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* Audio-RIL-InterfacePaul Kocialkowski2014-01-031-4/+4
| | | | Signed-off-by: Paul Kocialkowski <contact@paulk.fr>
* libaudio: add 32000 Hz capture sampling rateEric Laurent2012-06-131-1/+1
| | | | | | Bug 6615379. Change-Id: I344395d707bc842de20e8c9e91136f5b6d19dc3e
* Adjust output to 2 buffers of 880 frames eachGlenn Kasten2012-05-101-3/+3
| | | | Change-Id: I180a4b65a655bcc4ede9f7036d1c9b6daf4c8dbd
* Add USB audio HAL to the buildMike Lockwood2012-04-261-0/+16
| | | | | Change-Id: Id35f2ce611106c8c9ff1f2ef7333fd5c55a0c0b6 Signed-off-by: Mike Lockwood <lockwood@google.com>
* rename audio policy output flagsEric Laurent2012-04-181-2/+2
| | | | Change-Id: Ie1d79d0b7afe9e93f75b134162596ffffdd8a167
* audio policy: added configuration fileEric Laurent2012-03-212-4/+52
| | | | | | | Added audio policy manager configuration file. removed obsolete hasBackMicrophone() method from AudioPolicyManagerBase class. Change-Id: Ib5e5f8b5af9bb0b53a6c0c4f45be374bb8863c92
* Remove obsolete dependency on libmediaGlenn Kasten2012-03-194-4/+1
| | | | | | and unnecessary media includes Change-Id: Ic889aac0e12979d5c5fef6a58ee9917a4864039e
* Merge "Prepare to move system/media"Glenn Kasten2012-03-151-2/+2
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| * Prepare to move system/mediaGlenn Kasten2012-03-141-2/+2
| | | | | | | | Change-Id: Icf444d14be31c81a1018655b9319980f3d5c9f74
* | audio policy manager refactoring.Eric Laurent2012-03-122-8/+0
|/ | | | | | | Removed obsolete method a2dpUsedForSonification() and compilation switch WITH_A2DP. Change-Id: I3df7b81df92f018d6bbc1bff2ec22d1c53e8035e
* Fix build warning on uninitialized field avail_minGlenn Kasten2012-01-201-0/+2
| | | | | | | This struct is on the stack, so either we were lucky and the memory cell happened to be zero, or the value doesn't actual matter for our configuration. Change-Id: If0ac8f1087651c0de98e0e3f787ce6102d3163df
* Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF) DO NOT MERGESteve Block2012-01-081-18/+18
| | | | | | | See https://android-git.corp.google.com/g/#/c/157220 Bug: 5449033 Change-Id: I03a60758c6dad0d9ecbce42f092a0fe757bd7184
* Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF) DO NOT MERGESteve Block2012-01-051-11/+11
| | | | | | | See https://android-git.corp.google.com/g/157065 Bug: 5449033 Change-Id: I6d2e59149c2f007d6ba8f1d2990837a37f712ffe
* Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF) DO NOT MERGESteve Block2012-01-031-21/+21
| | | | | | | See https://android-git.corp.google.com/g/156016 Bug: 5449033 Change-Id: If4249034e0a90a502aba69c199173c8ad4af93b3
* am de21127c: Merge "libaudio: increase audio buffer size" into ics-mr1Eric Laurent2011-11-111-2/+2
|\ | | | | | | | | * commit 'de21127cf8438adbd481978fe874314433a06b30': libaudio: increase audio buffer size
| * libaudio: increase audio buffer sizeEric Laurent2011-11-101-2/+2
| | | | | | | | | | | | | | | | | | | | | | Bring playback and capture buffer size back to their initial size. Current size was dictated by constraints in early voice processing implementation but is not necessary anymore. Increasing buffer size will help in power consumption and limit possible occurence of audio skipping during playback and capture. Change-Id: I7837c62e11700ed5c9a26f52fb27170add09721d
* | Rename LOGV(_IF) to ALOGV(_IF) DO NOT MERGESteve Block2011-10-261-55/+55
|/ | | | | | | See https://android-git.corp.google.com/g/#/c/143865 Bug: 5449033 Change-Id: I73e1ab4f4eadb55e747b3b2be4b6c1824dce7b2c
* audio HWL: removed unused code.Eric Laurent2011-10-104-745/+0
| | | | | | | Removed C++ implementations of echo reference and resampler not needed anymore now that libaudioutils is used. Change-Id: Ibedf96fbaeeb38ea06b35adf7c95ed49cbafa916
* audio HAL: fix some audio mode transitionsEric Laurent2011-09-081-3/+5
| | | | | | | | | The logic controlling the CP clock and audio path in setMode() was incorrect for certain transitions. In particular, the sequence IN_CALL -> RINGTONE -> IN_COMMUNICATION -> IN_CALL would cause a loss of call audio. Change-Id: I4a6bcca32e6e33f965874ad2219f2728b9035e4a
* audio HAL: C interface to libaudioutils.Eric Laurent2011-08-243-92/+142
| | | | | | | | Modifications for new C interface to echo reference and resampler. Also release echo reference if needed when closing an output stream. Change-Id: I9e8b524effad66798a61e80b5fec1779558cce1e
* Create library for common audio HAL functionsEric Laurent2011-08-182-21/+6
| | | | | | | Moved echo reference and resampler code to audio utils library so that they can be shared by other audio HALs. Change-Id: I7ab88843cc58ecc276bc4ccbbdc826c8c0b4a430
* audio HAL: new echo reference designAndy Gough2011-08-124-139/+225
| | | | | | | | The echo reference discards or inserts (silent) frames according to underruns or overruns. It always returns an echo buffer which delay from current capture buffer is 0. Change-Id: Ifea06a47fe87f2b75d4d04737c495a9867d1c4bd
* audio HAL: Update to latest tinyalsaEric Laurent2011-08-011-0/+6
| | | | | | add control for ALSA thresholds in pcm_config structure. Change-Id: I07f6f623b4102bfcb31dd772d93c8443a2d7f236
* Fix issue 4604090: notification sound interrupted.Eric Laurent2011-07-251-10/+0
| | | | | | | Removed wake lock acquisition from audio HAL. This is now done by AudioFlinger via PowerManagerService. Change-Id: I548a35eaf65fb27ea2cbed67cf89bef8f7eff830
* Audio HAL: added echo referenceEric Laurent2011-07-258-222/+1055
| | | | | | | | | | | Added EchoReference class enabling the input stream to access the audio data written to the output stream and push it as echo reference to the AEC. Also added methods to calculate the echo delay. Moved ReSampler class to a separate source file. Change-Id: I9c3388f39101d567240545eab271eb61c97e7b56
* Audio HAL: added audio pre processingEric Laurent2011-07-172-68/+177
| | | | | | Added calls to audio preprocessings for input streams. Change-Id: I1c655005b62b235e5d5d671634227f4a99eee43e
* Merge "Remove the simulator target from all makefiles. Bug: 5010576"Jeff Brown2011-07-121-5/+1
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| * Remove the simulator target from all makefiles.Jeff Brown2011-07-111-5/+1
| | | | | | | | | | | | Bug: 5010576 Change-Id: Ie6fa05f8d72bdf6da3b743a7f8c2464f9febc868
* | Audio HAL: added interface for audio preprocessingEric Laurent2011-07-112-26/+41
|/ | | | Change-Id: I0fcfbbddc9716e5c39b3d78032222c7babaa46bd
* Audio HAL: replaced downsampler by speex resamplerEric Laurent2011-07-053-385/+151
| | | | | | | Use speex resampler for more flexibility and prepare integration of audio pre processing which require upsampling. Change-Id: I7dd234bd89116d028655a043c84c1a18faf3bc67
* Use tinyalsa libraryEric Laurent2011-07-0510-2070/+45
| | | | | | | | | | Audio HAL now uses tinyalsa library to interface to alsa kernel drivers. Removed local thin alsa user space implementation. Also modified value of AUDIO_HW_IN_SAMPLERATE to match actual sampling rate when reading from kernel pcm device. Change-Id: Id0b2d166f3ab2f2291bf49b36c7085b21135ceea
* crespo/audio: build hal modules instead of libaudioDima Zavin2011-04-205-18/+27
| | | | | Change-Id: Ia4134c310a8e854d85c3584907a4da07355cc2b3 Signed-off-by: Dima Zavin <dima@android.com>
* am f127e463: am 2aedd44b: Merge "Fix issue 4126225: setMicMute() execution ↵Eric Laurent2011-03-181-9/+20
|\ | | | | | | | | | | | | time" into gingerbread * commit 'f127e46304c21af95f3e17e70db4581b40d46e02': Fix issue 4126225: setMicMute() execution time
| * Fix issue 4126225: setMicMute() execution timeEric Laurent2011-03-171-9/+20
| | | | | | | | | | | | | | | | | | | | | | | | The cause is that the thread executing setMicMute() can wait for several seconds before acquiring the input thread mutex when audio capture is active. The fix consists in forcing a sleep of the capture thread when standby() (called by setMicMute()) wants the mutex. Applied the same fix to output stream and to setParameters() function. Change-Id: I3e55670d2aad16b67d44ca8582ed16398143ff6e
* | am e8106736: am d7657374: SOUND: set incall volume when user starts call.UK KIM2011-03-052-4/+17
|\ \ | |/ | | | | | | * commit 'e8106736688d60b730873969dba4b60a64cae5e3': SOUND: set incall volume when user starts call.
| * SOUND: set incall volume when user starts call.UK KIM2011-03-032-4/+17
| | | | | | | | | | | | | | | | | | modem side doesn't store volume setting when device is rebooted. When audio mode is incall, set current voice volume to modem. Current incall volume is stored when upper layer sets or AudioService is started. Change-Id: Icfeb0273dda55354e207d464884ef99f41c5fdec Signed-off-by: UK KIM <w0806.kim@samsung.com>
* | am 8164eaba: am d9e21463: Add additional statements to accept the crespo4g ↵Maarten Hooft2011-03-011-1/+1
|\ \ | |/ | | | | | | | | | | product * commit '8164eaba5b4846431997aa545f44bd158916b9e3': Add additional statements to accept the crespo4g product
| * Add additional statements to accept the crespo4g productMaarten Hooft2011-02-281-1/+1
| | | | | | | | Change-Id: I0e410d7b641c19a773c3bc57cb78e8aa3e27a3ff
* | am 2c9d4f75: am 303e0247: Merge "Fix issue 3436738." into gingerbreadEric Laurent2011-02-101-8/+10
|\ \ | |/ | | | | | | * commit '2c9d4f757248cae8ac673930f6ed2126f4ec3362': Fix issue 3436738.
| * Fix issue 3436738.Eric Laurent2011-02-091-8/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The problem is that the AudioPolicyManager refreshes the device attached to the hardware output stream every time another output stops (in this case A2DP output). As there is no active stream on the HW output, the selected device is 0. As the shutter sound is short and there is some additional delay to setup the A2DP output, the sound is not yet out of the AudioFlinger mixer when the device is set to 0. On Crespo, device 0 is a valid audio path (means audio off) which is not the case on other platforms and therefore the output is disabled while the shutter sound is actually playing. The fix consists in ignoring requests to set the output device to 0 in the audio HAL. Change-Id: I7366e359a7d3a0f8207e7a5c879ced9078224002
* | am 4b36da0e: am 962f66ca: Added support for TTYEric Laurent2011-02-022-4/+56
|\ \ | |/ | | | | | | * commit '4b36da0e37942aa0547d2e31731f88b947f48a64': Added support for TTY
| * Added support for TTYEric Laurent2011-02-012-4/+56
| | | | | | | | Change-Id: I60e8ae30da4c3879bdd6a272dd65f6add4d1f520
| * do not merge bug 3370834 Cherrypick from master CL 79417Jean-Michel Trivi2011-01-214-44/+46
| | | | | | | | | | | | | | | | | | | | | | | | Rely on the audio input source parameter for mixer configuration. The generic APM implementation now passes the audio recording source as a configuration parameter, and uses the enum defined in mediarecorder.h. But the driver uses a string to define the "input source state". This change maps the input source to the string used by the driver. Change-Id: I5fce44579a3cda01ed73f67fb8c3091ef05cce76
* | am 92786a1c: am 27cc0c78: Merge "Fix issue 3305305." into gingerbreadEric Laurent2011-01-212-16/+62
|\ \ | |/ | | | | | | * commit '92786a1ce7e7dd078d1a5c05411715d4a617dc57': Fix issue 3305305.
| * Fix issue 3305305.Eric Laurent2011-01-182-16/+62
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The problem is that when the voice search tone is started, audio capture is still active and the output stream write function needs to place the input stream in standby to reconfigure the kernel driver. To do so it has to acquire the input stream mutex but as the input stream thread holds the lock most of the time while sleeping in the driver waiting for more data, this is very difficult and can take several seconds. The fix consists in forcing a sleep in the next read() when another function needs to acquire the input stream lock. The same change is done for output stream write() function. Also removed the workaround for issue 3201189 in setMode() (thread priority bump) as this change addresses the same problem. Change-Id: I3a5e672717752f83dfedce822a18748b165b0a5a
* | am 8e805326: am ee0ab929: Fix issue 3198397Eric Laurent2010-12-011-5/+6
|\ \ | |/ | | | | | | * commit '8e8053263ab53b1200d7204f44c6a480e5459862': Fix issue 3198397
| * Fix issue 3198397Eric Laurent2010-11-301-5/+6
| | | | | | | | | | | | | | | | Use only 15 bits for fractional part of the increment in resample_441_320() to avoid overflow when multiplying by the difference between previous and next samples in interpolated value computation. Change-Id: I9f5a726d11f6b051db390df3d13312f1ee782d3a
* | resolved conflicts for merge of d59093ec to masterEric Laurent2010-12-0115-3327/+4439
|\ \ | |/ | | | | Change-Id: Iab2b74f8eb505e3f474f2d768b1e00964c214526
| * Removed ALSA user space library and utilitiesEric Laurent2010-11-3016-3327/+4441
|/ | | | Change-Id: I0a6492f7c834ea572531e77f75486bcc385e345b
* Fix issue 3119463.Eric Laurent2010-10-221-2/+2
| | | | | | | | | | There is no call audio uplink when headphones w/o mic are connected. This change together with a corresponding change in the kernel driver adds a separate device for headphonjes without mic: output is routed to headphones and input is routed from built-in mic. Change-Id: I19955f76ece19f661ae25d6a42bbcbe235a9e652
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