] ASoC: cs4271: preserve "Master Playback Switch" setting
- From: Daniel Mack <zonque@xxxxxxxxx>
- [PATCH 47/53] ASoC: switch over to use snd_soc_register_component() on tegra30 i2s
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH] sound: max98090: Remove executable bit
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] ALSA: hda - Introduce "Headset Mic" name
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH 13/54 v2] ASoC: switch over to use snd_soc_register_component() on ep93xx i2s
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH] sound: soc: fix invalid free of devm_ allocated data
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] sound: soc: fix invalid free of devm_ allocated data
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH 45/53] ASoC: switch over to use snd_soc_register_component() on spear spdif out
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH 1/2] ALSA: snd-usb: mixer: propagate errors up the call chain
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH 27/53] ASoC: switch over to use snd_soc_register_component() on omap dmic
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 14/54 v2] ASoC: switch over to use snd_soc_register_component() on davinci i2s
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 42/53] ASoC: switch over to use snd_soc_register_component() on sh4 ssi
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 43/53] ASoC: switch over to use snd_soc_register_component() on sh4 siu
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH] sound: max98090: Remove executable bit
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
- From: Daniel Mack <zonque@xxxxxxxxx>
- Re: [PATCH] SoC: dma-sh7760: Fix compile error
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] sound: max98090: Remove executable bit
- From: Joe Perches <joe@xxxxxxxxxxx>
- [PATCH 1/2] ASoC: tegra: add Tegra114 support to the AHUB driver
- From: Stephen Warren <swarren@xxxxxxxxxxxxx>
- Re: [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH 39/53] ASoC: switch over to use snd_soc_register_component() on samsung i2s
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 16/54 v2] ASoC: switch over to use snd_soc_register_component() on davinci mcasp
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
- From: "Gabriel M. Beddingfield" <gabrbedd@xxxxxxxxx>
- [PATCH 33/53] ASoC: switch over to use snd_soc_register_component() on mmp sspa
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: ASoC: atmel-pcm-dma: Leaking memory in atmel_pcm_hw_params
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- Re: [PATCH 0/2] ASoC: dmaengine_pcm: support generic DMA binding users
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- Re: ASoC: atmel-pcm-dma: Leaking memory in atmel_pcm_hw_params
- From: Bo Shen <voice.shen@xxxxxxxxx>
- [PATCH 10/54 v2] ASoC: switch over to use snd_soc_register_component() on bf5xx i2s
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 09/54 v2] ASoC: switch over to use snd_soc_register_component() on bf5xx ac97
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 08/54 v2] ASoC: switch over to use snd_soc_register_component() on bf6xx i2s
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 07/54] ASoC: switch over to use snd_soc_register_component() on au1x ac97c
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH 1/3] ASoC: omap-mcpdm: Collect link direction configuration under a struct
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH] ASoC: fsl: imx-pcm-fiq: Use 'unsigned int' for period
- From: Fabio Estevam <festevam@xxxxxxxxx>
- [PATCH 01/18] ASoC: switch over to use snd_soc_register_component() on s3c i2s
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH] ASoC: fsl: imx-pcm-fiq: Use 'unsigned int' for period
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [TINYCOMPRESS] Remove local copies of kernel headers?
- From: Vinod Koul <vinod.koul@xxxxxxxxxxxxxxx>
- [PATCH 05/54 v2] ASoC: switch over to use snd_soc_register_component() on au1x psc-ac97
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 06/54 v2] ASoC: switch over to use snd_soc_register_component() on au1x psc-i2s
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 11/54 v2] ASoC: switch over to use snd_soc_register_component() on bf5xx tdm
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 35/53] ASoC: switch over to use snd_soc_register_component() on s3c24xx i2s
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH 0/2] ASoC: dmaengine_pcm: support generic DMA binding users
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH 34/53] ASoC: switch over to use snd_soc_register_component() on s6000 i2s
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH 19/54 v2] ASoC: switch over to use snd_soc_register_component() on mpc5200 i2s
- From: Stephen Warren <swarren@xxxxxxxxxxxxx>
- Re: [PATCH 43/54 v2] ASoC: switch over to use snd_soc_register_component() on sh4 ssi
- From: Stephen Warren <swarren@xxxxxxxxxxxxx>
- [PATCH 30/53] ASoC: pxa2xx-ac97: move EXPORT_SYMBOL_GPL() next to definition
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 28/53] ASoC: switch over to use snd_soc_register_component() on omap hdmi
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH] sound: max98090: Remove executable bit
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH 31/53] ASoC: switch over to use snd_soc_register_component() on pxa2xx ac97
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH 0/2] ASoC: dmaengine_pcm: support generic DMA binding users
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- [PATCH 2/2 resend] ASoC: cs4271: preserve "Master Playback Switch" setting
- From: Daniel Mack <zonque@xxxxxxxxx>
- [PATCH 0/53] ASoC: switch over to use snd_soc_register_component()
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 1/2 resend] ASoC: cs4271: switch to mute_stream
- From: Daniel Mack <zonque@xxxxxxxxx>
- Re: [PATCH 19/54 v2] ASoC: switch over to use snd_soc_register_component() on mpc5200 i2s
- From: Stephen Warren <swarren@xxxxxxxxxxxxx>
- Re: [PATCH 0/54 v2] ASoC: switch over to use snd_soc_register_component()
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH] ALSA: hda - Introduce "Headset Mic" name
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 2/2] ASoC: dmaengine_pcm: add snd_dmaengine_generic_pcm_open()
- From: Sebastien Guiriec <s-guiriec@xxxxxx>
- Re: [PATCH 0/2] ASoC: dmaengine_pcm: support generic DMA binding users
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] ALSA: usb: Parse UAC2 extension unit like for UAC1
- From: Daniel Mack <zonque@xxxxxxxxx>
- Re: [PATCH 2/2 resend] ASoC: cs4271: preserve "Master Playback Switch" setting
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH 29/53] ASoC: switch over to use snd_soc_register_component() on pxa ssp
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH 45/54 v2] ASoC: switch over to use snd_soc_register_component() on sh4 hac
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH 0/2] ASoC: dmaengine_pcm: support generic DMA binding users
- From: Lars-Peter Clausen <lars@xxxxxxxxxx>
- [PATCH 23/53] ASoC: switch over to use snd_soc_register_component() on mxs saif
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH] Sound: sgtl5000 Allow codec clock frequency to be set.
- From: Timur Tabi <timur@xxxxxxxx>
- [PATCH 09/53] ASoC: switch over to use snd_soc_register_component() on bf5xx tdm
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 20/53] ASoC: switch over to use snd_soc_register_component() on jz4740 i2s
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 24/53] ASoC: switch over to use snd_soc_register_component() on nuc900 ac97
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH] ALSA: hda - Introduce "Headset Mic" name
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH 08/53] ASoC: switch over to use snd_soc_register_component() on bf5xx i2s
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH 2/2] ASoC: dmaengine_pcm: add snd_dmaengine_generic_pcm_open()
- From: Vinod Koul <vinod.koul@xxxxxxxxx>
- [PATCH 46/54 v2] ASoC: switch over to use snd_soc_register_component() on spear spdif out
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH] Sound: sgtl5000 Allow codec clock frequency to be set.
- From: Timur Tabi <timur@xxxxxxxx>
- [PATCH 26/53] ASoC: switch over to use snd_soc_register_component() on omap mcpdm
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 21/53] ASoC: switch over to use snd_soc_register_component() on kirkwood i2s
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 25/53] ASoC: switch over to use snd_soc_register_component() on omap mcbsp
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH 1/2 resend] ASoC: cs4271: switch to mute_stream
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH 22/53] ASoC: switch over to use snd_soc_register_component() on sst
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 19/53] ASoC: switch over to use snd_soc_register_component() on imx ssi
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH 44/54 v2] ASoC: switch over to use snd_soc_register_component() on sh4 siu
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: [PATCH 50/54 v2] ASoC: switch over to use snd_soc_register_component() on tegra20 ac97
- From: Stephen Warren <swarren@xxxxxxxxxxxxx>
- Re: [PATCH 0/2] ASoC: dmaengine_pcm: support generic DMA binding users
- From: Russell King - ARM Linux <linux@xxxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: Add optional pointer to machine audio routes to snd_soc_card
- From: Jarkko Nikula <jhnikula@xxxxxxxxx>
- Re: [PATCH] ASoC: Add optional pointer to machine audio routes to snd_soc_card
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>
- Re: [PATCH] ASoC: Add optional pointer to machine audio routes to snd_soc_card
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>
- Re: [PATCH] sound, core, pcm_lib: remember to always call va_end() on stuff that we va_start()
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCHes 00/10] ASoC: Samsung: Enable latest SoCs
- From: Jassi Brar <jassisinghbrar@xxxxxxxxx>
- [PATCH] ASoC: Remove some unused defines from WM8903
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH] alsa-python mixer: add poll_fds property
- From: Milan Nikolic <gen2brain@xxxxxxxxx>
- Re: [PATCH] ASoC: mcbsp: Add McBSP support for OMAP4
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH] ASoC: mcbsp: Add McBSP support for OMAP4
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- Re: [PATCH 1/4] ASoC: clean up cache read/write functions
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 2/4] ASoC: clean up spi cache read/write functions
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 1/4] ASoC: clean up cache read/write functions
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 1/4] ASoC: clean up cache read/write functions
- From: Dimitris Papastamos <dp@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 4/4] ASoC: clean up cache accesser
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH 4/4] ASoC: Factor out cache_ops implementations
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 4/4] ASoC: clean up cache accesser
- From: Dimitris Papastamos <dp@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 4/4] ASoC: clean up cache accesser
- From: Dimitris Papastamos <dp@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: ASoC cache code (looking toward 2.6.38 merge window)
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: ASoC cache code (looking toward 2.6.38 merge window)
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 4/4] ASoC: clean up cache accesser
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 4/4] ASoC: clean up cache accesser
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: ASoC cache code (looking toward 2.6.38 merge window)
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH 4/4] ASoC: clean up cache accesser
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] ASoC: twl6040: Convert HF and HS drivers to use DAPM OUT_DRV widget
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 4/4] ASoC: clean up cache accesser
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH] ASoC: twl6040: Convert HF and HS drivers to use DAPM OUT_DRV widget
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- Re: ASoC cache code (looking toward 2.6.38 merge window)
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 4/4] ASoC: clean up cache accesser
- From: Dimitris Papastamos <dp@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Channel configuration
- From: Irfan Shaikh <irfan.shaikh@xxxxxxxxxx>
- [PATCH 4/4] ASoC: clean up cache accesser
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH 3/4] ASoC: clean up i2c cache read / write functions
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH 2/4] ASoC: clean up spi cache read/write functions
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH 1/4] ASoC: clean up cache read/write functions
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: ASoC cache code (looking toward 2.6.38 merge window)
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: ASoC cache code (looking toward 2.6.38 merge window)
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- ASoC cache code (looking toward 2.6.38 merge window)
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH] ASoC: Fix WM8994/58 3D stereo control definitions
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 0/3] Re: Internal Mic vs Int Mic
- From: Takashi Iwai <tiwai@xxxxxxx>
- Not able to set my playback device to mono
- From: Irfan Shaikh <irfan.shaikh@xxxxxxxxxx>
- [PATCH 3/3] ALSA: HDA: Rename "e-Mic" and "i-Mic" to "Mic" and "Internal Mic"
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- [PATCH 2/3] ALSA: HDA: Rename "Ext Mic" and "External Mic" to "Mic"
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- [PATCH 1/3] ALSA: HDA: Rename "Int Mic" to "Internal Mic"
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- [PATCH 0/3] Re: Internal Mic vs Int Mic
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- Re: [PATCHes 00/10] ASoC: Samsung: Enable latest SoCs
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 05/10] ASoC: SMDK6450: Enable I2S device
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 10/10] ASoC: SMDKV310: Add I2S support
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: Internal Mic vs Int Mic
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: periods*period_size vs. buffer_size; should they be equal?
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] ASoC: WM8580: Debug interface index
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: Add optional pointer to machine audio routes to snd_soc_card
- From: Jarkko Nikula <jhnikula@xxxxxxxxx>
- Re: [PATCH] ASoC: Add optional pointer to machine audio routes to snd_soc_card
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [RFC/PATCH v7 12/12] v4l: Make v4l2_subdev inherit from media_entity
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- [RFC/PATCH v7 11/12] v4l: Make video_device inherit from media_entity
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- [RFC/PATCH v7 10/12] v4l: Add a media_device pointer to the v4l2_device structure
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- [RFC/PATCH v7 09/12] media: Pipelines and media streams
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- [RFC/PATCH v7 08/12] media: Links setup
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- [RFC/PATCH v7 07/12] media: Entities, pads and links enumeration
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- [RFC/PATCH v7 06/12] media: Media device information query
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- [RFC/PATCH v7 05/12] media: Reference count and power handling
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- [RFC/PATCH v7 04/12] media: Entity graph traversal
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- [RFC/PATCH v7 03/12] media: Entities, pads and links
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- [RFC/PATCH v7 02/12] media: Media device
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- [RFC/PATCH v7 01/12] media: Media device node support
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- [RFC/PATCH v7 00/12] Media controller (core and V4L2)
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- Re: [PATCH] ALSA: HDA: Add auto-mute for Thinkpad SL410/SL510
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] ASoC: Add optional pointer to machine audio routes to snd_soc_card
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>
- Re: [PATCHes 00/10] ASoC: Samsung: Enable latest SoCs
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: sh: fsi: modify improper dependent
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: Add optional pointer to machine audio routes to snd_soc_card
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Internal Mic vs Int Mic
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- [PATCH] ALSA: HDA: Add auto-mute for Thinkpad SL410/SL510
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- [PATCH 10/10] ASoC: SMDKV310: Add I2S support
- From: Jassi@xxxxxxxxxxxxxxxx, "Brar <jassisinghbrar"@gmail.com
- [PATCH 04/10] ASoC: SMDK6440: Enable I2S device
- From: Jassi@xxxxxxxxxxxxxxxx, "Brar <jassisinghbrar"@gmail.com
- [PATCH 03/10] ASoC: SMDK_WM8580: Make I2S0 as default dai
- From: Jassi@xxxxxxxxxxxxxxxx, "Brar <jassisinghbrar"@gmail.com
- [PATCH 05/10] ASoC: SMDK6450: Enable I2S device
- From: Jassi@xxxxxxxxxxxxxxxx, "Brar <jassisinghbrar"@gmail.com
- [PATCH 01/10] ASoC: Samsung: Set default rclk source rate
- From: Jassi@xxxxxxxxxxxxxxxx, "Brar <jassisinghbrar"@gmail.com
- [PATCH 07/10] ASoC: SMDKV210: Enable I2S device
- From: Jassi@xxxxxxxxxxxxxxxx, "Brar <jassisinghbrar"@gmail.com
- [PATCH 08/10] ASoC: SMDKC110: Enable I2S device
- From: Jassi@xxxxxxxxxxxxxxxx, "Brar <jassisinghbrar"@gmail.com
- [PATCH 06/10] ASoC: SMDK6442: Enable I2S device
- From: Jassi@xxxxxxxxxxxxxxxx, "Brar <jassisinghbrar"@gmail.com
- [PATCH 02/10] ASoC: Samsung: I2S: Flush FIFO after stop
- From: Jassi@xxxxxxxxxxxxxxxx, "Brar <jassisinghbrar"@gmail.com
- [PATCH 09/10] ASoC: SMDKV310: Enable AC97 device
- From: Jassi@xxxxxxxxxxxxxxxx, "Brar <jassisinghbrar"@gmail.com
- [PATCHes 00/10] ASoC: Samsung: Enable latest SoCs
- From: Jassi@xxxxxxxxxxxxxxxx, "Brar <jassisinghbrar"@gmail.com
- [PATCH] ASoC: sh: fsi: modify improper dependent
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- ALSA HDA-Intel on HP Mini 110-3100er
- From: Болгар Василий <reanimator-zon@xxxxxxx>
- Re: lenovo G560 internal microphone fix
- From: Raymond Yau <superquad.vortex2@xxxxxxxxx>
- Re: periods*period_size vs. buffer_size; should they be equal?
- From: Raymond Yau <superquad.vortex2@xxxxxxxxx>
- Re: periods*period_size vs. buffer_size; should they be equal?
- From: Jassi Brar <jassisinghbrar@xxxxxxxxx>
- Re: Handling non-buffer-sized MMAP requests
- From: Raymond Yau <superquad.vortex2@xxxxxxxxx>
- lenovo G560 internal microphone fix
- From: Kayne Richens <kayno@xxxxxxxxx>
- hda_intel driver and HDMI audio problems.
- From: James Courtier-Dutton <james.dutton@xxxxxxxxx>
- Re: [PATCH] ASoC: Add optional pointer to machine audio routes to snd_soc_card
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- periods*period_size vs. buffer_size; should they be equal?
- From: Stephen Warren <swarren@xxxxxxxxxx>
- Re: Handling non-buffer-sized MMAP requests
- From: louis@xxxxxxxxxxxxxxxx
- [PATCH] ASoC: Add optional pointer to machine audio routes to snd_soc_card
- From: Jarkko Nikula <jhnikula@xxxxxxxxx>
- Re: [PATCH] ASoC: soc-dapm: Introduce the new snd_soc_dapm_virt_mux type
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH] regulator: Clean up logging a bit
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [GIT PULL] ASoC updates for 2.6.38
- From: Takashi Iwai <tiwai@xxxxxxx>
- [GIT PULL] ASoC updates for 2.6.38
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH 2/2] ASoC: sh: fsi: Add over/under run counter
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 1/2] ASoC: sh: fsi: move fsi_irq_enable function to fsi_dai_trigger
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: sh: fsi: remove unnecessary dependent from Kconfig
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 2/2] ASoC: Do DAPM control updates in the middle of DAPM sequences
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: CONFIG_SND_HDA_INTEL regression on 2.6.37-rc3 - no audio
- From: Adam Williamson <awilliam@xxxxxxxxxx>
- Re: question on static dapm routes
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 1/2] ASoC: sh: fsi: move fsi_irq_enable function to fsi_dai_trigger
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: sh: fsi: remove unnecessary dependent from Kconfig
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: soc-dapm: Introduce the new snd_soc_dapm_virt_mux type
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: callbacks order while capturing from Line-IN
- From: Manu Abraham <abraham.manu@xxxxxxxxx>
- question on static dapm routes
- From: "Harsha, Priya" <priya.harsha@xxxxxxxxx>
- Re: callbacks order while capturing from Line-IN
- From: Jaroslav Kysela <perex@xxxxxxxx>
- Re: CONFIG_SND_HDA_INTEL regression on 2.6.37-rc3 - no audio
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: callbacks order while capturing from Line-IN
- From: Manu Abraham <abraham.manu@xxxxxxxxx>
- Re: [PATCH 2/2] ASoC: Do DAPM control updates in the middle of DAPM sequences
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>
- Re: callbacks order while capturing from Line-IN
- From: Jaroslav Kysela <perex@xxxxxxxx>
- Re: callbacks order while capturing from Line-IN
- From: Manu Abraham <abraham.manu@xxxxxxxxx>
- callbacks order while capturing from Line-IN
- From: Manu Abraham <abraham.manu@xxxxxxxxx>
- Re: [PATCH] ASoC: Davinci: Replace use of IO_ADDRESS with ioremap()
- From: "Bedia, Vaibhav" <vaibhav.bedia@xxxxxx>
- [PATCH 2/2] ASoC: sh: fsi: Add over/under run counter
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH 1/2] ASoC: sh: fsi: move fsi_irq_enable function to fsi_dai_trigger
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- [PATCH] ASoC: sh: fsi: remove unnecessary dependent from Kconfig
- From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
- Re: CONFIG_SND_HDA_INTEL regression on 2.6.37-rc3 - no audio
- From: Adam Williamson <awilliam@xxxxxxxxxx>
- Re: CONFIG_SND_HDA_INTEL regression on 2.6.37-rc3 - no audio
- From: Adam Williamson <awilliam@xxxxxxxxxx>
- Re: [PATCH] ALSA: hda - factorize an automute_mic realtek quirk function
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] ALSA: hda - factorize an automute_mic realtek quirk function
- From: Anisse Astier <anisse@xxxxxxxxx>
- Re: [PATCH] ALSA: hda - factorize an automute_mic realtek quirk function
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH] ASoC: soc-dapm: Introduce the new snd_soc_dapm_virt_mux type
- From: Dimitris Papastamos <dp@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] regulator: Convert WM835x to use get_voltage_sel()
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH 2/2] ASoC: Do DAPM control updates in the middle of DAPM sequences
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 2/2] ASoC: Do DAPM control updates in the middle of DAPM sequences
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>
- Re: [PATCH v4] ASoC: twl6040: Add ramp up/down volume for HS and HF
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH 2/2] ASoC: Do DAPM control updates in the middle of DAPM sequences
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>
- [PATCH] regulator: Convert WM835x to use get_voltage_sel()
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: Davinci: Replace use of IO_ADDRESS with ioremap()
- From: "Bedia, Vaibhav" <vaibhav.bedia@xxxxxx>
- [PATCH] ALSA: hda - factorize an automute_mic realtek quirk function
- From: Anisse Astier <anisse@xxxxxxxxx>
- [PATCH] ASoC: Davinci: Replace use of IO_ADDRESS with ioremap()
- From: Vaibhav Bedia <vaibhav.bedia@xxxxxx>
- Re: [PATCH 2/2] ASoC: Do DAPM control updates in the middle of DAPM sequences
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>
- Re: hda_codec_ops hotplug events
- From: Raymond Yau <superquad.vortex2@xxxxxxxxx>
- [PATCH v4] ASoC: twl6040: Add ramp up/down volume for HS and HF
- From: Margarita Olaya <magi.oc@xxxxxxxxx>
- Re: [PATCH v3 2/2] ASoC: twl6040: Add ramp up/down volume for HS and HF
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: mapping externally allocated Scatter Gather DMA buffers
- From: Manu Abraham <abraham.manu@xxxxxxxxx>
- Re: [PATCHv2.1 0/3] ASoC: Extend DAPM to handle cross-device paths
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 1/2] ASoC: Remove unused DAPM_DOUBLE control types
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- hda_codec_ops hotplug events
- From: James Courtier-Dutton <james.dutton@xxxxxxxxx>
- Re: [PATCH] regulator: Allow modular build of mc13xxx-core
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH] regulator: Allow modular build of mc13xxx-core
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH v3 1/2] ASoC: twl6040: Set default gains to minimun value
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 1/2] ASoC: Remove unused DAPM_DOUBLE control types
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: ALSA: HDA: Enable subwoofer on Asus G73Jw
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: ALSA: HDA: Enable subwoofer on Asus G73Jw
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- Re: ALSA: HDA: Enable subwoofer on Asus G73Jw
- From: Takashi Iwai <tiwai@xxxxxxx>
- ALSA: HDA: Enable subwoofer on Asus G73Jw
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- Re: [PATCH] ALSA: HDA: Fix auto-mute on Lenovo Edge 14
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH] ALSA: HDA: Fix auto-mute on Lenovo Edge 14
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- [PATCH v3 2/2] ASoC: twl6040: Add ramp up/down volume for HS and HF
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v3 1/2] ASoC: twl6040: Set default gains to minimun value
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- Re: [RFC/PATCH v6 03/12] media: Entities, pads and links
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- Re: [RFC/PATCH v6 03/12] media: Entities, pads and links
- From: Raymond Yau <superquad.vortex2@xxxxxxxxx>
- Re: [PATCH] pxa2xx-ac97: Use platform device IRQ resource.
- From: Eric Miao <eric.y.miao@xxxxxxxxx>
- Re: [PATCH v2 00/16] twl6040 updates
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH v4 0/3] Add Jack support for
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: TWL4030: Fix 24bit support
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: ASoC updates for 2.6.38
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: ASoC update for 2.6.37
- From: Takashi Iwai <tiwai@xxxxxxx>
- ASoC updates for 2.6.38
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- ASoC update for 2.6.37
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: soc-cache: A few minor stylistic changes
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH 2/2] ASoC: Do DAPM control updates in the middle of DAPM sequences
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH 1/2] ASoC: Remove unused DAPM_DOUBLE control types
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- XMOS Board Follow Up
- From: "George Ioakimedes" <GeorgeIoak@xxxxxxxxxxx>
- Re: Questions regarding ASoC
- From: "Bensaid, Selma" <selma.bensaid@xxxxxxxxx>
- Re: regarding setting of fragment size in usbaudio driver
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: Calculated average playback latency
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: [PATCH] ASoC: soc-cache: A few minor stylistic changes
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: Explicitly clear WM8993 ramp controls on power down
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCHv2.1 0/3] ASoC: Extend DAPM to handle cross-device paths
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [RFC/PATCH v6 03/12] media: Entities, pads and links
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- [PATCH] ASoC: soc-cache: A few minor stylistic changes
- From: Dimitris Papastamos <dp@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [RFC/PATCH v6 03/12] media: Entities, pads and links
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- Re: [RFC/PATCH v6 03/12] media: Entities, pads and links
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- Re: [PATCH v2] ASoC: dapm: Add output driver widget
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [RFC/PATCH v6 03/12] media: Entities, pads and links
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [RFC/PATCH v6 03/12] media: Entities, pads and links
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: [RFC/PATCH v6 03/12] media: Entities, pads and links
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: [RFC/PATCH v6 03/12] media: Entities, pads and links
- From: Laurent Pinchart <laurent.pinchart@xxxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: TWL4030: Fix 24bit support
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Calculated average playback latency
- From: Jean-Yves Avenard <jyavenard@xxxxxxxxx>
- Re: [PATCH] ASoC: TWL4030: Fix 24bit support
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>
- Re: [PATCH] ASoC: TWL4030: Fix 24bit support
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH] ASoC: TWL4030: Fix 24bit support
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>
- [PATCH] ASoC: Explicitly clear WM8993 ramp controls on power down
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCHv2.1 3/3] ASoC: Extend DAPM to handle power changes on cross-device paths
- From: Jarkko Nikula <jhnikula@xxxxxxxxx>
- [PATCHv2.1 2/3] ASoC: Move widgets from DAPM context to snd_soc_card
- From: Jarkko Nikula <jhnikula@xxxxxxxxx>
- [PATCHv2.1 1/3] ASoC: Move DAPM paths from DAPM context to snd_soc_card
- From: Jarkko Nikula <jhnikula@xxxxxxxxx>
- [PATCHv2.1 0/3] ASoC: Extend DAPM to handle cross-device paths
- From: Jarkko Nikula <jhnikula@xxxxxxxxx>
- Re: sound: usb-audio: full capture/playback/spdif support for Digidesign Mbox 2
- From: Damien Zammit <damien.zammit@xxxxxxxxx>
- Re: [PATCH 5/5] sound: ml403-ac97cr: Use vsprintf extension %pR for struct resource
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH] ASoC: Fix AC'97 registration unwind
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH] regulator: Update LDO2 for WM8958
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- [PATCH] ASoC: Fix AC'97 registration unwind
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [RFC 2.6.38] ASoC: Fix build error caused by merging a fix for 2.6.37 into 2.6.38
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 2.6.37] ASoC: Fix bias power down of non-DAPM codec
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [RFC/PATCH v6 03/12] media: Entities, pads and links
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: Joining the development
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- [PATCH] regulator: Update LDO2 for WM8958
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: Joining the development
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: regarding setting of fragment size in usbaudio driver
- From: Amit Nagal <helloin.amit@xxxxxxxxx>
- Re: regarding setting of fragment size in usbaudio driver
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: really no idea? Re: low volume with hda base card (see attached alsa-info output
- From: Halim Sahin <halim.sahin@xxxxxxxxxx>
- Re: [PATCH] ASoC: Fix widgets for WM8994/58 AIF2 source control
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [Alsa-user] regarding setting of fragment size in usbaudio driver
- From: Bill Unruh <unruh@xxxxxxxxxxxxxx>
- Re: regarding setting of fragment size in usbaudio driver
- From: Amit Nagal <helloin.amit@xxxxxxxxx>
- Re: 10 Sec sleep in wait_for_avail_min function
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: regarding setting of fragment size in usbaudio driver
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: Custom IOCTL call
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: Unable to set ALSA buffer size
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: Handling non-buffer-sized MMAP requests
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: Gigabyte 880GM-USB3 MIDI and Dolby 5.1/7.1 issues
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: Joining the development
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: "Unknown symbol" with alsa snapshot (tiwai)
- From: Raymond Yau <superquad.vortex2@xxxxxxxxx>
- Re: [PATCH 2/3] ASoC: CS4271 codec support
- From: Jamie Iles <jamie@xxxxxxxxxxxxx>
- Re: [PATCH 0/2] Fixed issues/defects reported by Coverity tool.
- From: Sudarshan Bisht <sudarshan.bisht@xxxxxxxxx>
- Re: Custom IOCTL call
- From: Eliot Blennerhassett <eblennerhassett@xxxxxxxxxxxxxxxx>
- Re: snd_hda_intel error
- From: Jack Byer <ftn768@xxxxxxxxx>
- [PATCH] ALSA: aoa: Remove wrong i2c_set_clientdata in onyx_i2c_remove()
- From: Axel Lin <axel.lin@xxxxxxxxx>
- Re: [PATCH] asoc, codec: Add SGTL5000 codec support to asoc
- From: Zeng Zhaoming <zhaoming.zeng@xxxxxxxxxxxxx>
- Re: [PATCH 2/3] ASoC: CS4271 codec support
- From: Jamie Iles <jamie@xxxxxxxxxxxxx>
- [PATCH] asoc, codec: Add SGTL5000 codec support to asoc
- From: <zhaoming.zeng@xxxxxxxxxxxxx>
- [PATCH] ASoC: Fix resource leak if soc_register_ac97_dai_link failed
- From: Axel Lin <axel.lin@xxxxxxxxx>
- [PATCH] ASoC: Fix snd_soc_instantiate_card error path
- From: Axel Lin <axel.lin@xxxxxxxxx>
- Re: Bug in alsa drivers reported when launching pulseaudio
- From: kieran magee <kiergbm@xxxxxxxxx>
- Re: [PATCH v2 04/16] ASoC: twl6040: Set default gains to minimun value
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH v2 03/16] ASoC: twl6040: Add headset and handset mux controls
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH v4 2/3] ASoC: twl6040: Add jack support for headset and handset
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH v2 02/16] ASoC: twl6040: Modify the IRQ handler
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH v4 2/3] ASoC: twl6040: Add jack support for headset and handset
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- regarding setting of fragment size in usbaudio driver
- From: Amit Nagal <helloin.amit@xxxxxxxxx>
- [PATCH v2] ASoC: dapm: Add output driver widget
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 16/16] ASoC: twl6040: Fix TLV dB step values for gains
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 14/16] ASoC: twl6040: Increase timeout for power up
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 15/16] ASoC: twl6040: Use correct offset for LineInAmp Right
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 13/16] ASoC: twl6040: Enable plug detection interrupts
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 11/16] ASoC: twl6040: Add ramp up/down volume for HS and HF
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 12/16] ASoC: twl6040: Clear interrupt status at boot time
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 10/16] ASoC: twl6040: Enable automatic power for phoenix 1.1
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 09/16] ASoC: twl6040: Fix analog Mic L & R mux controls
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 08/16] ASoC: twl6040: Support other sample rates.
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 07/16] ASoC: twl6040: Fix PCM error handling ops
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 06/16] ASoC: twl6040: Restore bias level at resume
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 04/16] ASoC: twl6040: Set default gains to minimun value
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 05/16] ASoC: twl6040: Support all DAIs
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 03/16] ASoC: twl6040: Add headset and handset mux controls
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 02/16] ASoC: twl6040: Modify the IRQ handler
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 01/16] ASoC: twl6040: Update twl IO macro
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v2 00/16] twl6040 updates
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v4 3/3] ASoC: sdp4430: Add Jack support
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v4 2/3] ASoC: twl6040: Add jack support for headset and handset
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCHv4 1/3] ASoC: sdp4430: Enable FM stereo pins
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v4 0/3] Add Jack support for
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- Re: [PATCH 3/3] ASoC: Automatically manage WM8903 deemphasis rate
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH 2/3] ASoC: Remove open coded symmetry implementation from WM8903
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH 1/3] ASoC: Implement WM8903 oversampling rate controls
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [RFC 2.6.38] ASoC: Fix build error caused by merging a fix for 2.6.37 into 2.6.38
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH 2.6.37] ASoC: Fix bias power down of non-DAPM codec
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: Implement WM8903 high pass filter support
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH v2] ASoC: tlv320dac33: Power down digital parts, when not needed
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: tabs versus space fill
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: tabs versus space fill
- From: Vasily Khoruzhick <anarsoul@xxxxxxxxx>
- tabs versus space fill
- From: Peter Hsiang <Peter.Hsiang@xxxxxxxxxxxx>
- Custom IOCTL call
- From: Radivoje Jovanovic <radivojejovanovic@xxxxxxxxx>
- [PATCH 3/3] ASoC: Automatically manage WM8903 deemphasis rate
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH 2/3] ASoC: Remove open coded symmetry implementation from WM8903
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH 1/3] ASoC: Implement WM8903 oversampling rate controls
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [RFC 2.6.38] ASoC: Fix build error caused by merging a fix for 2.6.37 into 2.6.38
- From: Jarkko Nikula <jhnikula@xxxxxxxxx>
- [PATCH 2.6.37] ASoC: Fix bias power down of non-DAPM codec
- From: Jarkko Nikula <jhnikula@xxxxxxxxx>
- [PATCH] ASoC: Implement WM8903 high pass filter support
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH v4] ASoC: Add HP iPAQ H1940 support
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [RFC 2.6.38] ASoC: Fix bias power down of non-DAPM codec
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- "Unknown symbol" with alsa snapshot (tiwai)
- From: Simos Xenitellis <simos.lists@xxxxxxxxxxxxxx>
- Re: [PATCH v4] ASoC: Add HP iPAQ H1940 support
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: Implement WM8994/58 DAC and ADC oversampling control
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: Optimise WM9081 FLL performance
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- [RFC 2.6.38] ASoC: Fix bias power down of non-DAPM codec
- From: Jarkko Nikula <jhnikula@xxxxxxxxx>
- Re: [PATCH 1/3] ASoC: EP93xx I2S and PCM fixes
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: tlv320dac33: Power down digital parts, when not needed
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH v2] ASoC: tlv320dac33: Power down digital parts, when not needed
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>
- Re: [PATCH] ASoC: tlv320dac33: Power down digital parts, when not needed
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>
- Re: [PATCH] ALSA: aoa: Remove wrong i2c_set_clientdata in onyx_i2c_remove()
- From: Takashi Iwai <tiwai@xxxxxxx>
- Unable to set ALSA buffer size
- From: Irfan Shaikh <irfan.shaikh@xxxxxxxxxx>
- Re: [PATCH] ASoC: tlv320dac33: Power down digital parts, when not needed
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH] ASoC: tlv320dac33: Power down digital parts, when not needed
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>
- Handling non-buffer-sized MMAP requests
- From: louis@xxxxxxxxxxxxxxxx
- Re: [PATCH 1/3] ASoC: EP93xx I2S and PCM fixes
- From: Alexander <subaparts@xxxxxxxxx>
- Re: Help wanted: emachines em350 internal mic
- From: Eliot Blennerhassett <eblennerhassett@xxxxxxxxxxxxxxxx>
- [PATCH v4] ASoC: Add HP iPAQ H1940 support
- From: Vasily Khoruzhick <anarsoul@xxxxxxxxx>
- Re: Gigabyte 880GM-USB3 MIDI and Dolby 5.1/7.1 issues
- From: Jonathan Gardner <jgardner@xxxxxxxxxxxxxxxxxxx>
- Re: ASoC updates for 2.6.37
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: ASoC updates for 2.6.38
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH] ASoC: Optimise WM9081 FLL performance
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- ASoC updates for 2.6.38
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- ASoC updates for 2.6.37
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: Gigabyte 880GM-USB3 MIDI and Dolby 5.1/7.1 issues
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: [PATCH 1/3] ASoC: EP93xx I2S and PCM fixes
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 1/3] ASoC: EP93xx I2S and PCM fixes
- From: Alexander <subaparts@xxxxxxxxx>
- Re: [PATCH] asoc, codec: Add SGTL5000 codec support to asoc
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH] ASoC: Implement WM8994/58 DAC and ADC oversampling control
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: Gigabyte 880GM-USB3 MIDI and Dolby 5.1/7.1 issues
- From: Jonathan Gardner <jgardner@xxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: WM8580: Fix R8 initial value
- From: Jassi Brar <jassisinghbrar@xxxxxxxxx>
- Re: Gigabyte 880GM-USB3 MIDI and Dolby 5.1/7.1 issues
- From: Clemens Ladisch <clemens@xxxxxxxxxx>
- Re: Gigabyte 880GM-USB3 MIDI and Dolby 5.1/7.1 issues
- From: Jonathan Gardner <jgardner@xxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: Remove unnecessary structure definitions
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Gigabyte 880GM-USB3 MIDI and Dolby 5.1/7.1 issues
- From: Jonathan Gardner <jgardner@xxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: EP93xx: sampling rate range extended
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: WM8580: Fix R8 initial value
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: fix deemphasis control in wm8904/55/60 codecs
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 1/3] ASoC: EP93xx I2S and PCM fixes
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: WM8580: Fix R8 initial value
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: Remove unnecessary structure definitions
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: EP93xx: fixed LRCLK rate and DMA oper. in I2S code
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: EP93xx: sampling rate range extended
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH] ASoC: fix deemphasis control in wm8904/55/60 codecs
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH v3 0/3] Add Jack support for
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH 1/2] ASoC: tlv320dac33: Fix compillation error
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- Re: [PATCH v2 1/2] ASoC: tlv320dac33: Rename outpup amplifier widget
- From: Liam Girdwood <lrg@xxxxxxxxxxxxxxx>
- [PATCH] ASoC: WM8580: Fix R8 initial value
- From: Jassi Brar <jassi.brar@xxxxxxxxxxx>
- Re: [PATCH] [ALSA] ice1712 - working M-Audio Delta 66E support
- From: Takashi Iwai <tiwai@xxxxxxx>
- [PATCH] ALSA: HDA: Quirk for Dell Vostro 320 to make microphone work
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- [PATCH] ASoC: Remove unnecessary structure definitions
- From: Seungwhan Youn <sw.youn@xxxxxxxxxxx>
- [PATCH] ASoC: EP93xx: fixed LRCLK rate and DMA oper. in I2S code
- From: Alexander <subaparts@xxxxxxxxx>
- [PATCH] ASoC: EP93xx: sampling rate range extended
- From: Alexander <subaparts@xxxxxxxxx>
- Re: [PATCH 1/3] ASoC: EP93xx I2S and PCM fixes
- From: Alexander <subaparts@xxxxxxxxx>
- Re: [PATCH 2/3] ASoC: CS4271 codec support
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 2/3] ASoC: CS4271 codec support
- From: Alexander <subaparts@xxxxxxxxx>
- [PATCH] [ALSA] ice1712 - working M-Audio Delta 66E support
- From: Brian Bloniarz <brian.bloniarz@xxxxxxxxx>
- [PATCH] ASoC: fix deemphasis control in wm8904/55/60 codecs
- From: Dmitry Artamonow <mad_soft@xxxxxxxx>
- Re: [PATCH] asoc, codec: Add SGTL5000 codec support to asoc
- From: Arnaud Patard (Rtp) <arnaud.patard@xxxxxxxxxxx>
- Re: [PATCH v3 3/3] ASoC: sdp4430: Add Jack support
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH v3 2/3] ASoC: twl6040: Add jack support for headset and handset
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH v3 3/3] ASoC: sdp4430: Add Jack support
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v3 2/3] ASoC: twl6040: Add jack support for headset and handset
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v3 1/3] ASoC: sdp4430: Enable FM stereo pins
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- [PATCH v3 0/3] Add Jack support for
- From: "Olaya, Margarita" <magi.olaya@xxxxxx>
- Re: ASoC updaets for 2.6.38
- From: Takashi Iwai <tiwai@xxxxxxx>
- Re: [PATCH v2 1/2] ASoC: tlv320dac33: Rename outpup amplifier widget
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- ASoC updaets for 2.6.38
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: ASoC updates for 2.6.37
- From: Takashi Iwai <tiwai@xxxxxxx>
- ASoC updates for 2.6.37
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 2/3] ASoC: CS4271 codec support
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- [PATCH v2 2/2] ASoC: tlv320dac33: Move DAC LR power on to a supply widget
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>
- [PATCH v2 1/2] ASoC: tlv320dac33: Rename outpup amplifier widget
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>
- Re: [PATCH] ASoC: soc-core: Fix null pointer dereference
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 2/2] ASoC: tlv320dac33: Move DACL/R power on to PGA widget
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>
- [PATCH] ASoC: Fix widgets for WM8994/58 AIF2 source control
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 2/2] ASoC: tlv320dac33: Move DACL/R power on to PGA widget
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 1/2] ASoC: tlv320dac33: Fix compillation error
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 2/3] ASoC: CS4271 codec support
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH 3/3] ASoC: EDB93xx machine sound driver
- From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
- Re: [PATCH] Fix possible sprintf overrun in snd_pcm_hw_open
- From: David Henningsson <david.henningsson@xxxxxxxxxxxxx>
- [PATCH 2/2] ASoC: tlv320dac33: Move DACL/R power on to PGA widget
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>
- [PATCH 1/2] ASoC: tlv320dac33: Fix compillation error
- From: Peter Ujfalusi <peter.ujfalusi@xxxxxxxxx>