- 03 8月, 2010 4 次提交
-
-
由 Axel Lin 提交于
da7210 should be kfreed if da7210_init() return error. This patch also fixes the error handing in the case of snd_soc_register_dai() fail by adding snd_soc_unregister_codec() in error path. Signed-off-by: NAxel Lin <axel.lin@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Axel Lin 提交于
ak4642 should be kfreed if ak4642_init() return error. Signed-off-by: NAxel Lin <axel.lin@gmail.com> Reviewed-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Axel Lin 提交于
ad1836 is allocated in ad1836_spi_probe() but is not freed if ad1836_register() return -EINVAL (if another ad1836 is registered). Signed-off-by: NAxel Lin <axel.lin@gmail.com> Acked-by: NBarry Song <21cnbao@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Ian Lartey 提交于
The WM8741 is a very high performance stereo DAC designed for audio applications such as professional recording systems, A/V receivers and high specification CD, DVD and home theatre systems. The device supports PCM data input word lengths from 16 to 32-bits and sampling rates up to 192kHz. The WM8741 also supports DSD bit-stream data format, in both direct DSD and PCM-converted DSD modes. TODO: Expand wm8741_set_dai_sysclk and rate_constraint members to allow for all supported sample rate / Master Clock frequency combinations. Fully enable control of supplies. Signed-off-by: NIan Lartey <ian@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 02 8月, 2010 5 次提交
-
-
由 Takashi Iwai 提交于
Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
-
由 Peter Ujfalusi 提交于
The use of sDMA packet mode in THRESHOLD mode removes the restriction on the period size. With the extended THRESHOLD mode user space can ask for any period size it wishes, and the driver will configure the sDMA and McBSP FIFO accordingly. Replace the hw_rule for the period size with static constraint, which will make sure that the period size will be always even (to avoid prime period size, which could be possible in mono stream) Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NJarkko Nikula <jhnikula@gmail.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Peter Ujfalusi 提交于
Utilize the sDMA controller's packet syncronization mode, when the McBSP FIFO is in use (by extending the THRESHOLD mode). When the sDMA is configured for packet mode, the sDMA frame size does not need to match with the McBSP threshold configuration. Uppon DMA request the sDMA will transfer packet size number of words, and still trigger interrupt on frame boundary. The patch extends the original THRESHOLD mode by doing the following: if (period_words <= max_threshold) Current THRESHOLD mode configuration Otherwise (period_words > max_threshold) McBSP threshold = sDMA packet size sDMA frame size = period size With the extended THRESHOLD mode we can remove the constraint for the maximum period size, since if the period size is bigger than the maximum allowed threshold, than the driver will switch to packet mode, and picks the best (biggest) threshold value, which can divide evenly the period size. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NJarkko Nikula <jhnikula@gmail.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Peter Ujfalusi 提交于
To make the code a bit more readable, change the indexed references to the omap_mcbsp_dai_dma_params elements with pointer. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NJarkko Nikula <jhnikula@gmail.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Peter Ujfalusi 提交于
In preparation for the extended threshold mode (sDMA packet mode support), the code need to be restructured. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NJarkko Nikula <jhnikula@gmail.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
- 30 7月, 2010 3 次提交
-
-
由 Kuninori Morimoto 提交于
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Kuninori Morimoto 提交于
Current FSI driver id is not only 0 Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Kuninori Morimoto 提交于
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 29 7月, 2010 2 次提交
-
-
由 Peter Ujfalusi 提交于
Platform parameter to enable automatic FIFO configuration when the codec is in Mode1 or Mode7 FIFO mode. When this mode is selected, the controls for changing nSample (in Mode1), and UTHR (in Mode7) are not added. The driver configures the FIFO configuration based on the stream's period size in a way, that every burst will read period size of data from the host. In Mode7 we need to use a formula, which gives close enough aproximation for the burst length from the host point of view. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Peter Ujfalusi 提交于
Replace the hardwired latency definition with platform data parameter, and simplify the nSample parameter calculation. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
- 27 7月, 2010 1 次提交
-
-
由 Peter Ujfalusi 提交于
There is no need to handle POST_PMU, POST_PMD event with the Capture Route widget. It is enough to handle POST_REG event, since that will come when the user changes the routing, and we will switch the needed bits in the registers. Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
- 23 7月, 2010 1 次提交
-
-
由 Kuninori Morimoto 提交于
HeadPhone Playback Volume control register of DA7210 has reserved area. This patch considered it as mute. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 21 7月, 2010 2 次提交
-
-
由 Takashi Iwai 提交于
Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
-
由 Peter Ujfalusi 提交于
When digital microphones are connected to twl, delay is needed after enabling the digimic interface of the codec. Add new parameter for the setup data, which can be used to pass the apropriate delay in ms after the digimic interface has been enabled. Without certain delay (in certain HW configuration) the beggining of the recorded sample contains a glitch, which is generated by the digital microphones. Delaying the micbias1, 2 (which is the bias for the digimic0 or 1) does not help, since the glitch is coming after switching the digimic interface. Reversing the micbias and digimic enable order does not work either (in that case the wait need to be added after the micbias enabled). Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
- 20 7月, 2010 8 次提交
-
-
由 Mark Brown 提交于
AIF1ADC TDM mode has no effect other than causing the ADCDAT line to be tristated rather than driven low on clock cycles where there is no data to be transmitted. If the clock cycle is idle then there should be no devices using the data so tristating should have no adverse effects. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Sekhar Nori 提交于
Currently the EDMA queue to be used by for servicing ASP through internal RAM is fixed to EDMAQ_0 and that to service internal RAM from external RAM is fixed to EDMAQ_1. This may not be the desirable configuration on all platforms. For example, on DM365, queue 0 has large fifo size and is more suitable for video transfers. Having audio and video transfers on the same queue may lead to starvation on audio side. platform data as defined currently passes a queue number to the driver but that remains unused inside the driver. Fix this by defining one queue each for ASP and RAM transfers in the platform data and using it inside the driver. Since EDMAQ_0 maps to 0, thats the queue that will be used if the asp queue number is not initialized. None of the platforms currently utilize ping-pong transfers through internal RAM so that functionality remains unchanged too. This patch has been tested on DM644x and OMAP-L138 EVMs. Signed-off-by: NSekhar Nori <nsekhar@ti.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Chanwoo Choi 提交于
This patch modify I2Sv2 driver to support Samsung SoC(S5PV210). Signed-off-by: NChanwoo Choi <cw00.choi@samsung.com> Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: NKyungmin Park <kyungmin.park@samsung.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Mark Brown 提交于
-
由 Chanwoo Choi 提交于
Otherwise all machine drivers need to do so. Signed-off-by: NChanwoo Choi <cw00.choi@samsung.com> Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: NKyungmin Park <kyungmin.park@samsung.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Eric Bénard 提交于
Signed-off-by: NEric Bénard <eric@eukrea.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 18 7月, 2010 7 次提交
-
-
由 Kulikov Vasiliy 提交于
If kzalloc() fails we must exit with -ENOMEM. Also we must free allocated runtime->private_data on error as it would be lost on next call to snd_imx_open(). Signed-off-by: NKulikov Vasiliy <segooon@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Kulikov Vasiliy 提交于
If kzalloc() fails we must exit with -ENOMEM. Also we must free allocated runtime->private_data on error as it would be lost on next call to snd_imx_open(). Signed-off-by: NKulikov Vasiliy <segooon@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Kulikov Vasiliy 提交于
The code checks 'davinci_vc' after kzalloc() and do not checks 'davinci_vcif_dev' that kzalloc() result is assigned to. It seems that it is a typo (autocompletion?). Signed-off-by: NKulikov Vasiliy <segooon@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Kuninori Morimoto 提交于
Specified ID is necessary, when some codecs are used with FSI. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Mark Brown 提交于
-
由 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Cc: stable@kernel.org
-
由 Jorge Eduardo Candelaria 提交于
In order to reduce pop-noise at powering up/down of the DACs and Drivers, these components have to be handled in a specific sequence. Headset, Handsfree, and Earphone drivers are now registered as PGA components to ensure DACs are enabled first. Also, add a delay to leave time for DACs to settle before continuing power up/down sequence. Signed-off-by: NJorge Eduardo Candelaria <jorge.candelaria@ti.com> Signed-off-by: NMargarita Olaya Cabrera <magi.olaya@ti.com> Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
- 15 7月, 2010 4 次提交
-
-
由 Mark Brown 提交于
When a device is powered down volatile registers can't be read so attempts to display codec_reg will show error values, and obviously it is also possible for there to be hardware errors too. Check for errors from reads and display them more clearly when formatting codec_reg. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
-
由 Mark Brown 提交于
-
由 Axel Lin 提交于
snd_soc_unregister_codec is called twice if snd_soc_register_dai fail. Signed-off-by: NAxel Lin <axel.lin@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
由 Axel Lin 提交于
otherwise the error path will always be executed. Signed-off-by: NAxel Lin <axel.lin@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
- 13 7月, 2010 3 次提交
-
-
由 Manuel Lauss 提交于
Annotate platform probe callback with __devinit instead of plain __init. Signed-off-by: NManuel Lauss <manuel.lauss@googlemail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
The controller has mute/unmute capability and some bootloader may mute them at boot. If it's not handled, all things will seem to be working but no sound will come out of the speaker/headphone. Signed-off-by: NArnaud Patard <arnaud.patard@rtp-net.org> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-
Kirkwood controller needs to be informed if the audio stream is mono or not. Failing to do so will result in playing at the wrong speed. Signed-off-by: NArnaud Patard <arnaud.patard@rtp-net.org> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
-