| Commit message (Collapse) | Author | Age | Files | Lines |
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The sub mic is on the right capture path, so when the front
end portion of the route is selected, the mic choice must
be taken into account. Fixes the lack of sound in camcorder.
Fixes bug 5350006
Change-Id: I347922af04a0114a8e269b9edea3eec260175f79
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When the phone is ringing the active output device is always
the speaker, perhaps with a secondary output device such as
headset. When we answer the call the active output device is
still speaker, and set_mode() causes the modem PCM to be
opened for this route. However, we never use the speaker as
our initial audio route for call audio. This change forces
speaker off when we set up the initial in-call state so we
don't have to change it immediately when out_set_parameters()
is called with a different route.
This works for earpiece, headset and headphones. It doesn't
help bluetooth because the SCO connection is only begun
after the call is started.
Change-Id: Ie9f411c61570749fc26ab2ffa18cd1477e68a7e6
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Mono mics were previously only routed to a single channel
on each mux. Route through both instead.
Change-Id: Ie954a436ec24e377e6821b85b994ed5294a6c4d8
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This reverts commit 6844413bd8fcc4139eb106a4bdf903aaf90598df.
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Add basic support of HDMI output during playback.
Note that if multiple output devices are set, if
HDMI is one of them, only HDMI will be used.
Change-Id: I0a3ccdd6824a73553649e63b2d6ccde6aa99310e
Signed-off-by: Chris Kelly <c-kelly@ti.com>
Signed-off-by: Eric Laurent <elaurent@google.com>
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Call ril_set_call_audio_path() after the modem PCMs are
opened so that if it blocks, there will at least be call
audio.
Change-Id: Ibf4305150cf18cad83b88d57e3be4ac8399ae77f
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In addition, stop turning on the headset DACs when only
the earpiece is required.
DO NOT MERGE
Change-Id: Ie26e705520efece8cdb0dbc93bcd98411c804563
Signed-off-by: PankajJindal <pankajjindal@ti.com>
Signed-off-by: Simon Wilson <simonwilson@google.com>
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Cross-dependency on kernel change:
I4b85eebf18e99b106816131bd927cf0962055dcd
The earpiece volume has been increased by 6dB because of
dynamic route gain adjustment, so the sidetone gain must
be decreased by the same amount otherwise there is too
much feedback and we are outside specification.
Change-Id: I6b268105553ab68e9b0e9f18d41c018823d1e6cb
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The output devices in adev->devices are cleared sometimes when
making a call. The sequence is as follows:
1. do_output_standby() (clears bits in adev->devices)
2. set_mode to IN_CALL state
3. select_output_device() reads the bits in adev->devices, but
none are set.
As a result, with no valid route, call audio fails.
Fixes bug 5309421
Change-Id: I81efe325d8b482f7474750c08d353ca989da9939
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This reverts commit 494a9150218d73774796c40bc101928034094082.
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Cross-dependency with kernel change:
I65a3555569bf4698619130c80d5c391bb6bb9b46
Change-Id: Ibfd6a884626a21ad1a06572e3458cca1b31e3afc
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This reverts commit b48dbbdeab6f28cf99dc25da67e696ac1399c659.
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Support for PORT_VX as an input capture device was not completely
removed and the bluetooth uplink was still incorrectly using the
VX MUX. PORT_VX support has been completely removed and bluetooth
now uses the correct MUX for uplink.
Fixes bug 5279972
Change-Id: I8664abf7cff61f894f447dc7a3c49241dce4087b
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The OMAP4 ALSA kernel code can now handle output routing
changes when the PCM is opened. This avoids pops when
closing PCMs to change the route between speaker and
headset for example, and makes a noticeable difference
when notifications occur when playing music.
Change-Id: I957d96fae6764a3049d4f3c00074a9295a18d66d
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EC & NR function can be duplicated in modem and bluetooth device.
If BT device want to use own function, modem has to turn off own
functions.
This can be related with clicking sound and sound quality in some
case of somde bluetooth device or modem's configuration.
Change-Id: Ifebc824e04afc06cd861a67138a1e06ce3f462f1
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Change-Id: Ic055b9680623ad9d9ad1d8edfbc9bafceab4c43a
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Calibrate the input levels for voice recognition
on the main microphone (bottom mic) and headset inputs to
the value expected in this use case.
Change-Id: I6c0743bb9ae4c00194a8baeed43f523918a1a10e
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This filters out frequencies that can damage the speaker.
Change-Id: I35946c9ee3e80be673643ef40129e7e5214a0d8b
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PORT_VX and PORT_MM2_UL cannot be opened at the same time,
and doing so causes loss of audio. When a voice call is
taken when a video call is in progress, the modem is opened
before the capture stream is ended so the problem occurs.
Using PORT_MM2_UL ensures we don't hit this case.
Fixes bug 5221406
Change-Id: Id6aa26e5321e74375a51b455aa55723df2287c35
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Added support for audio pre processing and echo reference
for AEC.
Also:
- added defines for ABE ports sampling rates
- always select sub mic for camcorder and VoIP on speakerphone
even if headset with mic is present
- change mutex locking order: first hw device then stream.
This allows calling functions on active output and input streams
without releasing the hw device mutex.
Aquiring the hw device mutex systematically in dtream read and write
guarantees that a low priority thread waiting on the stream mutex will
get it in a timely manner.
Change-Id: I4abc9e56b30e7b72109db1961af76c6fd4c03be0
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This change applies conversions so that analog codec volumes
can be specified in dB. It also restores the DL2 ABE gain to
0dB now that the ABE kernel code has been patched to prevent
speaker distortion. The headset and speaker volumes are
adjusted to take this change into account.
Change-Id: I5cfe465e30e0c6a2424bd05e4a412eae8d878eba
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Change-Id: I4ea7f3795ba571fdd395f3fff4cd3e485d0e89a8
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This change causes bluetooth SCO audio during a call to be
unreliable: often there is no call audio. Reverting this
causes the downlink (bluetooth earpiece) audio to sound
robotic, but the audio is more reliable.
This reverts commit 3772f57d8332e7b2113bd35cd297b8fe00d20d15.
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The DAC widgets are turned off when not in use in order to
save power. They must not be turned off when the output goes
into standby if a call is in progress.
Change-Id: I3d294a85a86e45c8acc257a8d92b92d7d9b2d4c3
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Fixes bug 5223164
Change-Id: I95557589b6d17df96de4235e8003157c6324917a
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Use resampler from libaudioutils instead of speex resampler directly.
This change prepares integration of audio pre processing.
Change-Id: Id985f7e46284fa038f16ecccaaa002b75e375a0f
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- Input volume is set based on capture, voip or voice call modes
Change-Id: I8be69b6ac7a9c34aa27acbf69c42418256e2158d
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Actually, modem and bluetooth devices in tuna use I2S mode.
If channel is setted 1, McBSP is setted PCM mode.
On the other hand, if channel is setted 2, McBSP is setted I2S mode.
To use I2S mode, We have to change channel from 1 to 2.
Change-Id: I0c56ffd03805060783c428b4c70094103480bb4f
Signed-off-by: Lee Min <min47.lee@samsung.com>
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Also increase the codec volume to compensate.
Change-Id: I34bd16141d70cd154df23ff815800bbace887e88
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1. clock sync: after AP (mcbsp2) is enabled
2. audio path: according to AP's output device
Change-Id: I5e0214bea31a722ce43fe92fb1d54bffb291cae9
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The headset volume needs to be reduced for two reasons:
1. Loud volumes trigger headset detection interrupts.
2. The current volume can damage hearing even on low
sensitivity and high impedance headphones.
Change-Id: I639bc8bc1505b6d2f22a8f5581c16583a721770c
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Change-Id: Ia38d9bea3c9abcd1ea505e7302382cb9f6b016c2
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This fixes issue 5099308: AudioManager.isMicrophoneMute() always returns true on Prime.
Change-Id: I7edf7aade2f46725e1fa9685744f21d63a4529a6
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Change-Id: I5b3ef5b111bb4b50fd6d2100f0ed34f47f85cbe3
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Fixes bug 5118167 - when headphones without a mic are connected,
use the main microphone during a voice call.
Change-Id: I846d923d71e88e65adf43001ab13a4d1927a696d
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Using the sub mic is necessary to avoid echo due to the physical
placement of the main mic.
Change-Id: I716db818ec439d812f162b3f4170195c98c51539
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A mixer gain of 118 (-2dB) was used to avoid noise caused by
saturation in the ABE. However, this has been fixed with ABE
firmware 9.46, so these gains can be restored to 0dB for
maximum dynamic range.
Change-Id: Id6a6ae5faeac8673faad3444d5e46e6469d5bd2d
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In the latest ASoC kernel code, the earpiece enable control name
has been renamed. Make the corresponding change in the HAL and
print an error if all controls are not found to aid debugging.
Change-Id: Idc56c383ab9d7b45afad4f54c02237cc4d0db236
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input routing supports switching between MM_UL2/VX_UL
supported capture paths main mic, sub mic, headset mic and BT.
If in call, the input is selected based on the current ouput device.
This also manages the selected device per stream.
Sub-mic capture path not tested.
Change-Id: Ic6da0ef56cfa073b6383fcc389c5ad01a39a7c48
Signed-off-by: Chris Kelly <c-kelly@ti.com>
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Change-Id: I90c957ce7436b52d2aa4339b21d91921e9e612ad
Signed-off-by: Chris Kelly <c-kelly@ti.com>
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The maguro and toro devices have different radios, each
supporting a different maximum volume. Determine the maximum
volume for the device from the ro.config.vc_call_vol_steps
property and use that to calculate the volume to send to
the RIL.
Change-Id: I02921ed41ddbae90f8d3a149c05d37d3e87deab0
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The device lock must be held when using the mixer because
mixer_* calls are not thread safe. This fixes a bug where some
mixer controls including the earpiece volume were not being
set at boot.
Bug: 5073076
Change-Id: Ide060ccad49e7276b7555428d2ff3ab399a5ce40
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The audio output is put into standby during a call so this
causes the earpiece and other output routes not to function
in that state. Reverting until a better fix is ready.
This reverts commit b1695f85e6d4a0baaf8bd3d190d02fe20d537824.
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Now supports capture at 8000, 11025, 16000, 22050, 24000, 32000, 44100 and 48000 Hz.
Change-Id: I61526e94b8f0d315a1bf8d7587363a44c7d643ae
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This should eliminate random clicks and pops heard when audio
enters standby since the output stage is disabled before closing
the PCM. In addition, this should provide a power savings in
standby.
Change-Id: Ief0a193e0b31e9ee2f03a58641eaebd2a0d344cb
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To better support multiple paths and prepare for input
routing, the mixer controls were reorganized into front end
and back end paths for the supported routes. BT-SCO downlink
was also added.
This allows more flexibiity in setting controls but does
sacrifice some amunt of abstraction of the underliying ABE
design.
Change-Id: Ie225ae5bf90b1727178093a5f06636e6b17a737b
Signed-off-by: Chris Kelly <c-kelly@ti.com>
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Change-Id: I48ffc54219360fbb5f22c695dea63ca269e6fb68
Signed-off-by: Chris Kelly <c-kelly@ti.com>
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The RIL needs to have its voice volume reset every time a call
is made, not every time the volume is changed from the upper
layer.
Change-Id: Id042da241de65f9dfb8d5c52e1b4bb910c7c0219
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This prevents a seg fault if no ril is present.
Change-Id: I5f9443e31bdcab07df21d9f12ed2dd92807300f8
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The problem is that the audio HAL returns a NULL string when
get_parameters() is called from AudioFlinger.
It should return return an empty string.
Change-Id: I99365b54eb5f3c3b6694cb3e122842dff1799bfd
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