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* Subject: [PATCH 050/104] ASoC: twl6040: Remove pll and headset mode dependencyMisael Lopez Cruz2012-08-17
| | | | | | | | | | | | | | Remove dependency between pll (hppll, lppll) and headset power mode (low-power, high-performance), as headset power mode can be used with any pll. A new control is created to allow headset power mode configuration from userspace. Changing headset power mode during earpiece related usecases is not propagated down to the codec as earpiece requires HS DAC in HP mode. Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
* Subject: [PATCH 049/104] ASoC: twl6040: Support other sample rates in ↵Liam Girdwood2012-08-17
| | | | | | | | | | constraints. Add other supported sample rates to LP and HP modes. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Subject: [PATCH 048/104] ASoC: twl6040: add all ABE DAIsLiam Girdwood2012-08-17
| | | | | | | | Add all DAIs to fully support OMAP4 ABE. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Subject: [PATCH 044/104] ASoC: twl6040: Convert into TWL6040 MFD childMisael Lopez Cruz2012-08-17
| | | | | | | | | | | | Convert TWL6040 CODEC driver into a TWL6040 MFD child, it implies that MFD-level operations like register accesses, clock setting and power management are done through MFD APIs, not directly by CODEC driver anymore. To avoid conflicts with the other MFD child, vibrator registers are skipped in CODEC driver. Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Subject: [PATCH 042/104] MFD: twl4030-audio: Rename platform dataPeter Ujfalusi2012-08-17
| | | | | | | | | | | | Allign the platform data names for twl4030 audio submodule: twl4030_audio_data: for the core MFD driver twl4030_codec_data: for ASoC codec driver twl4030_vibra_data: for the input/ForceFeedback driver To avoid breakage, change all depending drivers, files to use the new types. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
* Subject: [PATCH 041/104] MFD: twl4030-codec -> twl4030-audio: Rename the driverPeter Ujfalusi2012-08-17
| | | | | | | | | Rename the driver, and header file from twl4030-codec to twl4030-audio. To avoid breakage change depending drivers at the same time. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> CC: Misael Lopez Cruz <misael.lopez@ti.com>
* Subject: [PATCH 036/104] ASoC: core - platform widget IOLiam Girdwood2012-08-17
| | | | | | Allow platform driver widgets to perform any IO required via DAPM. Signed-off-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 035/104] ASoC: core - add API call to register platform ↵Liam Girdwood2012-08-17
| | | | | | | | | | kcontrols. In preparation for Dynamic PCM (AKA DSP) support. Allow platform drivers to register kcontrols. Signed-off-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 034/104] ASoC: core - add platform IO tracingLiam Girdwood2012-08-17
| | | | Signed-off-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 033/104] ASoC: core - Add platform read and write.Liam Girdwood2012-08-17
| | | | | | | | | In preparation for ASoC Dynamic PCM (AKA DSP) support. Allow platform driver to perform IO. Intended for platform DAPM. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Subject: [PATCH 032/104] ASoC: core - Make platform probe more like codec probe.Liam Girdwood2012-08-17
| | | | | | | | | | In preparation for ASoC dynamic PCM support (AKA ASoC DSP) Platform will also support DAPM so separate out the probe function to simplify the code (just like the codec probe). Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Subject: [PATCH 031/104] ASoC: Fix DAPM sequence run for per-widget I/O methodsMark Brown2012-08-17
| | | | | | | | | | | | | Previously we were using the DAPM context rather than a widget as the argument for update_bits() so we didn't need to care that our list walk of widgets left us one beyond the end of the list. Now we're using them for the register update we need to make sure we're pointing at an actual widget not the list_head. Fix originally suggested by Liam on IM. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 030/104] ASoC: dapm - Refactor widget IO functions in ↵Liam Girdwood2012-08-17
| | | | | | | | | | | | | | | | preparation for platform widgets. This time with soc_widget_update_bits reflecting recent soc_update_bits changes. Currently widget IO is tightly coupled to the CODEC drivers. Future platform DSP devices have mixer components that can alter power usage and hence require full DAPM support. This provides a generic widget IO operation wrapper in preparation for future patches that implement platform driver DAPM. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Subject: [PATCH 028/104] ASoC: Don't use codec->control_data in bulk writeMark Brown2012-08-17
| | | | | | | | | In order to facilitate merging with the register map I/O replace the use of control_data for the bulk writes with direct lookup of the client data from the device. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 027/104] ASoC: Add weak routes for sidetone style pathsMark Brown2012-08-17
| | | | | | | | | | | | | | | | | | Normally DAPM will power up any connected audio path. This is not ideal for sidetone paths as with sidetone paths the audio path is not wanted in itself, it is only desired if the two paths it provides a sidetone between are both active. If the sidetone path causes a power up then it can be hard to minimise pops as we first power up either the sidetone or the main output path and then power the other, with the second power up potentially introducing a DC offset. Address this by introducing the concept of a weak path. If a path is marked as weak then DAPM will ignore that path when walking the graph, though all the relevant controls are still available to the application layer to allow these paths to be configured. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 026/104] ASoC: Move register I/O code into a separate fileMark Brown2012-08-17
| | | | | | | | | For clarity and to help ongoing refactoring in this area create a new file to contain the physical I/O functions, separating them out from the cache operations. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 025/104] ASoC: Factor out redundant read() functionsMark Brown2012-08-17
| | | | | | | | | We've got a whole bunch of functions which just call straight through to do_hw_read(). Simplify this situation by removing them and using hw_read() directly. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 024/104] ASoC: pcm - rename snd_codec_close() to ↵Liam Girdwood2012-08-17
| | | | | | | | | snd_pcm_close(). Make sure we follow naming convention for all PCM ops. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Subject: [PATCH 023/104] ASoC: core - PCM mutex per rtdLiam Girdwood2012-08-17
| | | | | | | | | | | | | | | | | | | | | In preparation for the new ASoC Dynamic PCM support (AKA DSP support). The new ASoC Dynamic PCM core allows DAIs to be dynamically re-routed at runtime between the PCM device end (or Frontend - FE) and the physical DAI (Backend - BE) using regular kcontrols (just like a hardware CODEC routes audio in the analog domain). The Dynamic PCM core therefore must be able to call PCM operations for both the Frontend and Backend(s) DAIs at the same time. Currently we have a global pcm_mutex that is used to serialise the ASoC PCM operations. This patch removes the global mutex and adds a mutex per RTD allowing the PCM operations to be reentrant and allow control of more than one DAI at at time. e.g. a frontend PCM hw_params() could configure multiple backend DAI hw_params() with similar or different hw parameters at the same time. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Subject: [PATCH 022/104] ASoC: core - Separate out PCM operations into new file.Liam Girdwood2012-08-17
| | | | | | | | | | | | In preparation for Dynamic PCM support (AKA DSP support). There will be future patches that add support to allow PCMs to be dynamically routed to multiple DAIs at startup and also during stream runtime. This patch moves the ASoC core PCM operaitions into a new file called soc-pcm.c. This will in simplify the ASoC core features into distinct files. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Subject: [PATCH 021/104] ASoC: twl6040 - According to TWL6040 specification, ↵Ricardo Neri2012-08-17
| | | | | | | | gain start at 6dB and not -6dB. Signed-off-by: Ricardo Neri <ricardo.neri@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 020/104] ASoC: Suppress restore of default register values ↵Mark Brown2012-08-17
| | | | | | | | | | | | | | | | for rbtree cache sync Currently the rbtree code will write out the entire register map when doing a cache sync which is wasteful and will slow things down. Check to see if the value we're about to write is the default and don't bother restoring it if it is, either the value will have been retained or the device will have been reset and holds the value already. We should really store the defaults in the nodes but this resolves the immediate issue. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 019/104] ASoC: core - Allow components to probe/remove in ↵Liam Girdwood2012-08-17
| | | | | | | | | | | | | sequence. Some ASoC components depend on other ASoC components to provide clocks and power resources in order to probe() and vice versa for remove(). Allow components to be ordered so that components can be probed() and removed() in sequences that conform to their dependencies. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Subject: [PATCH 018/104] ASoC: core - Optimise and refactor pcm_new() to ↵Liam Girdwood2012-08-17
| | | | | | | | | | | | pass only rtd Currently pcm_new() passes in 3 arguments :- card, pcm and DAI. Refactor this to only pass in 1 argument (i.e. the rtd) since struct rtd contains card, pcm and DAI along with other members too that are useful too. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Subject: [PATCH 017/104] ASoC: Only provide a default bias level update for ↵Mark Brown2012-08-17
| | | | | | | | | | CODEC contexts This allows the card driver to use the bias level variable more easily in multi component systems. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 016/104] ASoC: Add context parameter to card DAPM callbacksMark Brown2012-08-17
| | | | | | | | | The card callback will get called for each DAPM context in the card so it can be useful for it to know which device is currently undergoing a transition. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 015/104] ASoC: Simplify logic in snd_soc_dapm_set_bias_level()Mark Brown2012-08-17
| | | | | | | No functional changes but much less indentation. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 014/104] ASoC: Remove trace for DAPM bias level loggingMark Brown2012-08-17
| | | | | | | It's redundant now thanks to the use of the generic trace infrastructure. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 013/104] ASoC: Indentation fix for null loop operationMark Brown2012-08-17
| | | | | | | More with the legibility. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 012/104] ASoC: Don't bring the CODEC up to full power for ↵Mark Brown2012-08-17
| | | | | | | | | | | | | | | supplies and biases If the only widgets active within a CODEC are supplies and micbiases we are not passing audio, we are probably just doing microphone detection. This will not generally require either fully accurate reference voltages or much power so If this turns out to be unsuitable for some systems we can provide a facility to override this decision. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 011/104] ASoC: Specify target bias state directly as a bias ↵Mark Brown2012-08-17
| | | | | | | | | | | | state Rather than a simple flag to say if we want the DAPM context to be at full power specify the target bias state. This should have no current effect but is a bit more direct and so makes it easier to change our decisions about the which bias state to go into in future. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 010/104] ASoC: Enforce the mask in snd_soc_update_bits()Mark Brown2012-08-17
| | | | | | | | Avoids issues if someone does a read followed by restore and doesn't mask out only the bits being updated. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 009/104] ASoC: OMAP4: Add HDMI Audio machine driver for ↵Ricardo Neri2012-08-17
| | | | | | | | | | | | OMAP4 boards Add machine driver for HDMI audio on OMAP4 boards. This driver is in charge of putting together the HDMI audio codec and the CPU DAI and register the HDMI sound card with ALSA. Signed-off-by: Ricardo Neri <ricardo.neri@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 008/104] ASoC: OMAP: Add CPU DAI driver for HDMIRicardo Neri2012-08-17
| | | | | | | | | | Addition of the HDMI CPU DAI driver for OMAP4. This driver is in charge of configuring DMA settings for HDMI. Also, it finds the HDMI video device and determines if audio playback can proceed. Signed-off-by: Ricardo Neri <ricardo.neri@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 007/104] ASoC: Use explicit endianness conversion in ↵Mark Brown2012-08-17
| | | | | | | snd_soc_16_8_write() Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 006/104] ASoC: Use cpu_to_be16() in 8x16 writeMark Brown2012-08-17
| | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 005/104] ASoC: Convert 7x9 write to use cpu_to_be16()Mark Brown2012-08-17
| | | | | | | Run the data through cpu_to_be16() so it's at least clear what we're up to. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 003/104] ASoC: Convert 16x16 write to use cpu_to_be16()Mark Brown2012-08-17
| | | | | | | Make it clear what we're doing. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* Subject: [PATCH 002/104] ASoC: soc-cache: Cache a pointer to the last ↵Dimitris Papastamos2012-08-17
| | | | | | | | | | | | | | accessed rbnode Whenever we are doing a read or a write through the rbtree code, we'll cache a pointer to the rbnode. To avoid looking up the register everytime we do a read or a write, we first check if it can be found in the cached register block, otherwise we traverse the rbtree and finally cache the rbnode for future use. Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Subject: [PATCH 001/104] ASoC: soc-cache: Block based rbtree compressionDimitris Papastamos2012-08-17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch prepares the ground for the actual rbtree optimization patch which will save a pointer to the last accessed rbnode that was used in either the read() or write() functions. Each rbnode manages a variable length block of registers. There can be no two nodes with overlapping blocks. Each block has a base register and a currently top register, all the other registers, if any, lie in between these two and in ascending order. The reasoning behind the construction of this rbtree is simple. In the snd_soc_rbtree_cache_init() function, we iterate over the register defaults provided by the driver. For each register value that is non-zero we insert it in the rbtree. In order to determine in which rbnode we need to add the register, we first look if there is another register already added that is adjacent to the one we are about to add. If that is the case we append it in that rbnode block, otherwise we create a new rbnode with a single register in its block and add it to the tree. In the next patch, where a cached rbnode is used by both the write() and the read() functions, we also check if the register we are about to add is in the cached rbnode (the least recently accessed one) and if so we append it in that rbnode block. Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm8994: Ensure there are enough BCLKs for four channelsMark Brown2012-08-13
| | | | | | | | | | | | | BugLink: http://bugs.launchpad.net/bugs/1034988 commit b8edf3e5522735c8ce78b81845f7a1a2d4a08626 upstream. Otherwise if someone tries to use all four channels on AIF1 with the device in master mode we won't be able to clock out all the data. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org> Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com>
* ASoC: wm8962: Allow VMID time to fully rampMark Brown2012-08-13
| | | | | | | | | | | | BugLink: http://bugs.launchpad.net/bugs/1034988 commit 9d40e5582c9c4cfb6977ba2a0ca9c2ed82c56f21 upstream. Required for reliable power up from cold. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org> Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com>
* ALSA: mpu401: Fix missing initialization of irq fieldTakashi Iwai2012-08-13
| | | | | | | | | | | | | | | | BugLink: http://bugs.launchpad.net/bugs/1034988 commit bc733d495267a23ef8660220d696c6e549ce30b3 upstream. The irq field of struct snd_mpu401 is supposed to be initialized to -1. Since it's set to zero as of now, a probing error before the irq installation results in a kernel warning "Trying to free already-free IRQ 0". Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=44821 Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org> Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com>
* ALSA: snd-usb: fix clock source validity indexDaniel Mack2012-08-13
| | | | | | | | | | | | | | | | | | | | BugLink: http://bugs.launchpad.net/bugs/1034988 commit aff252a848ce21b431ba822de3dab9c4c94571cb upstream. uac_clock_source_is_valid() uses the control selector value to access the bmControls bitmap of the clock source unit. This is wrong, as control selector values start from 1, while the bitmap uses all available bits. In other words, "Clock Validity Control" is stored in D3..2, not D5..4 of the clock selector unit's bmControls. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Andreas Koch <andreas@akdesigninc.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org> Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com>
* ALSA: hda - Add support for Realtek ALC282David Henningsson2012-08-13
| | | | | | | | | | | | | | | | BugLink: http://bugs.launchpad.net/bugs/1034988 commit 4e01ec636e64707d202a1ca21a47bbc6d53085b7 upstream. This codec has a separate dmic path (separate dmic only ADC), and thus it looks mostly like ALC275. BugLink: https://bugs.launchpad.net/bugs/1025377 Tested-by: Ray Chen <ray.chen@canonical.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org> Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com>
* ASoC: tlv320aic3x: Fix codec pll configure bugHebbar, Gururaja2012-07-23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | BugLink: http://bugs.launchpad.net/bugs/1025406 commit c9fe573a6584034670c1a55ee8162d623519cbbf upstream. In sound/soc/codecs/tlv320aic3x.c data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); In the above code, pll-p value is OR'ed with previous value without clearing it. Bug is not seen if pll-p value doesn't change across Sampling frequency. However on some platforms (like AM335x EVM-SK), pll-p may have different values across different sampling frequencies. In such case, above code configures the pll with a wrong value. Because of this bug, when a audio stream is played with pll value different from previous stream, audio is heard as differently(like its stretched). Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org> Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com>
* ALSA: hda - Add Realtek ALC280 codec supportDavid Henningsson2012-07-23
| | | | | | | | | | | | | | | | BugLink: http://bugs.launchpad.net/bugs/1025406 commit befae82e2906cb7155020876a531b0b8c6c8d8c8 upstream. This chip looks very similar to ALC269 and ALC27* variants. The bug reporter has verified that sound was working after this patch had been applied. BugLink: https://bugs.launchpad.net/bugs/1017017 Tested-by: Richard Crossley <richardcrossley@o2.co.uk> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org> Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com>
* ALSA: usb-audio: fix rate_list memory leakClemens Ladisch2012-07-03
| | | | | | | | | | | | | | | BugLink: http://bugs.launchpad.net/bugs/1013748 commit 5cd5d7c44990658df6ab49f6253c39617c53b03d upstream. The array of sample rates is reallocated every time when opening the PCM device, but was freed only once when unplugging the device. Reported-by: "Alexander E. Patrakov" <patrakov@gmail.com> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org> Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com>
* ALSA: HDA: Lessen CPU usage when waiting for chip to respondDavid Henningsson2012-05-25
| | | | | | | | | | | | | | | | | | | | | BugLink: http://bugs.launchpad.net/bugs/1002880 commit 32cf4023e689ad5b3a81a749d8cc99d7f184cb99 upstream. When an IRQ for some reason gets lost, we wait up to a second using udelay, which is CPU intensive. This patch improves the situation by waiting about 30 ms in the CPU intensive mode, then stepping down to using msleep(2) instead. In essence, we trade some granularity in exchange for less CPU consumption when the waiting time is a bit longer. As a result, PulseAudio should no longer be killed by the kernel for taking up to much RT-prio CPU time. At least not for *this* reason. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Tested-by: Arun Raghavan <arun.raghavan@collabora.co.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org> Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
* ALSA: echoaudio: Remove incorrect part of assertionMark Hills2012-05-25
| | | | | | | | | | | | | | | | | | | | | | | | | | | BugLink: http://bugs.launchpad.net/bugs/1002880 commit c914f55f7cdfafe9d7d5b248751902c7ab57691e upstream. This assertion seems to imply that chip->dsp_code_to_load is a pointer. It's actually an integer handle on the actual firmware, and 0 has no special meaning. The assertion prevents initialisation of a Darla20 card, but would also affect other models. It seems it was introduced in commit dd7b254d. ALSA sound/pci/echoaudio/echoaudio.c:2061 Echoaudio driver starting... ALSA sound/pci/echoaudio/echoaudio.c:1969 chip=ebe4e000 ALSA sound/pci/echoaudio/echoaudio.c:2007 pci=ed568000 irq=19 subdev=0010 Init hardware... ALSA sound/pci/echoaudio/darla20_dsp.c:36 init_hw() - Darla20 ------------[ cut here ]------------ WARNING: at sound/pci/echoaudio/echoaudio_dsp.c:478 init_hw+0x1d1/0x86c [snd_darla20]() Hardware name: Dell DM051 BUG? (!chip->dsp_code_to_load || !chip->comm_page) Signed-off-by: Mark Hills <mark@pogo.org.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org> Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
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