- 09 Feb, 2007 39 commits
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Liam Girdwood authored
This patch adds pxa2xx I2S ASoC audio support. Features:- o Supports playback/capture o 16 bit PCM o 8k - 96k sample rates o Supports master and slave mode. From: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by:
Richard Purdie <rpurdie@rpsys.net> Signed-off-by:
Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by:
Takashi Iwai <tiwai@suse.de> Signed-off-by:
Jaroslav Kysela <perex@suse.cz>
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Liam Girdwood authored
This patch adds pxa2xx ASoC DMA audio support. It's based on sound/arm/pxa-pcm.c by Nicolas Pitre with the following differences. o Modified driver structure to use ASoC core PCM callbacks and data structures. o Registration with ASoC core. From: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by:
Nicolas Pitre <nico@cam.org> Signed-off-by:
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Clemens Ladisch authored
Load the DSP and controller microcode using request_firmware(), if possible, instead of using the built-in firmware. Signed-off-by:
Clemens Ladisch <clemens@ladisch.de> Signed-off-by:
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James Courtier-Dutton authored
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James Courtier-Dutton <James@superbug.co.uk> Signed-off-by:
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James Courtier-Dutton authored
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Takashi Iwai authored
Fixed the irq handler in soc/at91-at91rm9200-i2s.c to follow the new style without pt_regs. Signed-off-by:
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James Courtier-Dutton authored
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Remy Bruno authored
Added the support for mixer matrix of RME9632 rev 152. Signed-off-by:
Remy Bruno <remy.bruno@trinnov.com> Signed-off-by:
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Clemens Ladisch authored
Let the emu10k1 driver select FW_LOADER because the new Emu1010 support requires it. Signed-off-by:
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Clemens Ladisch authored
Let the AudioScience, Echoaudio and Riptide drivers select FW_LOADER instead of depending on it so that they can be configured without having to enable FW_LOADER manually. Signed-off-by:
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Clemens Ladisch authored
Properly quote a string that had an embedded newline. Signed-off-by:
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Frank Mandarino authored
This patch adds a Makefile and Kconfig to build the ASoC AT91RM9200 support. Signed-off-by:
Frank Mandarino <fmandarino@endrelia.com> Signed-off-by:
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Frank Mandarino authored
This patch adds support for the Endrelia ETI_B1 machine using the WM8731 codec and the AT91RM9200 platform. Signed-off-by:
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Frank Mandarino authored
This patch adds I2S support to the Atmel AT91RM9200 CPU. Features:- o Playback/Capture supported. o 16 Bit data size. o 8k - 48k sample rates. o ssc0, ssc1 and ssc2 supported as I2S ports. Signed-off-by:
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Frank Mandarino authored
This patch adds ASoC audio DMA support to the Atmel AT91RM9200 CPU. Features:- o Playback/Capture supported. o 16 Bit data size. Signed-off-by:
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Richard Purdie authored
This patch adds an ASoC Makefile and Kconfig for the WM8731, WM8750 and WM9712 codecs. Signed-off-by:
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Richard Purdie authored
This patch allows the std Alsa AC97 codec driver to use any AsoC AC97 controller driver. Currently, only HiFi playback and Capture are supported atm. Signed-off-by:
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Richard Purdie authored
This patch adds ASoC support for the WM9712 codec. Supported features:- o Capture/Playback/Sidetone/Bypass. o Aux DAC. o 8k - 48k sample rates. o DAPM. Signed-off-by:
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Richard Purdie authored
This patch adds ASoC support for the WM8750 codec. Supported features:- o Capture/Playback/Sidetone/Bypass. o 16 & 24 bit audio. o 8k - 96k sample rates. o DAPM. Signed-off-by:
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Richard Purdie authored
This patch adds ASoC support for the WM8731 codec. Supported features:- o Capture/Playback/Sidetone/Bypass. o 16 & 24 bit audio. o 8k - 96k sample rates. o DAPM. Signed-off-by:
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Liam Girdwood authored
This patch adds documentation describing the ASoC architecture and a maintainer entry for ASoC. The documentation includes the following files:- codec.txt: Codec driver internals. DAI.txt: Description of Digital Audio Interface standards and how to configure a DAI within your codec and CPU DAI drivers. dapm.txt: Dynamic Audio Power Management. platform.txt: Platform audio DMA and DAI. machine.txt: Machine driver internals. pop_clicks.txt: How to minimise audio artifacts. clocking.txt: ASoC clocking for best power performance. Signed-off-by:
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Liam Girdwood authored
This patch adds support for building the ASoC core and the dynamic audio power management support. Signed-off-by:
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Richard Purdie authored
This patch adds Dynamic Audio Power Management (DAPM) to ASoC. Dynamic Audio Power Management (DAPM) is designed to allow portable and handheld Linux devices to use the minimum amount of power within the audio subsystem at all times. It is independent of other kernel PM and as such, can easily co-exist with the other PM systems. DAPM is also completely transparent to all user space applications as all power switching is done within the ASoC core. No code changes or recompiling are required for user space applications. DAPM makes power switching decisions based upon any audio stream (capture/playback) activity and audio mixer settings within the device. DAPM spans the whole machine. It covers power control within the entire audio subsystem, this includes internal codec power blocks and machine level power systems. There are 4 power domains within DAPM:- 1. Codec domain - VREF, VMID (core codec and audio power) Usually controlled at codec probe/remove and suspend/resume, although can be set at stream time if power is not needed for sidetone, etc. 2. Platform/Machine domain - physically connected inputs and outputs Is platform/machine and user action specific, is configured by the machine driver and responds to asynchronous events e.g when HP are inserted 3. Path domain - audio subsystem signal paths Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. 4. Stream domain - DAC's and ADC's. Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. All DAPM power switching decisions are made automatically by consulting an audio routing map of the whole machine. This map is specific to each machine and consists of the interconnections between every audio component (including internal codec components). Signed-off-by:
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Frank Mandarino authored
This patch is the core of ASoC functionality. The ASoC core is designed to provide the following features :- o Codec independence. Allows reuse of codec drivers on other platforms and machines. o Platform driver code reuse. Reuse of platform specific audio DMA and DAI drivers on different machines. o Easy I2S/PCM digital audio interface configuration between codec and SoC. Each SoC interface and codec registers their audio interface capabilities with the core at initialisation. The capabilities are subsequently matched and configured at run time for best power and performance when the application hw params are known. o Machine specific controls/operations: Allow machines to add controls and operations to the audio subsystem. e.g. volume control for speaker amp. To achieve all this, ASoC splits an embedded audio system into 3 components :- 1. Codec driver: The codec driver is platform independent and contains audio controls, audio interface capabilities, codec dapm and codec IO functions. 2. Platform driver: The platform driver contains the audio dma engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. 3. Machine driver: The machine driver handles any machine specific controls and audio events. i.e. turning on an amp at start of playback. Signed-off-by:
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Richard Purdie authored
This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by:
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Takashi Iwai authored
Use pci_iomap and ioread*/iowrite*() functions for accessing hardwares. pci_iomap is suitable for hardwares like ICH and compatible that have both PIO and MMIO. Signed-off-by:
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Jaroslav Kysela authored
Users ask us many times about the maximum DMA size for PCM devices. This file gives them a hint in KB. Signed-off-by: -
Jaroslav Kysela authored
See ALSA bug#2481 . Signed-off-by: -
Takashi Iwai authored
Fixed a compile warning regarding print format for size_t. Signed-off-by:
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Johannes Berg authored
This patch converts most uses of list_for_each to list_for_each_entry all across alsa. In some place apparently an item can be on a list with different pointers so of course that isn't compatible with list_for_each, I therefore didn't touch those places. Signed-off-by:
Johannes Berg <johannes@sipsolutions.net> Signed-off-by:
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Johannes Berg authored
This patch changes i2sbus_attach_codec to implement a proper error handling strategy using labels to jump to the right part. Since it has an elaborate set-up sequence it also needs that tear-down, which I had hard-coded inbetween all the checks. This increases readability and should reduce .text size as well. Signed-off-by:
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Johannes Berg authored
This patch makes a few whitespace cleanups and makes i2sbus assign the new struct device pointer in struct snd_pcm so that the proper device symlink shows up in sysfs. Signed-off-by:
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Johannes Berg authored
This patch adds a struct device pointer to struct snd_pcm in order to be able to give it a different device than the card. It defaults to the card's device, however, so it should behave identically for drivers not touching the field. Signed-off-by:
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Johannes Berg authored
This patch adds snd_register_device_for_dev taking a struct device pointer to link the new device to and makes snd_register_device a simple static inline wrapper around it. Signed-off-by:
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Jochen Voss authored
Enable the analog loopback of the Revolution 5.1 card. This patch adds support for the PT2258 volume controller and modifies the Revolution 5.1 driver to make use of this facility. This allows to control the analog loopback of the card. Signed-off-by:
Jochen Voss <voss@seehuhn.de> Signed-off-by:
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Jochen Voss authored
Enable capture from line-in and CD on the Revolution 5.1 card. This patch adds support for switching between the 5 input channels of the AK5365 ADC and modifies the Revolution 5.1 driver to make use of this facility. Previously the capture channel was fixed to channel 0 (microphone on the Revolution 5.1 card). Signed-off-by:
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Clemens Ladisch authored
Add pause capabilities for both USB playback and capture streams. Signed-off-by:
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Clemens Ladisch authored
The hardware information structures for playback and capture streams, respectively, are the same, so we can use just one structure for both streams. Signed-off-by:
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James Courtier-Dutton authored
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- 08 Feb, 2007 1 commit
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Linus Torvalds authored
* 'upstream-linus' of master.kernel.org:/pub/scm/linux/kernel/git/mfasheh/ocfs2: (22 commits) configfs: Zero terminate data in configfs attribute writes. [PATCH] ocfs2 heartbeat: clean up bio submission code ocfs2: introduce sc->sc_send_lock to protect outbound outbound messages [PATCH] ocfs2: drop INET from Kconfig, not needed ocfs2_dlm: Add timeout to dlm join domain ocfs2_dlm: Silence some messages during join domain ocfs2_dlm: disallow a domain join if node maps mismatch ocfs2_dlm: Ensure correct ordering of set/clear refmap bit on lockres ocfs2: Binds listener to the configured ip address ocfs2_dlm: Calling post handler function in assert master handler ocfs2: Added post handler callable function in o2net message handler ocfs2_dlm: Cookies in locks not being printed correctly in error messages ocfs2_dlm: Silence a failed convert ocfs2_dlm: wake up sleepers on the lockres waitqueue ocfs2_dlm: Dlm dispatch was stopping too early ocfs2_dlm: Drop inflight refmap even if no locks found on the lockres ocfs2_dlm: Flush dlm workqueue before starting to migrate ocfs2_dlm: Fix migrate lockres handler queue scanning ocfs2_dlm: Make dlmunlock() wait for migration to complete ocfs2_dlm: Fixes race between migrate and dirty ...
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