Squashed commit of cm-10.1 to 2013.06.09

Change-Id: I61fe59d1c3ee8075492041eecc9d90a54ee604a3
Squashed commit of the following:

commit 5b1cb9a1af3a726f3b858b686a2e32fcdda19a38
Merge: e995938 bfafd64
Author: Wizard0f0s <jamie.gilmer@gmail.com>
Date:   Mon Jul 1 23:43:26 2013 -0400

    Merge branch 'cm-10.1' of git://github.com/CyanogenMod/android_hardware_qcom_audio-caf into cm-temp

    Conflicts:
    	alsa_sound/ALSADevice.cpp
    	alsa_sound/Android.mk
    	alsa_sound/AudioHardwareALSA.cpp
    	alsa_sound/AudioHardwareALSA.h
    	alsa_sound/AudioPolicyManagerALSA.cpp
    	alsa_sound/AudioPolicyManagerALSA.h
    	alsa_sound/AudioSessionOut.cpp
    	alsa_sound/AudioUsbALSA.cpp
    	alsa_sound/AudioUsbALSA.h
    	libalsa-intf/msm8960_use_cases.h
    	msm8660/AudioHardware.cpp

    Change-Id: Iac2d92716f947e06e6e02c246c66dcdfc1dfbbeb

commit bfafd64ba687214548cf795221548a10ef850293
Merge: 4fed5f5 c757e34
Author: Daniel Hillenbrand <codeworkx@cyanogenmod.org>
Date:   Fri Jun 28 07:25:36 2013 -0700

    Merge "audio: provide a hook for audio amplifier implementations" into cm-10.1

commit 4fed5f51baea98f68fe49edceda62fb351b99080
Author: Steve Kondik <shade@chemlab.org>
Date:   Wed Jun 26 13:57:34 2013 -0700

    alsa_sound: Add support for VR-on-wired-headset device

    Change-Id: Ib206649194794f46480b7f87eaeb9e3fdb0df1b1

commit e912e3ef4ed5132e16121c17cfb6a99e823195d3
Author: Steve Kondik <shade@chemlab.org>
Date:   Wed Jun 26 12:48:22 2013 -0700

    alsa_sound: Add support for seperated headset mic devices

     * Some recent devices such as the Galaxy S4 use dedicated 3-pole
     * headset mic devices while in-call.
     * This also supports the seperated VOIP headset mic device.
     * Only engage the ES325 dualmic for speaker/earpiece usecases.

    Change-Id: I1c0b01a9df70fbb7364ba69004aa7828e3f1f01c

commit f7b50cad942d2184e3932e02705977cafa813b33
Author: Steve Kondik <shade@chemlab.org>
Date:   Wed Jun 26 00:07:01 2013 -0700

    alsa_sound: Add support for seperated VOIP devices

     * Similar to the VOC devices tuned for voice calls, some new devices
       are using VOIP devices tuned for SIP and Hangouts.
     * Enable with BOARD_USES_SEPERATED_VOIP

    Change-Id: I3a49be26ca2ad3e1aafb8b0960935ac8c735fe51

commit 1dba7824f46c76b791d4b6044b887a075b65ed0f
Author: Mike Grissom <mikeyman77@gmail.com>
Date:   Sun Jun 23 16:04:36 2013 -0700

    msm8660: Fix call/voip audio routing for Samsung audio

    Change-Id: I021344b1c4672b2fb92503059b3eaec6121bdc19

commit ffee8ea27b1998e872378c1783fc6ccd2078a26e
Author: Mike Grissom <mikeyman77@gmail.com>
Date:   Fri Jun 21 10:25:12 2013 -0700

    msm8660: Fix VoIP device for Samsung Audio

    * Add USE_SAMSUNG_VOIP_DEVICE for voip devices
    * Added ifdef for MVS include

    Change-Id: I57b52cc159eb6ef5cadadf8acacffceaf2bc77aa

commit 12bca578ba9edb518127946afdc18c00e5d04737
Author: Preetam Singh Ranawat <apranawat@codeaurora.org>
Date:   Mon Jun 10 22:05:40 2013 +0530

    alsa-sound: Fix to sound burst on switching from speaker to headset

     -During device switch stream mute(vol 0) for compressed or lpa
      stream is getting applied after the device set for primary output
      because first it check for primary output, set the device then it
      check for lpa or tunnel output and mute the stream.

     -Set the stream mute for other output(LPA or Compressed) followed
      by device setting for lpa or compressed output and then set the
      device for primary output.

    CRs-Fixed: 480077

    Change-Id: I6bc1ba5d8df766fbbbffaba4d44f14f7a0be666f

commit 6f82cba6dd1036f6054edf34dd7883bb363b7c51
Author: Chaithanya Krishna Bacharaju <chaithan@codeaurora.org>
Date:   Thu Jun 13 09:48:43 2013 +0530

    alsa_sound: Fix for short break in music when touch tone plays

    -When touch tone plays for the first time while music playback is
    going on, a short break in music happens.
    -In checkDeviceMuteStrategies whenever there is a change in device
    on an output, tempMute is set. However the same tempMute is used to
    decide whether to mute an already existing session on a different output
    irrespective of the current active device on that output. Due to this
    mute and unmute happens even when there is no real change in device on
    a particular output which might cause a short break in playback.
    -Add check to ensure that active streams are temporarily muted only
    in case of a change in device on that particular output.

    CRs-Fixed: 495009
    Change-Id: Icbc2afa4516f700b170c1f6801135cc5ca61fbfe

commit 170d3eb2789c23f1f8aabe08cdc97463ec52ccd9
Merge: 102705c 6479193
Author: Steve Kondik <shade@chemlab.org>
Date:   Tue Jun 18 19:27:28 2013 -0700

    Merge tag 'AU_LINUX_ANDROID_JB_2.5.04.02.02.040.375' of git://codeaurora.org/platform/hardware/qcom/audio into cm-10.1

    AU_LINUX_ANDROID_JB_2.5.04.02.02.040.375 based on quic/aosp/jb_2.5

commit 102705c1d290e21351f4bafad4d46669326bb4eb
Merge: a1e3e66 49d7909
Author: Steve Kondik <shade@chemlab.org>
Date:   Tue Jun 18 19:24:30 2013 -0700

    Merge branch 'av-test' of git://github.com/CyanogenMod/android_hardware_qcom_audio-caf into cm-10.1

    Change-Id: Ib07512ab88052716106383f321ddaad0e0586fb1

commit c757e34b94959982cef3bdaae7979daf4a8dd2ae
Author: Daniel Hillenbrand <codeworkx@cyanogenmod.org>
Date:   Thu May 23 06:40:42 2013 +0200

    audio: provide a hook for audio amplifier implementations

    Example: http://review.cyanogenmod.org/#/c/38221

    Change-Id: Ib236598a5888b2af19bcfb81e285f644a0e84c0d

commit 49d7909d104f8102a3b0e9726f5a1e6844e75ffe
Merge: 9dd43ae 3130937
Author: Giulio Cervera <giulio.cervera@cyanogenmod.org>
Date:   Thu Jun 13 10:50:28 2013 +0200

    Merge remote-tracking branch 'caf/jb_2.5' into HEAD

    Conflicts:
    	alsa_sound/ALSADevice.cpp

    Change-Id: Ibc60d3694d814a4ad8ad464e95217ccca8b31936

commit 647919337208834a2c7f80d891faae19a90ce097
Merge: 062aece a0c61a4
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Jun 12 22:16:00 2013 -0700

    Merge "alsa_sound: Add EC supported Key"

commit a0c61a4fd72eb09c1da6fe6493ca3b994be5af51
Author: Vidyakumar Athota <vathota@codeaurora.org>
Date:   Mon Jun 10 19:11:43 2013 -0700

    alsa_sound: Add EC supported Key

    Add echo cancellation key to HAL getparameters

    Change-Id: Ic1935c0e245a30f9bcc8939e61e58201cb2774e3

commit a1e3e662cf6e3da011458332f7e7544313296d99
Merge: d347e37 afbe38a
Author: Ricardo Cerqueira <cyanogenmod@cerqueira.org>
Date:   Tue Jun 11 04:21:41 2013 -0700

    Merge "alsa_sound: change noisy debug statements to verbose" into cm-10.1

commit 062aecefb8679acd9536983f97ec945c9a320d9a
Merge: cb0bad7 9e7657f
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Jun 10 04:21:21 2013 -0700

    Merge "  alsa_sound: Patch to fix Voice call setup"

commit cb0bad7bbb6cec448a9558c188272311ad634dfc
Merge: 3130937 fd5ea63
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Jun 10 04:21:14 2013 -0700

    Merge "alsa_sound: Get FM volume control from UCM"

commit 3130937e67504a1171bd0bee9f98dbf8e13ec729
Merge: 9c5c09c 18e59b2
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Fri Jun 7 12:36:50 2013 -0700

    Merge "alsa_sound: Add initial volume setting for USB headset"

commit 9c5c09cdb9720722c2b8d2d35ccf073bc4f747b3
Merge: 45277f9 6697adb
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Jun 6 04:58:59 2013 -0700

    Merge "Revert "audio: getNewDevice on primaryOutput for strategy phone and enforced audio""

commit 45277f93c1d7ee0ccde4b2a5c3c6ffb99d2c4353
Merge: 12b58f1 e5b5bea
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Jun 5 20:10:58 2013 -0700

    Merge "alsa_sound: Fix for IMS hold issue"

commit 12b58f1bd563f237f6c8b5fbf0f297996559953b
Merge: 6c8d9ca d5fc66e
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Jun 5 17:19:07 2013 -0700

    Merge "alsa_sound: Fix for sound burst for short duration"

commit 6c8d9cafa13344a83be7afde94212c904bb45f55
Merge: ed22975 ca70c8b
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Jun 4 12:56:23 2013 -0700

    Merge "alsa_sound: Increase the stop threshold of proxy driver for a2dp"

commit ed229752f28a5af9954f6c6912da0a8a08d159a1
Merge: 14567ba f2a261f
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Jun 3 23:27:35 2013 -0700

    Merge " alsa_sound: Fix to avoid mute of an already muted stream."

commit 9e7657fc20fd06d040104f79040541d179f953cf
Author: Sidipotu Ashok <sashok@codeaurora.org>
Date:   Thu Mar 28 11:44:44 2013 +0530

      alsa_sound: Patch to fix Voice call setup

      - When BT is in A2DP profile, try to do a Voice call,
        unable to hear any voice on BT/earpiece/Speaker when in call.
      - enablevoicecall() is failed due to BTA2DP is not supported
        in voicecall, later that no trials to setup voice call path
        from HAL side even with the proper device.
      - Added check to detect voice path setup failures, such that
        later requests to setup voice path can be processed.

      CRs-fixed: 462159

    Change-Id: I8609983c785522879a13120c86b8984a5853f9d7

commit f2a261f64056c8b8386c83a6a35549202aaa9ad8
Author: Srikanth Katta <skatta@codeaurora.org>
Date:   Wed May 15 17:47:11 2013 +0530

     alsa_sound: Fix to avoid mute of an already muted stream.

    -Sound is not audible for a video file in the following
    scenario: play a video file, receive an alarm notification,
    connect wired headset and BT headset, and now remove BT
    headset and wired headset, dismiss alarm and resume video
    playback. Video playback continues with out any audio.

    -In the above scenario, when we connect wired headset all
    streams on strategy media gets muted (as media strategy takes
    in wired headset as device and alarm considers combo device
    speaker+headset as new device) and mMuteCount gets incremented as
    its a device switch, similarly when BT headset is connected, all
    media streams are muted again and mMuteCount is incremented. And
    when BT headset is removed mMuteCount gets incremented again as its
    a device switch from BT to wired headset, and mMuteCount is incremented
    accordingly. Finally when wired headset is disconnected, media streams
    are unmuted and mMuteCount is decremented. However effectively
    mMuteCount is still greater than zero because of which all media
    streams in non-LPA mode are muted.

    -Avoiding mute of an already muted stream resolves this.

    CRs-Fixed: 482262

    Change-Id: I204d434fd7b0abd0fff6afcb6de0250f4e86ba21

commit d347e3765c0cf28a0b733f2bf99d78be042c5549
Author: ehgrace.kim <ehgrace.kim@lge.com>
Date:   Mon Jun 3 11:47:54 2013 +0200

    audio: modify the SPEAKER_RX_ACDB to support both MONO and STEREO speaker

    Change-Id: Iea7d8d5cdb7b04b347be84343f5bd44caeb51609

commit 9dd43ae5494694dd13a39cdac7fef6e8ed8251e3
Author: ehgrace.kim <ehgrace.kim@lge.com>
Date:   Mon Jun 3 11:47:54 2013 +0200

    audio: modify the SPEAKER_RX_ACDB to support both MONO and STEREO speaker

    Change-Id: Iea7d8d5cdb7b04b347be84343f5bd44caeb51609

commit 8e548a395fdfca2183ae9304710bc5a2796ce671
Author: Giulio Cervera <giulio.cervera@cyanogenmod.org>
Date:   Mon Jun 3 10:52:18 2013 +0200

    audio-caf: fix dbg message derp

    Change-Id: Id64e7c8fa04cafa57f422558c5aa84a21127b88f

commit 07197326b745c6df42178dad7e2ea66d1f6ea361
Merge: f3f7197 75e2a91
Author: Daniel Hillenbrand <codeworkx@cyanogenmod.org>
Date:   Sun Jun 2 07:05:50 2013 -0700

    Merge "alsa_sound: add HTC CSD CLIENT board flag for special cases (BT)" into cm-10.1

commit 4027a1eb3dba9937afcf2cdd3cba1fbc7d7a39d8
Author: Steve Kondik <shade@chemlab.org>
Date:   Sat Jun 1 20:34:23 2013 -0700

    alsa_sound: Kill LPA logspam

    Change-Id: I61e4e0500911865d2a119f552cc97c8bba99c3b8

commit 75e2a910038e37c2252f305c61943dbf3a18c7d9
Author: tbalden <illespal@gmail.com>
Date:   Sat Jun 1 21:10:12 2013 +0200

    alsa_sound: add HTC CSD CLIENT board flag for special cases (BT)

    New HTC phones like m7 have special settings for BTSco CSD device
    ids. Adding board flag BOARD_HAVE_HTC_CSDCLIENT to handle
    these special cases related to HTC's csd client.

    Change-Id: Iae3cb4568c0d1d6b5d75cef7a836d78ab3fa9522

commit f3f7197c9f6a2fb16577d98f60a04e6c352718ee
Merge: eeb9f79 b37186b
Author: Giulio Cervera <giulio.cervera@cyanogenmod.org>
Date:   Sat Jun 1 11:02:36 2013 -0700

    Merge "Revert "alsa_sound: Add metadata mode changes for LPA playback"" into cm-10.1

commit b37186b99dd948699bb29214bb421acee8bd35ab
Author: David van Tonder <david.vantonder@gmail.com>
Date:   Sat Jun 1 11:01:34 2013 -0700

    Revert "alsa_sound: Add metadata mode changes for LPA playback"

    This reverts commit 5496a3c345320c4340820daf06333828040418a9

    Change-Id: Id26b862abb362a481afbd6af58dc6d932aa0901e

commit eeb9f79a5f3abcefaa7316105a1366ac725f2282
Merge: 32352f0 5496a3c
Author: Giulio Cervera <giulio.cervera@cyanogenmod.org>
Date:   Sat Jun 1 07:46:22 2013 -0700

    Merge "alsa_sound: Add metadata mode changes for LPA playback" into cm-10.1

commit 775f9d7382ecf16591ef5d7bc43c10c8bb9a4602
Merge: 32352f0 bc5b4a0
Author: Steve Kondik <shade@chemlab.org>
Date:   Sat Jun 1 02:14:02 2013 -0700

    Merge branch 'jb_2.5' of git://codeaurora.org/platform/hardware/qcom/audio into cm-10.1

    Change-Id: I7cbe046f187b2486132d375141ba96510d516c32

commit 32352f0c903b7e8fe4ed3ea549b271a032e4ea4f
Author: Steve Kondik <shade@chemlab.org>
Date:   Fri May 31 16:27:31 2013 -0700

    alsa_sound: Support ES325 2Mic and Samsung CSD client

     * The dual mic feature on the Audience ES325 needs to be enabled via
       csd-client on the Galaxy S4. Add support for this.
     * Samsung is using a non-standard name for the functions in csd-client.
       Add support for this.

    Change-Id: I764ead9ea17729055da701a423edec97d96b8319

commit 18e59b2c19134b32f46399592ff818cbaa3870af
Author: Deepa Madiregama <dmadireg@codeaurora.org>
Date:   Wed May 1 13:27:53 2013 +0530

    alsa_sound: Add initial volume setting for USB headset

    Some USB headsets start at 0 volume because of this even
    if the Media Player volume is maximum cannot hear any audio.
    Initialize device volume before starting the playback.

    CRs-Fixed: 478906
    Change-Id: I8d38325ac9ba2d48c2be3fa627efde86e7a9d33c

commit 14567ba659bea928ebba0055f6695ae472b433ae
Merge: 7f5bd97 14b98b0
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu May 30 01:56:53 2013 -0700

    Merge "  alsa_sound: Fix to route Skype call through HDMI."

commit 7f5bd977b6531e32bf6ef9ff1ae4c00d96ed4412
Merge: bc5b4a0 b443bf1
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu May 30 01:56:51 2013 -0700

    Merge "  alsa_sound: Fix to avoid Ringtone leaks to speaker in Skype MO call"

commit e5b5bea4bc894d43b8d76db6783ca6c1c160cc7c
Author: Vicky Sehrawat <vickys@codeaurora.org>
Date:   Wed May 29 18:09:54 2013 -0700

    alsa_sound: Fix for IMS hold issue

    - doRouting is not called for going from
      IMS hold to IMS active. Hence, there is
      no audio.
    - Add check for hold state also along with
      active to call doRouting.

    Change-Id: Idc894dbad1ea14b3afff6062cff411108bfeced7
    CRs-fixed: 493474

commit 6abae3d2a4d9f95084c2749a4ab84067648df8d8
Author: Steve Kondik <shade@chemlab.org>
Date:   Mon May 27 14:35:24 2013 -0700

    msm8660: Fix mic in Hangouts

     * Add higher sample rates and stereo input to VOIP device

    Change-Id: I7b2a439325d1b427cb33b16f3ab5473edc352833

commit d5fc66e5869473d1b59ac4b3e9acc25dac6c0225
Author: Yamit Mehta <ymehta@codeaurora.org>
Date:   Tue May 14 12:36:09 2013 +0530

    alsa_sound: Fix for sound burst for short duration

    -Audio data volume is high for some time when TV is switch off and on
     when HDMI is connected.
    -The delay calculation for unmute is not correct.
    -Correct the calculation for unmute.

    CRs-Fixed: 486123
    Change-Id: Id3360a092b9d3803c6679ec7e455c96ae123ec8c

commit 6697adb772689e48bb77dbd39759f939106ba5fb
Author: Vicky Sehrawat <vickys@codeaurora.org>
Date:   Tue Apr 30 16:58:06 2013 -0700

    Revert "audio: getNewDevice on primaryOutput for strategy phone and enforced audio"

    This reverts commit 694d88c26e0c778fb27b4a389dc6c59fab54b2f6.

    Conflicts:

    	alsa_sound/AudioPolicyManagerALSA.cpp

    CRs-Fixed: 482213
    Change-Id: I54b54838efa0f714c070dc7118bcddf76e02debb

commit 14b98b06d029a56f8be62ea8738542012b4f39a2
Author: Sidipotu Ashok <sashok@codeaurora.org>
Date:   Wed May 22 20:08:51 2013 +0530

      alsa_sound: Fix to route Skype call through HDMI.

      - Observed that Skype call is not routing through HDMI sometimes.

      - HDMI device pair gets overwritten by Earpiece & Mic when Skype
        closes current input stream and opens another one.

      - Fix is to check the current rx device in audio HAL and picking up
        proper rx instead of handset_rx.

        CRs-Fixed: 478276

    Change-Id: Ie0c470d38b2d5cbcdec5616ad684d353afd34a59

commit 5496a3c345320c4340820daf06333828040418a9
Author: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
Date:   Thu Mar 14 09:57:57 2013 -0700

    alsa_sound: Add metadata mode changes for LPA playback

    -Seek to EOS was causing playback to hang for 3 seconds before
     switching to the next clip.
    -This is because the lpa driver works on period size. Partial
     buffers are not handled.
    -Add metadata mode changes to AudioSessionout for LPA playback to
     support partial frames.

    CRs-Fixed: 458904
    Change-Id: Ibe3225d50e1c4248ba5c41e8d2a7067fc20108d9

commit b443bf1fe9547307e9bd8593e496f32746287cb8
Author: Sidipotu Ashok <sashok@codeaurora.org>
Date:   Wed May 22 19:12:01 2013 +0530

      alsa_sound: Fix to avoid Ringtone leaks to speaker in Skype MO call

      - Observed Ringtone leaks to speaker sometimes during VoIP MO call
        end.

      - Skype is not sending stopOutput() on AUDIO_STREAM_RING immediately
        after sending stopOutput() on AUDIO_STREAM_VOICE_CALL.

      - Fix is to "Mute the AUDIO_STREAM_RING if HAL gets stopOutput() on
        AUDIO_STREAM_VOICE_CALL and there is a AUDIO_STREAM_RING still
        exists in the system.

        CRs-Fixed: 477665

    Change-Id: I4f964a779a0e15bd982b8ee634b1549109f0f304

commit 802b1d3bcdcb99f30b3678fab2c337dc11bc5ad0
Author: Karthik Reddy Katta <akkatta@codeaurora.org>
Date:   Fri Mar 22 19:04:07 2013 +0530

    alsa_sound : Fix for audio routing speaker though BT is connected

    - Speaker device is selected by getDeviceForStrategy() though BT is
      connected as mA2dpSuspended is still remaining as 'true' even
      after ending the voice call.
    - This flag is not updated by checkA2dpSuspend() as force usage
      for communication and record is still FORCE_BT_SCO after ending
      voice call.
    - Modified condition check to set mA2dpSuspended to 'false' as soon
      as phone state is changed to MODE_NORMAL

    Change-Id: Ibc0d19b361755d8715c446401d2e14d914fd7020
    CRs-Fixed: 464404

commit 2d3405ac255f6feae902d6febe9d33b87fd35d4d
Author: Yamit Mehta <ymehta@codeaurora.org>
Date:   Fri Apr 5 17:07:33 2013 +0530

    alsa_sound: Increase delay in temp mute

    -Silence for small time when headset is inserted
     and then touch tone is played.
    -Stream is unmuted in checkDeviceMuteStrategies before
     device switch occurs.
    -Increase delay in temp mute to ensure unmute after
     device switch.

    CRs-Fixed: 470160
    Change-Id: I6bc443abf77e5600675846086a57090f49bf9bd7

commit 4780440d44ae9add1b176a5833ab9e0ec708d9bc
Author: Sharad Sangle <assangle@codeaurora.org>
Date:   Thu Apr 25 18:34:11 2013 +0530

    alsa_audio:remove the device from device list if open fails

    -Opening second record session is failed,which is
    expected but  after that user is no more able to
    open record session
    -The device is already got added in alsa_handle after
    calling mALSADevice->open but if this open fails, need
    to be deleted from device list.
    -If mALSADevice->open is failed then erase the device from from
    mDeviceList so that next time it can opened.

    Change-Id: I9847f65718d8330292f312dffe18dfcb672f2d3d

commit 044c432d3e1d75e9a6462acfa18884bf4db0a853
Author: Preetam Singh Ranawat <apranawat@codeaurora.org>
Date:   Mon Jan 28 11:13:13 2013 +0530

    hardware/qcom/audio: Fix for Crash during Gtalk startup

     -Crash is observed when Hangouts and Gtalk start running
      simultaneously
     -Dirver does not support multiple instances of voip call.
     -Avoid creating multiple voip session if mVoipStreamCount
      is already 2
    CRs-Fixed: 440160
    (cherry picked from commit b9f1299e4ca8b96907f97914a0cf65fd5a3d3174)

    Change-Id: I7ef3cf76394e1469db9d21a1484aacf1251c4f25

commit cfc1430b045caeeeacf7520f6eac1f2dde4e6a02
Author: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
Date:   Wed Mar 27 18:31:45 2013 -0700

    alsa_sound: Fix corruption in recording over usb.

    -Sometimes the hardware pointer for proxy driver is not updated for
     successive callbacks. This is because in the recording case proxy
     driver and usb driver runs at real time. This does not mandate
     that the data is read by proxy driver as soon as data is written
     from userspace. Any delay in getting dat from usb also can cause
     the issue.
    -Fix is to increase the start threshold of proxy side for recording
     so that playback is not started without sufficient data.

    Change-Id: I86384bfc96b8262309a1402ca9fe5744af3b436d
    CRs-Fixed: 465317

commit c62d5715cadbfd3a208cc5d4d4a4166bd0c1fde6
Author: Vimal Puthanveed <vimalp@codeaurora.org>
Date:   Fri Mar 22 13:57:26 2013 -0700

    Change to handle A2dp suspend setparameter, which will suspend the thread

    When A2dp suspend setparameter is called we were not suspending the thread.
    We were suspending it only during stop. This will make sure we suspend the
    thread even during the suspend setparameter call

    CRs-Fixed: 415399
    Change-Id: I49ae25fc028e9d4927fd3d3131c8f97f3e873fc1

commit 12457d6fd0b05e06f1394f7e9a239bb42253b67c
Author: Sidipotu Ashok <sashok@codeaurora.org>
Date:   Tue Mar 5 17:11:55 2013 +0530

    alsa_sound: Patch to remove delay in second MT call during hold

    -When voice call is on hold, when another MT call comes, the ringtone
     is delayed by 6sec.
    -This is a special case for fusion, when the voice call is on hold and
     there is another incoming MT call, telephony will set MODE_IN_RINGTONE
     During this there is a change in device from Handset to Speaker,
     this is not propagated to MDM.
    -Fix is to propagate device change to MDM during MODE_IN_RINGTONE
     if there is a call on hold already.

    CRs-Fixed: 447560

    Change-Id: Ia29729bc97a585bd45657e558d04374104e1ce9c

commit 54f6b5260d5da5cbece8d2b4e1668e4d3d7e4171
Author: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
Date:   Wed Mar 20 19:15:06 2013 -0700

    alsa_sound: Fix for audio loss on switching from BT to AOA

    -When AOA is connected to device while playback is on BT, usb stream is
     created. The external out stream gets updated with usb stream even
     when a2dp stream is active.
    -Update external out stream with a2dp or usb stream in openExtOut only
     if it does not refer to any other stream.

    Change-Id: I625ee20ac91ef1be4b21e91fd5e9fffc54a08033
    CRs-Fixed: 464457

commit fddcccaf66f30d76fa26dd5550fd9abeb55b0924
Author: Jitendra Singh Naruka <jnaruk@codeaurora.org>
Date:   Wed Mar 27 18:35:36 2013 -0700

    alsa_sound: set ALSADevice::mInChannels to 0 after voice call finishes

    When a voice call is established, through  enableVoiceCall()
    mInChannels is set to non-zero value, so during disableVoiceCall()
    it should be reset to zero.

    Change-Id: Ic3f39990c7e1820e5c61e063b8c84e160cdd20bb
    CRs-Fixed: 467775

commit ed775c1c81dc6062d0be36d1c5a94df5a83d52eb
Author: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
Date:   Tue Mar 26 20:39:53 2013 -0700

    alsa_sound: Fix for Tunnelplayer IOCTL drain block

    -IOCTL drain was getting stuck when standby was
    called after seeking to EOS
    -This happens when seek was done to the
    end of the stream which would lead to only
    a zero length buffer being sent.
    In this case, IOCTL drain will block forever
    -To solve this, IOCTL drain was interrupted
    by calling flush from the context of standby

    CRs-Fixed: 467166

    Change-Id: I0d119c362670fd4f99b023dd4a1fdebde7832c02

commit bc5b4a03c5ace773a8cad86095f287879fc1a95c
Merge: 03e6235 ae249db
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Sat May 18 21:07:21 2013 -0700

    Merge "alsa_sound: Fix for IMS SRVCC and BT issue"

commit ae249dbca0e89bba83286bba9d3cf11980ed7b39
Author: Vicky Sehrawat <vickys@codeaurora.org>
Date:   Fri May 3 11:40:56 2013 -0700

    alsa_sound: Fix for IMS SRVCC and BT issue

    - Add a check in setParameters to better handle
      IMS call state change.
    - doRouting needs to be called when there is a
      call already active and not when a new call
      is started.
    - As doRouting is needed for SRVCC and not when
      we start an IMS call, as it results in an extra
      dorouting call which causes issues for BT device.

    CRs-fixed: 485997
    Change-Id: I3cf975827d908395d7eb5b2ef3af79cc76452721

commit fd5ea6314e9e5ba2f537279278b25f38cb1be4b1
Author: Damir Didjusto <damird@codeaurora.org>
Date:   Fri May 3 09:53:32 2013 -0700

    alsa_sound: Get FM volume control from UCM

    Read the FM volume control name from UCM and use it when
    setting the FM volume.

    Change-Id: Ieaf92c027d82cfd0e1ca50de371fe4492acf5211

commit 03e62359c952342705038b32acd595a54bb63422
Merge: 42e945f 4f0f3bb
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed May 15 06:30:11 2013 -0700

    Merge "alsa-sound: Add NULL check for mExtOutStream in switchExtOut"

commit 42e945faa8c87dc04e6b5bc357dc213312c9d9d8
Merge: 661c5ca 070ef6d
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed May 15 06:30:08 2013 -0700

    Merge "alsa_sound: Fix to enable wired headset mic during VoIP call."

commit afbe38af83abb018933764ffd5a8570dd0395bd7
Author: Ethan Chen <intervigil@gmail.com>
Date:   Tue May 14 15:41:24 2013 -0700

    alsa_sound: change noisy debug statements to verbose

    Change-Id: Iaa1f0603d9efc1b4593c61b0558a2369ac77d7dc

commit 0177414afec9498a5cbf3e10502b48e3a13c2238
Author: freexperia <jerpelea@gmail.com>
Date:   Wed May 30 09:30:10 2012 +0300

    audio/msm7x30: fix for SEMC devices with stereo hw speaker

    Signed-off-by: Michael Bestas <mikeioannina@gmail.com>

    Change-Id: I2a326b13101bfe589eebe46012559633d8b361df

commit 661c5caf07a556cb9b7766f5f59f150b68b4b6c2
Merge: e0bc583 8f883e2
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu May 9 11:17:47 2013 -0700

    Merge "alsa_sound: Avoid routing device 0 to Audio HAL"

commit 4f0f3bb062635672da5ec156f3201242969feec4
Author: Satya Krishna Pindiproli <satyak@codeaurora.org>
Date:   Thu May 9 16:54:56 2013 +0530

    alsa-sound: Add NULL check for mExtOutStream in switchExtOut

    - When BT is switched off, closeExtOutput() and closeA2dpOutput()
      are called which set mA2dpStream and mExtOutStream to NULL.
      Later when switchExtOut() is called, it results in a crash.
    - In switchExtOut(), there is a dereference of a NULL pointer
      which causes the crash.
    - Avoid the NULL pointer dereference.

    CRs-Fixed: 484724

    Change-Id: Ie37647aa004947be195fa9e0fccf09091d7ceef7

commit e0bc583e91434d712c2cce47f10ae4557056e40e
Merge: a245237 6771433
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed May 8 05:35:03 2013 -0700

    Merge "alsa_sound : Fix for audio routing speaker though BT is connected"

commit 070ef6df1ee15eccc1b103f5328167572109f230
Author: Srikanth Katta <skatta@codeaurora.org>
Date:   Wed May 8 16:43:26 2013 +0530

    alsa_sound: Fix to enable wired headset mic during VoIP call.

    -When in gtalk voice call with wired headset, voice call is routed
    to ear piece in stead of wired headset.

    -In gtalk voice call starts over wired headset, in setParameters(),
    by default DEVICE_BUILTIN_MIC was getting selected, because of which
    Rx routes to ear piece instead of headset.

    -Appropriate input (Tx)/output (Rx) device is now selected based on
    the device connected. Also there was a bug in parsing KeyValuePair
    in HAL, because of which by default DEVICE_BUILTIN_MIC was getting
    selected.

    CRs-Fixed: 483752

    Change-Id: I56d59c53ab1987b185147156fb11757c28c298d1

commit ca70c8b6f75bbc3d73de7d64c06ec42867458458
Author: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
Date:   Mon May 6 10:49:11 2013 -0700

    alsa_sound: Increase the stop threshold of proxy driver for a2dp

    -The stop threshold for proxy port for bt playback is buffer size.
     So if data is not read from proxy for duration of the buffer size
     alsa kernel framework issue a stop . This results in the fd being
     invalid. subsequently playback over bt is stopped.
    -Increase the stop threshold to a higher value so that stop is not
     issued while bt hal is not ready to consume data.

    Change-Id: I4e41186ee0b4c889a092fd696e6c0fd1ffeda354

commit a2452371bbbe2e5d3cfa70850904fe3fd8aac9ae
Merge: 2e65630 221f374
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon May 6 14:48:48 2013 -0700

    Merge "alsa_sound: Enable independent channel volume control"

commit 2e6563063b4732d579e60bc5d870a73357891dd2
Merge: 4fb820d b60faf2
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon May 6 14:47:46 2013 -0700

    Merge "alsa_audio:remove the device from device list if open fails"

commit 0c85f77350e1c1484d6aa0be46a37dcf442fac66
Author: Arne Coucheron <arco68@gmail.com>
Date:   Tue Apr 23 12:18:53 2013 +0200

    audio/msm7x30: #ifdef teh things

        BOARD_HAVE_QCOM_FM
        BOARD_USES_QCOM_AUDIO_LPA
        BOARD_USES_QCOM_AUDIO_SPEECH
        BOARD_USES_QCOM_AUDIO_VOIPMUTE
        BOARD_USES_QCOM_AUDIO_RESETALL
        WITH_QCOM_VOIP_OVER_MVS

    Change-Id: Ia3965096d36768468808b8e3f2ff34318e6aec89

commit 0de404b0aa95dbe9cf30b8457e93c407ba4b12a6
Author: Arne Coucheron <arco68@gmail.com>
Date:   Tue Apr 23 11:42:42 2013 +0200

    Revert "audio/8655: Enabling LPA playback by adding AudioSessionOutLPA class"

    This hasn't worked on 7x30 since ICS, and it also requires ION which
    most 7x30 devices can not use.

    This reverts commit b982f5a7b665873688c6710d53b4f3902a4c2876.

    Change-Id: If1f4ba6fbca611baca220e417a4ef5abd8c1c216

commit d501a1a2c295308b3fcc46fdc7d84165358dbd39
Author: Tiziano Carotti <tizianoc@fastwebnet.it>
Date:   Sun Apr 21 05:06:46 2013 +0200

    audio/msm7x30: fix a crash on audio initialize

    Fixes crash on mediaserver if the device_count got from
    audioalsa library is greater than DEVICE_COUNT in HAL

    Change-Id: Ic5a29ddbee4f7b79ec944e6b48ef8272177659e6

commit 1fdcc6ace5d5606dd6b067443252f62ab3e1c6d2
Author: Giulio Cervera <giulio.cervera@gmail.com>
Date:   Sun Apr 21 05:04:28 2013 +0200

    audio/msm7x30: Change fallback route to speaker

    * Add explict case for HANDSET, and use SPEAKER as fallback

    Change-Id: Ie6dd33f9acb7fd8e5536220e8230822b534798da

commit f0ec98dc09cc9dc2c6dd43fea4f032eb69dd8181
Author: Giulio Cervera <giulio.cervera@gmail.com>
Date:   Sun Apr 21 05:01:48 2013 +0200

    audio/msm7x30: fix voice volume on first call

    - Store float voice volume got from audioflinger even if
      msm_set_voice_rx_vol return an error and re-use it on next call.
    - Remove redundant volume set.

    Change-Id: I1653152d81add785f8a7eee148bf849fecacc841

commit 223cef21f3a47374acfdd08d10f0905e804078ff
Author: Ethan Chen <intervigil@gmail.com>
Date:   Sun Apr 21 00:58:16 2013 +0200

    audio/msm7x30: set mRecordState in AudioInputStreamMSM72xx::set

    This fixes audio input/recording, including voice dialer and
    voice search.

    Change-Id: Ica042715ffc7ee664507951f6f7591b23532c34c

commit 1844bcd7a02a2794786e5c1f2630b261dde9939b
Author: Giulio Cervera <giulio.cervera@gmail.com>
Date:   Sat Apr 20 08:09:30 2013 +0200

    audio/msm7x30: device enum debug

    Change-Id: I3490e21b4177b9e30e6dea548728679001247fce

commit c42eabbeab8e900adc5f307b7833150078b1affa
Author: Ethan Chen <intervigil@gmail.com>
Date:   Sat Apr 20 07:51:15 2013 +0200

    audio/msm7x30: perform null checks when calling libhtc_acoustic functions

    Fixes null pointer dereference when msm72xx_set_audpre_params and
    msm72xx_enable_audpre are called from libhtc_acoustic.so that does
    not have those functions (such as those for msm8x60 platform).

    Change-Id: Ifa4357835e938b374e09908c34c14c8c5c52ce10

commit 548b319cde20b154f5206bf42b402d50c99ecf87
Author: Giulio Cervera <giulio.cervera@gmail.com>
Date:   Sat Apr 20 07:44:26 2013 +0200

    audio/msm7x30: port forwarding HTC code from gb

    Change-Id: I89f0ad806a77c1368e10495287f6693ecb9282ba

commit 8f883e2a7d41da991dae0c7e12773ffda58a5a55
Author: Yamit Mehta <ymehta@codeaurora.org>
Date:   Mon May 6 09:53:55 2013 +0530

    alsa_sound: Avoid routing device 0 to Audio HAL

    -When BT headset, HDMI and wired headset are
     connected and wired headset is removed, playback
     of clip pauses.
    -Device 0 is sent to Audio HAL. Audio rendering is
     stops as it is not a valid device.
    -Correct the condition to avoid device 0 being
     sent to Audio HAL.

    CRs-Fixed: 471200

    Change-Id: I9dd169bb0c3d79791de8f653eae6281e3ed6d1b4

commit 341c8507f0f51d2513a576186d2d575957342af0
Author: Steve Kondik <shade@chemlab.org>
Date:   Sat May 4 01:55:32 2013 -0700

    alsa_sound: Disable verbose logging

     * Kill the logspam.

    Change-Id: I121cddb04c613e03a95b751b7c2fc86a46b89bd4

commit 764f8a9ff26d341f219f7f9a248e11eec24fbfec
Merge: 9a6c780 5accf62
Author: Giulio Cervera <giulio.cervera@cyanogenmod.org>
Date:   Sat May 4 00:19:15 2013 -0700

    Merge "audio: fix for device not disabled after ending voice call" into cm-10.1

commit 4fb820de684c2c9a9dcc536541670002baa8830a
Merge: 7cead94 fea0ec1
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Fri May 3 19:40:43 2013 -0700

    Merge "alsa_sound: Fix recording synchronization issue"

commit 9a6c78064fb3c355f122217d874b066a881e65ac
Author: Giulio Cervera <giulio.cervera@cyanogenmod.org>
Date:   Thu May 2 01:51:16 2013 +0200

    audio: conditionally restore session less CSD apis

    commit c2e34a73f2ce198009448ac278848c7e1596e976 introduced session
    id based CSD apis. As it is introduced in A8064AAAAANLYA161030A
    and all our libcsd-client was released before that, older api is
    selected by default.

    lucky user with newer client can re-enabled it with
      BOARD_HAVE_NEW_QCOM_CSDCLIENT := true

    Change-Id: Ia2f882095126efb239ef94f785cbcea4e379798e

commit 07eb3c2b60fc6081ceaa9c4f1498f20f4e0ac135
Author: Steve Kondik <shade@chemlab.org>
Date:   Fri May 3 18:15:25 2013 -0700

    alsa_sound: Add option to use seperated speakerphone device

     * Enable use of this with BOARD_USEDS_SEPERATED_VOICE_SPEAKER

    Change-Id: I3fb8f8c97c4116ce9717165878d63483e2facb06

commit 221f3744fbfc3bf34877234013a8396e6b8378b5
Author: Jay Wang <jaywang@codeaurora.org>
Date:   Fri May 3 16:16:05 2013 -0700

    alsa_sound: Enable independent channel volume control

    Enable independent channel volume control of LPA & Tunnel stream
    for MSM8960, APQ8064 and MSM8930 targets

    CRs-Fixed: 449284
    Change-Id: Ia9f75b44329e9ccf5b185644f89a7bde6cb0a6a1

commit 6771433e001755e1c491e19d7a255700fdea06fa
Author: Karthik Reddy Katta <akkatta@codeaurora.org>
Date:   Fri Mar 22 19:04:07 2013 +0530

    alsa_sound : Fix for audio routing speaker though BT is connected

    - Speaker device is selected by getDeviceForStrategy() though BT is
      connected as mA2dpSuspended is still remaining as 'true' even
      after ending the voice call.
    - This flag is not updated by checkA2dpSuspend() as force usage
      for communication and record is still FORCE_BT_SCO after ending
      voice call.
    - Modified condition check to set mA2dpSuspended to 'false' as soon
      as phone state is changed to MODE_NORMAL

    Change-Id: Ibc0d19b361755d8715c446401d2e14d914fd7020
    CRs-Fixed: 464404

commit 7cead94b4fb63d35949987ff35870cfe697b3e3c
Merge: 07e612c 2b0e59b
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Fri May 3 02:15:17 2013 -0700

    Merge "alsa_sound: Increase delay in temp mute"

commit 5accf62f09d6d6296ad4672911aca2fba606bf59
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Fri Apr 12 16:19:34 2013 -0700

    audio: fix for device not disabled after ending voice call

    - Issue: input/output devices are not disabled when voice
      call is ended
    - Cause: In AudioPolicyManager::setPhoneState(), a change is made
      to update devices for all outputs. GetNewdevice() on primary
      output is called again. However setParameters is not called to
      close i/o device in setOutputDevice().
    - Fix: revert to use Google's default logic for primary output.
      Update device only for non-primary outputs.

    Change-Id: If8e855591db22796c7117f2ec2570c6c525cdf54
    CRs-Fixed: 474403

commit 07e612c6fb041e0b04d1e5fd38d20b122d554c1f
Merge: ca11c8b bebeaeb
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu May 2 01:17:22 2013 -0700

    Merge "alsa_sound: Send FM TX PREPROC data from ACDB to the kernel"

commit ca11c8b9f5a8e7bf8d94fcd0a10534013c25d005
Merge: 46d3f48 7fa0f9f
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu May 2 01:17:21 2013 -0700

    Merge "alsa_ucm: Get volume mixer control and acdb cap from UCM"

commit 46d3f484bd6c509faaf89d48f0fec398537b1b5e
Merge: 5c70828 dd17056
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Apr 30 14:00:30 2013 -0700

    Merge "alsa_sound: Fix for voice call to reside on headset when camcording starts."

commit 67a0a0a07f8ce864465bfd87f7deec47dc5cf85d
Author: Mike Kasick <mike@kasick.org>
Date:   Fri Jan 4 16:12:55 2013 -0500

    alsa_sound: Allow SPEAKER_DUAL_MIC_* UCMs with USES_FLUENCE_INCALL

    Enables SND_USE_CASE_DEV_SPEAKER_DUAL_MIC_* UCMs when using speakerphone
    with USES_FLUENCE_INCALL defined.  This allows devices to use different
    DMIC settings for earpiece and speakerphone modes.

    Devices that wish to retain the old behavior should clone the DUAL_MIC_*
    UCMs for SPEAKER_DUAL_MIC_*.

commit 5c70828541414362efdd540a7319d450c34cb9da
Merge: 8a17bbe a82aa81
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Apr 29 06:55:44 2013 -0700

    Merge "  libalsa_intf: Added Dual Mic Endfire SGLTE devices"

commit f94edcb9fa6af28b28802f8a339fcd1b62dbf343
Author: detule <ogjoneski@gmail.com>
Date:   Sun Apr 28 19:51:24 2013 -0400

    audio-caf: fix compressed audio flags

    Change-Id: I8d3ca4d51a5dda9bb2723648629f1fd8a4979a28

commit 47cddcec1d420f4318139f63953e89c826e8c369
Merge: 94db555 2cb2791
Author: Steve Kondik <shade@chemlab.org>
Date:   Sun Apr 28 20:41:16 2013 -0700

    Merge "alsa: fix platform detection for m7 devices" into cm-10.1

commit 94db555b74073be54c115ad0747321f94d7631b4
Author: dhacker29 <davidhackerdvm@gmail.com>
Date:   Fri Apr 26 21:32:42 2013 -0500

    Add back ifdef QCOM_COMPRESSED_AUDIO_ENABLED for sound/compress_params.h

    Motorola Qcom devices haven't completed a 3.4 kernel and use a 3.0.31
    kernel.  This brings back ability to use audio-caf on these devices.
    This was originally part of commit b0f0fce093190136a2def290051a2102be2ba0e4
        audio: Initial bring up of QCOM added audio feature(jb_cho).

    Change-Id: Id83897b3e9b5b027d4c27ab3fa82f37fb4846f15

commit 2cb2791aa00eddfd7f9b6282b2e2a172f5f19147
Author: Daniel Hillenbrand <codeworkx@cyanogenmod.org>
Date:   Fri Apr 26 22:47:49 2013 +0200

    alsa: fix platform detection for m7 devices

    Change-Id: I4e6a884f5905d53acd85cf4a8386f8374e99266e

commit 8a17bbe32f1955283dd5fef410a78c2f4239d8d9
Merge: b2901be df9175f
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Fri Apr 26 09:56:15 2013 -0700

    Merge "alsa_sound: Parallelize MDM call sequence with APQ codec setup"

commit b60faf2f605f92fa6ca695e44c95e739d24eb63f
Author: Sharad Sangle <assangle@codeaurora.org>
Date:   Thu Apr 25 18:34:11 2013 +0530

    alsa_audio:remove the device from device list if open fails

    -Opening second record session is failed,which is
    expected but  after that user is no more able to
    open record session
    -The device is already got added in alsa_handle after
    calling mALSADevice->open but if this open fails, need
    to be deleted from device list.
    -If mALSADevice->open is failed then erase the device from from
    mDeviceList so that next time it can opened.

    Change-Id: I9847f65718d8330292f312dffe18dfcb672f2d3d

commit bebeaebe79d0563bfabcbab3192ee4c401cef4b9
Author: Damir Didjusto <damird@codeaurora.org>
Date:   Fri Apr 19 15:13:43 2013 -0700

    alsa_sound: Send FM TX PREPROC data from ACDB to the kernel

    Send FM TX PREPROC data from ACDB to the kernel.

    Change-Id: I9f5922bcbdece888ad8264f248626b78ee24f699

commit 7fa0f9fb1b95448e11e98558bf75a18d2c3d1f6d
Author: Damir Didjusto <damird@codeaurora.org>
Date:   Fri Apr 19 13:38:58 2013 -0700

    alsa_ucm: Get volume mixer control and acdb cap from UCM

    Add the capability to get the volume mixer control and
    acdb capability field from UCM based on the current
    use case or device.

    Change-Id: I339d178c901ee262ef18ee1d3bbc3d298ec16823

commit dd1705664e52c3e48befde57654eb525e560266d
Author: Srikanth Katta <skatta@codeaurora.org>
Date:   Wed Apr 24 16:21:22 2013 +0530

    alsa_sound: Fix for voice call to reside on headset when camcording starts.

    -During voice call with wired headset/BT headset/HDMI, if video
    recording starts, voice call is routed to DUT earpiece.

    -When camcording starts, in getDeviceforInputSource(), by default
    DEVICE_IN_BUILTIN_MIC is selected, and accordingly DUT earpiece
    is selected as out device.

    -Appropriate input (Tx)/output (Rx) device (wired, BT headset, HDMI) is now
    selected, based on the device connected during camcording and the mode in
    which handset is operating (call/voip). If multiple devices are connected,
    the precedence is BT-> wired headset->speaker.

    CRs-Fixed: 476531

    Change-Id: I1ddc6460a91752fd7673d84fd41b70365d83d90c

commit a82aa81e0fdeced9d1ce9ea71365c874c7ffe33c
Author: Sidipotu Ashok <sashok@codeaurora.org>
Date:   Wed Apr 24 18:11:18 2013 +0530

      libalsa_intf: Added Dual Mic Endfire SGLTE devices

      Corrected the variables such that it matches to devices that are
      presented in UCM file.

      CRs-Fixed: 477121

    Change-Id: Ib01616d33ef77926a0f5ff9befac0f58da134e96

commit b2901be4c1ebcad6af45a6e990001c9924cfb8bd
Merge: 257ccc5 e50401a
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Apr 24 03:46:50 2013 -0700

    Merge "  alsa_sound: Fix for crash during open of multiple VoIP streams"

commit 257ccc5b024cf9b265c22dd962ddb8611d6c749a
Merge: bfefbdc 376e2ab
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Apr 23 18:47:40 2013 -0700

    Merge "alsa_sound: Add metadata mode changes for LPA playback"

commit 376e2ab38c8546ee8a84314932609b6a05a70737
Author: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
Date:   Thu Mar 14 09:57:57 2013 -0700

    alsa_sound: Add metadata mode changes for LPA playback

    -Seek to EOS was causing playback to hang for 3 seconds before
     switching to the next clip.
    -This is because the lpa driver works on period size. Partial
     buffers are not handled.
    -Add metadata mode changes to AudioSessionout for LPA playback to
     support partial frames.

    CRs-Fixed: 458904
    Change-Id: Ibe3225d50e1c4248ba5c41e8d2a7067fc20108d9

commit 41eaae732627d99533993de7120a497e6f416a37
Author: Arne Coucheron <arco68@gmail.com>
Date:   Wed Apr 17 05:17:23 2013 +0200

    audio/msm7x30: Stop the log spam during VOIP calls

    Change-Id: Idfe211b1fef6f2aeee51ebe7f1b6ce3f4a0787be

commit ed32cc68719368dc066b41a76b2a2b816a625189
Author: Susmitha Perla <perla@codeaurora.org>
Date:   Mon Dec 3 15:28:37 2012 +0530

    audio/msm7630: Fix for Echo in Gtalk

    - Echo is observed in Gtalk.
    - Audio HAL informs that clients that it handle the Echo
      cancellation and with AHAL client turns off the EC.
    - EC is not supported in this PL remove it from getparam
    CRs-Fixed: 427161

    Change-Id: I62eba7ed7566045f61c34a51c8ac36029b8596ea

commit d04614cfefed7eb53b3e1313edd02cd48f8c8711
Author: Arne Coucheron <arco68@gmail.com>
Date:   Tue Apr 16 05:11:59 2013 +0200

    audio/msm7x30: Use q5v2_msm_audio_mvs_frame struct

    Otherwise a packet size of 640, which 7x30 modem doesn't support,
    will be used. 7x30 expects a packet size of 320.

    Was changed in AudioHardware::AudioStreamOutDirect::write, but
    seems to have been forgotten here.

    Related kernel change: http://bit.ly/M7ZqMN

    Change-Id: Ie89da836cbe27ea6b6cd10ef5c830ecc7020a624

commit 3064498919fbaf1e7d194ca0253a83f39418bbf0
Author: Steve Kondik <shade@chemlab.org>
Date:   Sat Apr 13 07:40:52 2013 -0700

    msm8660: Fix compilation error for 8660

    Change-Id: I719a9dc910ff2a65c6fb23f3e6a3784c4495a173

commit 17300005363e3d66903dcfc59ab8950e6485d407
Author: Matt Filetto <matt.filetto@gmail.com>
Date:   Wed Apr 10 14:10:25 2013 -0700

    Fix VOIP audio routing issue and add usb audio hal profile

    Adapted from arco's audio_policy.conf for msm7x30

    Change-Id: I0c10f382cd96dd50f86c082d1f633ec360f87692

commit 73554f097128a6256c9cdc0761ab0ef7cbc9543a
Author: Arne Coucheron <arco68@gmail.com>
Date:   Thu Mar 28 13:01:31 2013 +0100

    audio/msm8660: Squashed set of updates

    * Updates cherry-picked from msm7x30's jb_2.5.3 branch.

    msm7630 : Send proper channleMask for getInputProfile()

    -Unable to perform Voice call recording.
    -Unable to find profile for the device.
    -Update proper channelMask for getInputProfile().

    Change-Id: Ic67a1d59c4ea54ecccab2958e3c94672282f710f

    audio/msm7630: Add usb audio hal profile.

    remove USB ACCESSORY support from STRATEGY_PHONE to force speaker

    Change-Id: I080d1e547310d48b43bdf9bae8ee789887766b3c

    audio/msm7630: Fix the issue of voice call when HDMI connected

    - Voice call is not working when HDMI cable is connected.
    - When HDMI is connected getDeviceForStrategy is picking
      up HDMI rx as active device, because of this voice call
      is not working.
    - Avoid updating the HDMI as active output device, when
      strategy is STRATGY_PHONE.

    (cherry picked from commit 93bc81a7cd494373c19e5611fed060732a14b31b)

    Change-Id: I74be601f771883d9dfecc751994a4068acaf2d03

    audio/msm7630: Fix to enable TTY device

    -Connecting TTY device is not showing proper RX and TX values.
    -Mismatch in Key values of TTY modes in HAL.
    -Change TTY mode string values as per the values in PhoneGlobals.java.

    CRs-Fixed: 449513
    Change-Id: I53e52b057bb535647731bdd7b99cb065b2e50940

    audio/msm7630: Fix for back to back VOIP calls failure

    - Back to back VOIP calls fail.
    - MVS driver is not closed in HAL at the end of VOIP call
      due to wrong book keeping of VOIP in and out streams.
      This prevents back to back VOIP calls.
    - Enhance bookkeeping code and close driver at end of VOIP call

    (cherry picked from commit 2884ae26b9b65f49acee44a49efbc6196ab6d15f)

    Patchset 2: fix unneeded bool usage (isDriverClosed)

    Change-Id: Ia99149fca9c47846bc7c5a71bc23e1cbc30a16fa

commit 672734efc42c82ad75794020368ca14efbbed24c
Author: Sunil Joseph <sunilj@codeaurora.org>
Date:   Fri Oct 26 17:19:49 2012 +0530

    audio/msm7630: Fix for back to back VOIP calls failure

    - Back to back VOIP calls fail.
    - MVS driver is not closed in HAL at the end of VOIP call
      due to wrong book keeping of VOIP in and out streams.
      This prevents back to back VOIP calls.
    - Enhance bookkeeping code and close driver at end of VOIP call

    (cherry picked from commit 2884ae26b9b65f49acee44a49efbc6196ab6d15f)

    Change-Id: I4aa578442c24949b56586a5260fefd5e73c9292a

commit 07e24e8f72de7358323cc69f1afb7fadf7ad4cc8
Author: Kalyani polepeddy <kpolep@codeaurora.org>
Date:   Mon Feb 25 11:22:07 2013 +0530

    audio/msm7630: Add latency support for LPA

    -Audio loss for few seconds on headset connection.
    -Stream is unmute in checkDeviceMuteStrategies before device switch
    occurs.
    -Increase delay in tempMute to ensure unmuting happens after device
    switch happens. Also change latency for LPA to correct value similar
    to that of hpcm.

    Change-Id: I539a4726f6b9d53bbfdc5f92a8801d5f14c360eb
    CRs-Fixed: 448355

commit 9a86b949818b2d8cd93829a26c49dfdb55f804f0
Author: Vineela Tummalapalli <vineela@codeaurora.org>
Date:   Wed Feb 13 15:00:03 2013 +0530

    audio/msm7630: Fix to enable TTY device

    -Connecting TTY device is not showing proper RX and TX values.
    -Mismatch in Key values of TTY modes in HAL.
    -Change TTY mode string values as per the values in PhoneGlobals.java.

    CRs-Fixed: 449513
    Change-Id: I53e52b057bb535647731bdd7b99cb065b2e50940

commit 510f9c3957f7d59fdcbba338e454304d445df710
Author: Steve Kondik <shade@chemlab.org>
Date:   Thu Apr 11 13:06:43 2013 -0700

    msm8660: Add dependency on kernel headers

     * Let's get rid of headers in device repositories.

    Change-Id: I6408a1368cc1e3cd2f11d80f7eed54085dcd60c0

commit bfefbdc1503d96e1228d2ac864b8569d12638610
Merge: 035c35e 40ae587
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Sat Apr 20 21:45:18 2013 -0700

    Merge "audio: fix for device not disabled after ending voice call"

commit e50401a9825f4394317431876775d00726d3acc9
Author: Sidipotu Ashok <sashok@codeaurora.org>
Date:   Fri Apr 5 17:53:14 2013 +0530

      alsa_sound: Fix for crash during open of multiple VoIP streams

      --Device crashes when try to open multiple VoIP streams.

      --There is no check on input/output VoIP streamcount.

      --Reject further requests to open VoIP streams, if input/output
        VoIP stream count is already 1.

      CRs-Fixed: 472277
    Change-Id: I68bd2576469832f79878c26eec09faa5d7715921

commit 41ff904741fa8fe2164711de9553e990447a52b9
Author: Andrew Dodd <atd7@cornell.edu>
Date:   Wed Apr 17 18:11:08 2013 -0400

    alsa_sound: Support using legacy CAF input buffersize logic

    Older CAF audio HALs only used getInputBufferSize for 5.1 channel
    audio.  Newer CAF HALs use it in numerous other use cases too,
    including for stereo audio recording.

    This logic breaks with older kernels using the legacy CAF
    logic for input buffer sizes.  Allow the board to force the old
    logic.

    Typical symptoms of needing this logic are if any form of
    audio recording (including video recording in the camera)
    fail.

    Examples of affected CAF tags:
    A8064AAAAANLYA101034
    A8064AAAAANLYA1334

    Change-Id: Idc79b63ee781d7a7ac7eed20bad261e1020abd65

commit df9175fa223c460c115afb0d4fe317ee237df3ae
Author: Vidyakumar Athota <vathota@codeaurora.org>
Date:   Mon Mar 25 13:58:05 2013 -0700

    alsa_sound: Parallelize MDM call sequence with APQ codec setup

    Parallelize codec configuration on APQ with CSD voice call
    sequence on MDM. This is to reduce in call device switch delay

    Change-Id: Id5770a90b50c65f05dcbea119880421e98b33f4f

commit bd30aeb4475c229ca389a98a5ac01ba5ce88160a
Author: Ravishankar Sarawadi <crsaraw@codeaurora.org>
Date:   Mon Apr 1 11:02:18 2013 -0700

    alsa_sound: Changes for adsp restart handling

    ALSA device open does not check for if adsp is up and running
    before calling open device. It keeps on looping on device open call
    which blocks apps processor to trigger adsp restart during subsystem
    restart handling. To avoid this add check adsp state before calling
    ALSA device open operation.

    Change-Id: I5a637ce0799ec07e5e87af17be1ddb6ab05e49a6

    Conflicts:
    	alsa_sound/ALSADevice.cpp
    	alsa_sound/AudioHardwareALSA.h

commit 5ec3314e627fb13b3ead4a6148877e43b5e75899
Author: Mike Kasick <mike@kasick.org>
Date:   Wed Dec 26 21:27:52 2012 -0800

    alsa_sound: Add support for "Voice Earpiece Extra Gain" mode

    Uses "Voice Earpiece Extra Gain" mode instead of "Voice Earpiece" during
    voice calls when persist.audio.voc_ep.xgain=1.

    Change-Id: I9f00a81687fa9605c68308a74e0ffaaea7e698fc

commit 8fadd478ab1668645e6b5e8922338a7024a9eefd
Author: Steve Kondik <shade@chemlab.org>
Date:   Mon Apr 15 23:55:12 2013 -0700

    audio: Dynamic library loading, compilation fixes.

    Squashed commit of the following:

      commit 45b2a0b5e2344ead1413d627875bf10f79e15163
      Author: Steve Kondik <shade@chemlab.org>
      Date:   Sun Dec 23 04:50:52 2012 -0800

          audio: Dynamic library loading, compilation fixes.

          Change-Id: I04d5e7b96611572a7758c610ebdf0fd00e8346aa

      commit 2c78505b30f2362295e4a995039abf2c650f0d7f
      Author: Mike Kasick <mike@kasick.org>
      Date:   Tue Jan 8 10:31:51 2013 -0500

          alsa_sound: Add missing assignment of acdb_handle, fixes in-call volume

          Assigns AudioHardwareALSA::mUcMgr->acdb_handle to mAcdbHandle, which is
          missing in the port of dynamic library loading from the AOSP/mako HAL
          (I04d5e7b96611572a7758c610ebdf0fd00e8346aa).

          Restores acdb_send_voice_cal and acdb_send_audio_cal calls, which among
          other things, are required for in-call volume adjustment to function.

    Change-Id: I5b991a0be410af5eaff133625b63504545d7aa86

commit fea0ec18c5637921224d3def5e32f8058b687f8b
Author: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
Date:   Wed Mar 13 00:45:17 2013 -0700

    alsa_sound: Fix recording synchronization issue

    Record thread close and open is not synchronized and was resulting
    in some scenarios. Add userspace fd to come out of poll. This
    fixes the ANR while partial removal of usb headset.

    CRs-Fixed: 454851
    Change-Id: Ic66acfaff0367bdc4d646d6952444de483dc1305

    Conflicts:
    	alsa_sound/AudioUsbALSA.cpp

commit 40ae5876cd2dbcf1ae6e2eaa54bb39848371d484
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Fri Apr 12 16:19:34 2013 -0700

    audio: fix for device not disabled after ending voice call

    - Issue: input/output devices are not disabled when voice
      call is ended
    - Cause: In AudioPolicyManager::setPhoneState(), a change is made
      to update devices for all outputs. GetNewdevice() on primary
      output is called again. However setParameters is not called to
      close i/o device in setOutputDevice().
    - Fix: revert to use Google's default logic for primary output.
      Update device only for non-primary outputs.

    Change-Id: If8e855591db22796c7117f2ec2570c6c525cdf54
    CRs-Fixed: 474403

commit 035c35ef82da46680dff5bfd78d0b5ff3a6111eb
Merge: 0bd26f2 4d1f8e0
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Apr 8 21:10:29 2013 -0700

    Merge "alsa_sound: Add ACDB ID to UCM mapping"

commit 152073ae8046328c90e4c48cce0921e82f2c16e9
Author: Steve Kondik <shade@chemlab.org>
Date:   Sat Apr 6 19:44:23 2013 +0000

    audio: Fix separated audio input support

     * Fix declarations for QCOM's refactoring.
     * Allow enablement via board flags.

    Change-Id: I162d6f1d9a955df2dd9f4e47fb04b081664fcaf4

commit 374afc0bca262a77cd852f971aa32286f98d90d9
Author: Steve Kondik <shade@chemlab.org>
Date:   Sat Apr 6 19:29:21 2013 +0000

    audio: enable fluence incall feature

commit e5066ed658c955afefda96ea1679214443aaa7ec
Author: Steve Kondik <shade@chemlab.org>
Date:   Sat Apr 6 19:23:58 2013 +0000

    alsa_sound: Add support for Audience A2220

    Change-Id: Iec6d384feef11fb23b1cbacc686d065085bd8a11

commit ac921408a3b8625277d2969eba184546cf93b5f2
Author: Araemo <araemo@gmail.com>
Date:   Sat Apr 6 19:20:45 2013 +0000

    Enhance ALSA for USB Dock Audio support

    Needed for http://review.cyanogenmod.org/#/c/30481/

commit d81f36f233e8db114b2a64f9bd7b28115b3b08ff
Author: Daniel Hillenbrand <codeworkx@cyanogenmod.org>
Date:   Sat Apr 6 18:56:57 2013 +0000

    import msm7x30/msm8660 audio from CyanogenMod/android_hardware_qcom_audio-caf

    commit efe9282ec3996157e6a4eea1b1813edae2bb2ee0
    Author: Kalyani polepeddy <kpolep@codeaurora.org>
    Date:   Wed Jan 23 16:17:02 2013 +0530

        7x30 : Port heap mask change to ion

        -mp3 clips fails to play intermittently.
        -ION fails to register with the driver.
        -Heap mask field passes as argument at allocation
        time to specify ion heap.

        CRs-Fixed: 444197

        Change-Id: I35b7f162c906b856b7b4f5ee2270cd10256144bf

    commit 8e4f7cca4a61ee4d85cd4d4ff7ccb4f9bb5c2c26
    Author: Deepa Madiregama <dmadireg@codeaurora.org>
    Date:   Thu Oct 18 18:01:29 2012 +0530

        audio/msm7630: Fix the issue of voice call when HDMI connected

        - Voice call is not working when HDMI cable is connected.
        - When HDMI is connected getDeviceForStrategy is picking
          up HDMI rx as active device, because of this voice call
          is not working.
        - Avoid updating the HDMI as active output device, when
          strategy is STRATGY_PHONE.

        (cherry picked from commit 93bc81a7cd494373c19e5611fed060732a14b31b)

        Change-Id: I74be601f771883d9dfecc751994a4068acaf2d03

    commit 485138c2e236ee472c39e02888fd7ce78117cd78
    Author: Vineela Tummalapalli <vineela@codeaurora.org>
    Date:   Mon Jan 21 10:12:02 2013 +0530

        audio/msm7630: Add usb audio hal profile.

        remove USB ACCESSORY support from STRATEGY_PHONE to force speaker

        Change-Id: I080d1e547310d48b43bdf9bae8ee789887766b3c

    commit 99b968e9a35762baf94bc0537a9667add8e3c49b
    Author: Vineela Tummalapalli <vineela@codeaurora.org>
    Date:   Wed Jan 16 13:03:58 2013 +0530

        msm7630 : Send proper channleMask for getInputProfile()

        -Unable to perform Voice call recording.
        -Unable to find profile for the device.
        -Update proper channelMask for getInputProfile().

        Change-Id: Ic67a1d59c4ea54ecccab2958e3c94672282f710f

    commit 2341deb2477e23db014679946618ecc8db1cdc85
    Author: Shashi Kumar <sulege@codeaurora.org>
    Date:   Mon Oct 22 16:33:07 2012 +0530
        audio/msm7630: Fix for volume change during voice call

        - Volume change is not working during voice call
        - Calculate voice volume is missing at AudioPolicyManager,
          due to this 7630 HAL did not set voice volume change.
        - Add voice volume calculation at AudioPolicyManager

        CRs-Fixed: 412991
        (cherry picked from commit 207e377f951b0b63891097bf226974be38079f98)

        Change-Id: I84d2b804c8cdd8fde30968e9cbec3debbf69ae33

    commit 54404d29ed8d1c901f0d883cd22d9ecaef98e5fb
    Author: Vineela Tummalapalli <vineela@codeaurora.org>
    Date:   Fri Jan 11 20:11:42 2013 +0530

        audio/msm8655: Fix for AAC recording failure.

        -Failure in AAC recording.
        -getInputBufferSize() returns zero input buffer size.
        -Changing the data type of channelCount parameter in getInputBufferSize().

        Change-Id: I1840f324949e2dd034110016359245cd7b045709

    commit 47b9b063a0c766ed59aaaf5eac3a775a167ea332
    Author: Vineela Tummalapalli <vineela@codeaurora.org>
    Date:   Sat Dec 29 12:42:39 2012 +0530

        audio/8655: Enabling LPA playback by adding AudioSessionOutLPA class

        -AAC and MP3 clips are not playing in LPA path.
        -Class AudioSessionOutLPA is not defined
        -Define and implement AudioSessionOutLPA

        Change-Id: Ie34c28d623d6c841bbddcbbe027317b0bd658660

    commit 2dab1aa403f0755762a48bb151ff6130bd3562b0
    Author: Kalyani polepeddy <kpolep@codeaurora.org>
    Date:   Wed Jan 2 15:40:18 2013 +0530

        Remove audio_poliy.conf from msm7630

        Change-Id: I5f0a6055c41c804e25d96df2447c606abd45d143

    commit 38b24cf972314ac94b6b9ee9ff187bf17b9066c0
    Author: Arne Coucheron <arco68@gmail.com>
    Date:   Mon Jan 7 06:08:03 2013 +0100

        audio/msm7x30: Support for Samsung audio devices

        Change-Id: I2a624cddd99e02a9e0c73e829cc28dd520902074

    commit 868374b297dda4d10d685a1c4c116599a4795fc4
    Author: Arne Coucheron <arco68@gmail.com>
    Date:   Sat Jan 5 00:50:43 2013 +0100
        audio/msm7x30: Fix log tags and disable debug logging

        Change-Id: I4712b09c86e3a067b0c847661fa43828be6eee23

    commit 8659a58e87eb5dc170427e90c5b1101d12149447
    Author: Arne Coucheron <arco68@gmail.com>
    Date:   Sat Jan 5 00:47:14 2013 +0100

        audio/msm7x30: Fix build in CM

        Change-Id: I78fa2a67cc36e88d5866ecbe67dd3cdd6d58dde1

    commit c4e2cdd717e225f3b4dea44acdcbd55d42dea188
    Author: Arne Coucheron <arco68@gmail.com>
    Date:   Sat Jan 5 00:28:23 2013 +0100

        audio/msm7x30: Rename folder to CM naming standard

        Change-Id: I6a2620a4daade6bae5cc5a5fc0e72e697f9808d9

    commit d149f409c12edf388d2faad9801208102bacf094
    Author: Steve Kondik <shade@chemlab.org>
    Date:   Sat Jan 26 02:10:07 2013 -0800

        msm8660: Update policy manager for MR1

         * From 7x30 HAL

        Change-Id: I4dc080e1fe3f81cb6e6ba3aa0e51c43143cd5b11

    commit c2de7f54ebd154501513496391fb55bb9cac3043
    Author: Steve Kondik <shade@chemlab.org>
    Date:   Fri Jan 25 23:51:18 2013 -0800

        msm8660: Update for audio device API 2.0

         * See 7801df9b35653562139ddaaa5c29c61f0eddc92f for 8960

        Change-Id: I6684955123ac1b1f3c09653aeb95a1acb7e345d1

    commit bbd2939c1f0733e4ff01a47c4a14e76227377022
    Author: Steve Kondik <shade@chemlab.org>
    Date:   Fri Jan 25 23:31:23 2013 -0800

        msm8660: Import audio HAL

        Change-Id: I81ca52d392e5bba337f4a3a1288f716ba265ffe1

commit 4d1f8e092b17d9bb06f697a961eebc3cbc3c6c27
Author: Jay Wang <jaywang@codeaurora.org>
Date:   Fri Apr 5 10:01:27 2013 -0700

    alsa_sound: Add ACDB ID to UCM mapping

    Add mono speaker mapping from ACDB ID to UCM device

    Change-Id: I2233830681c411ebfe847fbcf6a8c6a0c51cf16a
    CRs-Fixed: 467689

commit 2b0e59b4b23bfb48537cd812806a60baea5bf9e2
Author: Yamit Mehta <ymehta@codeaurora.org>
Date:   Fri Apr 5 17:07:33 2013 +0530

    alsa_sound: Increase delay in temp mute

    -Silence for small time when headset is inserted
     and then touch tone is played.
    -Stream is unmuted in checkDeviceMuteStrategies before
     device switch occurs.
    -Increase delay in temp mute to ensure unmute after
     device switch.

    CRs-Fixed: 470160
    Change-Id: I6bc443abf77e5600675846086a57090f49bf9bd7

commit 0bd26f27d87428fbd54e5d7aa70873e579d83446
Merge: 4711e5b f1ee977
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Apr 4 17:58:42 2013 -0700

    Merge " alsa_sound: FENS Support for VOIP calls"

commit f1ee9772e80d7bbcce175e9dbcd2118f9b3f8bcb
Author: Sidipotu Ashok <sashok@codeaurora.org>
Date:   Wed Mar 13 13:06:42 2013 +0530

     alsa_sound: FENS Support for VOIP calls

    --FENS support is not there for Voip calls
    --It was not supported earlier
    --Support is added by adding new Mixer control
    CRs-Fixed: 460878

    Change-Id: I61f77a812b1eca8a95c78114815feb90be66a3cd

commit 4711e5bb8cae71bed787603c82f9ba2f7b9bcefc
Merge: c964018 a2787ae
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Apr 4 14:06:00 2013 -0700

    Merge "alsa_sound: Configure period count for pcm recording"

commit c964018169bd12e71b4b6be4e5472482f1119ca0
Merge: c91fa65 1144c47
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Apr 4 12:06:43 2013 -0700

    Merge "alsa_sound: Fix corruption in recording over usb."

commit c91fa658415bd8a4014314d59b9b18f0b589682b
Merge: 5b7557a 3469c73
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Apr 3 13:44:35 2013 -0700

    Merge "alsa_sound: Patch to remove delay in second MT call during hold"

commit 5b7557a435faf906e4034e60bb1b5308aa8667b9
Merge: b4a38a2 675fabc
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Apr 2 20:46:20 2013 -0700

    Merge "alsa_sound: set ALSADevice::mInChannels to 0 after voice call finishes"

commit 675fabc2e69196b870022f432eecc285abcaf257
Author: Jitendra Singh Naruka <jnaruk@codeaurora.org>
Date:   Wed Mar 27 18:35:36 2013 -0700

    alsa_sound: set ALSADevice::mInChannels to 0 after voice call finishes

    When a voice call is established, through  enableVoiceCall()
    mInChannels is set to non-zero value, so during disableVoiceCall()
    it should be reset to zero.

    Change-Id: Ic3f39990c7e1820e5c61e063b8c84e160cdd20bb
    CRs-Fixed: 467775

commit b4a38a253b126079b1fac60cb53e3246056c3200
Merge: 52d5372 286d1ae
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Apr 1 14:54:59 2013 -0700

    Merge "alsa_sound: Fix for audio loss on switching from BT to AOA"

commit 52d53723b646a2a53950351702346be4b45385f2
Merge: d8451c4 1cb9107
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Apr 1 13:34:24 2013 -0700

    Merge "alsa_sound: Fix for Tunnelplayer IOCTL drain block"

commit 671cd2c096b79a06696a4b2c11cbc4addeebda08
Author: Daniel Hillenbrand <codeworkx@cyanogenmod.org>
Date:   Sat Mar 30 18:42:36 2013 +0100

    audio-caf: allow devices to only set min buffer sizes

    Change-Id: I182f86549b01d4c452cc94201c638c410592083e

commit d8451c47c3d3757b1d2d97ff3a12385e40636109
Merge: 2889a2d 0d458fb
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Fri Mar 29 19:06:23 2013 -0700

    Merge "hardware/qcom/audio: Fix for Crash during Gtalk startup"

commit 0d458fb8c574234919d9c1b1c34218bf73f941be
Author: Preetam Singh Ranawat <apranawat@codeaurora.org>
Date:   Mon Jan 28 11:13:13 2013 +0530

    hardware/qcom/audio: Fix for Crash during Gtalk startup

     -Crash is observed when Hangouts and Gtalk start running
      simultaneously
     -Dirver does not support multiple instances of voip call.
     -Avoid creating multiple voip session if mVoipStreamCount
      is already 2
    CRs-Fixed: 440160
    (cherry picked from commit b9f1299e4ca8b96907f97914a0cf65fd5a3d3174)

    Change-Id: I7ef3cf76394e1469db9d21a1484aacf1251c4f25

commit a2787aeebf61bf4151d48b3627c127ea06804042
Author: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
Date:   Thu Mar 28 17:49:22 2013 -0700

    alsa_sound: Configure period count for pcm recording

    -While pcm recording is going on, if buffers are not requested from
     dsp at real time then dsp drops the buffers and sends partial
     buffers. This was causing record failures. Pcm recording platform
     driver has 2 periods.
    -If system is loaded read request will not be at real time. Add
     flexible period count to pcm record platform driver so client can
     configure the period count.

    Change-Id: Ib142411f526f0cde53d5936b0291d84fdee76651
    CRs-Fixed: 467746

commit c6ef48fef0c0acc682091c538071d8b435031b74
Merge: d9e9ce8 a921baf
Author: Giulio Cervera <giulio.cervera@cyanogenmod.org>
Date:   Fri Mar 29 10:26:16 2013 +0100

    Merge tag 'A8064AAAAANLYA161031' into HEAD

    "A8064AAAAANLYA161031"

    Change-Id: Ia63fbe714bab537d9f21e44b2ddd84678480c0f7

commit d9e9ce8019a64a3608e840d85c793d568926f986
Merge: 98f68b9 317759d
Author: Giulio Cervera <giulio.cervera@cyanogenmod.org>
Date:   Fri Mar 29 10:23:14 2013 +0100

    Merge tag 'A8064AAAAANLYA161030A' into HEAD

    "A8064AAAAANLYA161030A"

commit 1cb91077e7a7459e5787c5cc90aeae83037707a3
Author: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
Date:   Tue Mar 26 20:39:53 2013 -0700

    alsa_sound: Fix for Tunnelplayer IOCTL drain block

    -IOCTL drain was getting stuck when standby was
    called after seeking to EOS
    -This happens when seek was done to the
    end of the stream which would lead to only
    a zero length buffer being sent.
    In this case, IOCTL drain will block forever
    -To solve this, IOCTL drain was interrupted
    by calling flush from the context of standby

    CRs-Fixed: 467166

    Change-Id: I0d119c362670fd4f99b023dd4a1fdebde7832c02

commit 2889a2de6627fb0cd2e013675084d83090eb1fc4
Merge: 74320b4 f61e98d
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Mar 28 11:58:23 2013 -0700

    Merge "alsa_sound: Patch to fix device switch in skype"

commit 1144c47d0cf602795f1645bb5ef9e38cf8920bed
Author: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
Date:   Wed Mar 27 18:31:45 2013 -0700

    alsa_sound: Fix corruption in recording over usb.

    -Sometimes the hardware pointer for proxy driver is not updated for
     successive callbacks. This is because in the recording case proxy
     driver and usb driver runs at real time. This does not mandate
     that the data is read by proxy driver as soon as data is written
     from userspace. Any delay in getting dat from usb also can cause
     the issue.
    -Fix is to increase the start threshold of proxy side for recording
     so that playback is not started without sufficient data.

    Change-Id: I86384bfc96b8262309a1402ca9fe5744af3b436d
    CRs-Fixed: 465317

commit 74320b4073a9f0f3bba5cdbf09071b0720ea9fb8
Merge: 36fc3c6 2601141
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Mar 26 21:15:28 2013 -0700

    Merge "Change to handle A2dp suspend setparameter, which will suspend the thread"

commit 36fc3c604684704f15e9ee5464d36e38c31c2ffc
Merge: b681fde 58c9893
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Mar 26 14:45:46 2013 -0700

    Merge "alsa_sound: Fix for FM volume issue on USB headset"

commit b681fde826395db729fa664134c1389429276010
Merge: a619cff 8ad6c3a
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Mar 26 14:45:37 2013 -0700

    Merge "alsa_sound: Use current device to parse edid and configure HDMI device"

commit 58c989311a225f7fe40ad7bbca262297f4478aca
Author: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
Date:   Mon Mar 18 23:45:04 2013 -0700

    alsa_sound: Fix for FM volume issue on USB headset

    -Adding FM as a device caused several issues
      such as volume change for USB and other
      devices had no effect. LPA clips were not
      able to play when FM was enabled.
    -FM was included as a device along with the
      other actual device that was connected.
      Thus the final device was not the correct
      one.
    -Fixed by not adding the FM device in
      the global device variable, instead use
      a local device variable to handle FM. Also
      added setFmVolume changes which earlier relied
      on the device variable to contain FM.

    Change-Id: Idfb31a1cc6b8978974650683f4689e8c1efd1a29
    CRs-fixed: 418806

    Conflicts:

    	alsa_sound/ALSADevice.cpp

commit a921bafcfe235724806bcb2c94b97fc47a4253c1
Merge: 70c1271 08bcfdd
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Mar 26 06:37:37 2013 -0700

    Merge "Merge commit 'a619cffbf71ed7b2d3244624263b07616a2eff7a' into HEAD"

commit 08bcfddccf8595b654cdee6b65481ae0ec451f39
Merge: 70c1271 a619cff
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Mar 26 00:43:37 2013 -0600

    Merge commit 'a619cffbf71ed7b2d3244624263b07616a2eff7a' into HEAD

commit 70c127123dee2c4a01cf5a5392944d493335802f
Merge: c14ba79 1ec562b
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Mar 25 17:05:30 2013 -0700

    Merge "Merge commit '317759d230f747879f2fbffb9ed45fe769c9cac6' into HEAD"

commit 1ec562b27355bf4cc2631e73d604bd5ba7f90c20
Merge: c14ba79 317759d
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Fri Mar 22 15:32:39 2013 -0600

    Merge commit '317759d230f747879f2fbffb9ed45fe769c9cac6' into HEAD

commit 260114178fcf07ca788e7e7db28f41923972901b
Author: Vimal Puthanveed <vimalp@codeaurora.org>
Date:   Fri Mar 22 13:57:26 2013 -0700

    Change to handle A2dp suspend setparameter, which will suspend the thread

    When A2dp suspend setparameter is called we were not suspending the thread.
    We were suspending it only during stop. This will make sure we suspend the
    thread even during the suspend setparameter call

    CRs-Fixed: 415399
    Change-Id: I49ae25fc028e9d4927fd3d3131c8f97f3e873fc1

commit a619cffbf71ed7b2d3244624263b07616a2eff7a
Merge: 317759d a762a3a
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Fri Mar 22 10:35:28 2013 -0700

    Merge "alsa_sound: Fix for audio glitch on headset during FM playback"

commit a762a3ae67ff73bb3c89ed339a151df29c6b60da
Author: Karthik Reddy Katta <akkatta@codeaurora.org>
Date:   Tue Feb 5 14:39:02 2013 +0530

    alsa_sound: Fix for audio glitch on headset during FM playback

    - checkDeviceMuteStrategies() is muting and unmuting the stream
      when headset is connected. This is resulting in audio glitch.
    - Remove checkDeviceMuteStrategies() function call.

    Change-Id: Idaed450a58eabfacda60beafeae6f9ab45d06e3b
    CRs-Fixed: 448577

commit 3469c73005e7c0365bf1589d8627fafe7906d540
Author: Sidipotu Ashok <sashok@codeaurora.org>
Date:   Tue Mar 5 17:11:55 2013 +0530

    alsa_sound: Patch to remove delay in second MT call during hold

    -When voice call is on hold, when another MT call comes, the ringtone
     is delayed by 6sec.
    -This is a special case for fusion, when the voice call is on hold and
     there is another incoming MT call, telephony will set MODE_IN_RINGTONE
     During this there is a change in device from Handset to Speaker,
     this is not propagated to MDM.
    -Fix is to propagate device change to MDM during MODE_IN_RINGTONE
     if there is a call on hold already.

    CRs-Fixed: 447560

    Change-Id: Ia29729bc97a585bd45657e558d04374104e1ce9c

commit 286d1ae16396f0575f424b024c4c511d061ce72f
Author: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
Date:   Wed Mar 20 19:15:06 2013 -0700

    alsa_sound: Fix for audio loss on switching from BT to AOA

    -When AOA is connected to device while playback is on BT, usb stream is
     created. The external out stream gets updated with usb stream even
     when a2dp stream is active.
    -Update external out stream with a2dp or usb stream in openExtOut only
     if it does not refer to any other stream.

    Change-Id: I625ee20ac91ef1be4b21e91fd5e9fffc54a08033
    CRs-Fixed: 464457

commit 8ad6c3abfa8be7490d4d68095a77de9375240dce
Author: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Date:   Wed Mar 20 15:35:54 2013 -0700

    alsa_sound: Use current device to parse edid and configure HDMI device

    The device stored in the handle is used to get EDID and configure
    HDMI device accordingly. Use the current active device appropriately
    to validate and configure if HDMI is the current active device

    Change-Id: Ie10bff3fc63b77bb6d056826c14d381295487f3f
    CRs-Fixed: 457096

commit 98f68b940f5ca2cc457c14777a1707595e4a8855
Merge: 8e43d92 c14ba79
Author: Giulio Cervera <giulio.cervera@cyanogenmod.org>
Date:   Tue Mar 19 22:06:00 2013 +0100

    Merge tag 'A8064AAAAANLYA161030' into cm-10.1

    "A8064AAAAANLYA161030"

    Change-Id: Id7efcc5b10028b5802c3c42c08c62e113e36fd0b

commit 317759d230f747879f2fbffb9ed45fe769c9cac6
Merge: bcab792 63ab11a
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Mar 18 12:15:33 2013 -0700

    Merge "Primary output check for sonification is removed."

commit bcab792ef0fd42a310c5050c8c2e9a263cba21a3
Merge: 8bf055d c2e34a7
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Mar 14 18:05:07 2013 -0700

    Merge "audio: refactor HAL to call session id based CSD apis"

commit c2e34a73f2ce198009448ac278848c7e1596e976
Author: Shiv Maliyappanahalli <smaliyap@codeaurora.org>
Date:   Sun Mar 3 18:37:28 2013 -0800

    audio: refactor HAL to call session id based CSD apis

    To support multiple sessions such as Voice2, VoLTE in CSD,
    the CSD apis are refactored to have session ID as argument to
    uniquely identify sessions.

    Add support in HAL to call session ID based CSD apis.

    Change-Id: I46ba0d033f3ddac2955cd6c59b61655739a32b2c

commit 63ab11a92f5c8a42282f9a94ddcf13ee69d374a0
Author: Vimal Puthanveed <vimalp@codeaurora.org>
Date:   Mon Mar 11 11:53:40 2013 -0700

    Primary output check for sonification is removed.

     - When there are 3 different tracks (Ringtone, Music and Touchtone)
       the device switch happens for touchtones to primary if the primary
       output is active. This can lead to music being played in combo
       due to timing issue.
     - This is mainly because we have a check for, primary output if
       active we return true for SONIFICATION. This will set SONIFICATION
       for touch tones and this device switch will come after music device
       is set.
     - Hence music will start playing as SONIFICATION.
     - We have removed the primar output check for getting device strategy

    CRs-Fixed: 456842
    Change-Id: I8d49d7bcd46f06c0dffe87f6f3944027dee1bcc8

    Conflicts:

    	alsa_sound/AudioPolicyManagerALSA.cpp

commit 8bf055d7855544b0686ca6ba35f22ab301495511
Merge: 81d5667 088fbd1
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Mar 14 07:44:34 2013 -0700

    Merge "alsa_sound: Add support for new UCM file for SGLTE"

commit 088fbd1461608e1dea1d6f7871443479f90e1b0f
Author: Laxminath Kasam <lkasam@codeaurora.org>
Date:   Mon Mar 11 15:08:09 2013 +0530

    alsa_sound: Add support for new UCM file for SGLTE

    -Adding support for SGLTE 8x30 QRD platform

    Change-Id: Ic58db70619cc92dab70c98751f112384afd77d17

commit 81d5667c02d4ae0d4634382ac4d99db2f003e9f0
Merge: 7daa107 ccd5aab
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Mar 13 18:06:44 2013 -0700

    Merge "alsa_sound: Fix race condition in close and open device"

commit 7daa10715972ae9ba982bc9940885079b3ba2f02
Merge: b0d34c8 c3d5034
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Mar 13 14:39:43 2013 -0700

    Merge "alsa_sound : Handle sound card detection failures"

commit b0d34c83e2e293f4eb4e830470f4d43b61dc95a2
Merge: f0a4a6a 83d616c
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Mar 13 14:38:51 2013 -0700

    Merge "alsa_sound: Change SGLTE to Voice2 for reusability of code"

commit ccd5aabb8577f5fda0ce36fac1d5c9e0a2f1e544
Author: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
Date:   Wed Mar 6 14:20:42 2013 -0800

    alsa_sound: Fix race condition in close and open device

    -Play ringtones over USB headset. Switch fast between ringtones
    -The playback thread exit and open device in USB are not locked
     causing a failure for the next ringtone session. This causes a
     native crash.
    -The fix is to lock the exit path and open/configuraton of usb/pcm
     device.

    Change-Id: I532b528c68e26b35f565b2a93b310146621d942d
    CRs-Fixed: 458590

commit f0a4a6a6491ede9b4fe92945d027d0caa636ac7c
Merge: be66cfe e70ace4
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Mar 12 15:45:37 2013 -0700

    Merge "libalsa_intf: Add USB acdb IDs for voice over USB"

commit c14ba7982f2146be206cb92dcb7d54cb5f82699c
Merge: 40f60e8 b580e90
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Mar 11 15:30:51 2013 -0700

    Merge "Merge commit 'AU_LINUX_ANDROID_JB_2.5.04.02.02.40.184' into HEAD"

commit b580e907a241c1f737fb551d186b2fc0335d644e
Merge: 40f60e8 78cdd9c
Author: Dhivya Subramanian <dthiru@codeaurora.org>
Date:   Mon Mar 11 11:08:28 2013 -0700

    Merge commit 'AU_LINUX_ANDROID_JB_2.5.04.02.02.40.184' into HEAD

    * commit 'AU_LINUX_ANDROID_JB_2.5.04.02.02.40.184':
      libalsa-intf: Add support to handle numerical amix commands
      alsa_sound: Fix for no ringtone heard while playing on WFD
      alsa-sound: EOS notification if drain ioctl is interrupted
      audio: fix input buffersize for AMR/EVRC formats
      alsa_sound: Add support for VoLTE over USB headset

    Change-Id: Ifa5bc337f87ba05e671576576fee610dbc3f8eca

commit 83d616cb5f61f4d90ed7a2c383d7d099841a5681
Author: Neema Shetty <nshetty@codeaurora.org>
Date:   Wed Feb 20 19:18:05 2013 -0800

    alsa_sound: Change SGLTE to Voice2 for reusability of code

    Replace SGLTE with Voice2 for reusability of code.

    Change-Id: I6f068fddaf2737dc79d42f06a15ed14bd24d5562

commit be66cfe43d8ebe9988cac9f773452c1977b0aa24
Merge: bdc27ce 694d88c
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Mar 11 10:34:36 2013 -0700

    Merge "audio: getNewDevice on primaryOutput for strategy phone and enforced audio"

commit bdc27ce608cdaf0decdeef145d666540e396eae2
Merge: 78cdd9c 5e92747
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Mar 11 10:34:34 2013 -0700

    Merge "alsa_sound: Add missing param.remove(key) in setParameters"

commit 5e92747244185fa77393a87f4963102528a7d636
Author: Satish Babu Patakokila <sbpata@codeaurora.org>
Date:   Mon Feb 25 16:53:13 2013 +0530

    alsa_sound: Add missing param.remove(key) in setParameters

    AudioHardwareALSA::setParameters(const String8& keyValuePairs) will
    return BAD_VALUE unless all keys provided have been consumed.

    Added the missing param.remove(key) to the processing blocks for
    TTY_MODE_KEY, AudioParameter::keyFluenceType, INCALLMUSIC_KEY,
    ANC_KEY, BTHEADSET_VGS and MODE_CALL_KEY so that the keys are properly
    consumed
    CRs-Fixed: 455037
    Change-Id: I886b0195dbbfcdf2a292d79d09c27bc8edb928a3

commit c3d503493a26c886e9007bf87c6d7350c3fd1c33
Author: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
Date:   Thu Feb 28 16:20:50 2013 -0800

    alsa_sound : Handle sound card detection failures

    -If sound card is not detected, the error was not handled
     gracefully. This results in media server crash.
    -If there is an error in AudioHardwareALSA constructor update the
     error and delete/clear HAL instance. This prevents the media
     server crash if sound card is not detected.

    Change-Id: I6e5991d678ed374a55de722f16072bf5b2b28141
    CRs-Fixed: 457318

commit 40f60e814ed7f9224d1801e678a3ce125f5fe5c0
Merge: e3471f4 5efc941
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Mar 7 17:57:07 2013 -0800

    Merge "Merge commit 'AU_LINUX_ANDROID_JB_2.5.04.02.02.40.173' into HEAD"

commit e70ace4459a5489eec25ec5fda16210b7e657b31
Author: Vidyakumar Athota <vathota@codeaurora.org>
Date:   Thu Mar 7 14:53:24 2013 -0800

    libalsa_intf: Add USB acdb IDs for voice over USB

    - Add USB Rx and Tx acdb ID for voice over USB

    Change-Id: I6ac9c7ed81dd0189dba5fa8ad3eabe3dbe93e1ef

commit 78cdd9c8e7f0ad07f72b8deffa99c06e3e49ee11
Merge: db974a1 722337a
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Mar 7 14:35:30 2013 -0800

    Merge "alsa-sound: EOS notification if drain ioctl is interrupted"

commit 694d88c26e0c778fb27b4a389dc6c59fab54b2f6
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Wed Mar 6 18:29:35 2013 -0800

    audio: getNewDevice on primaryOutput for strategy phone and enforced audio

    - only device for primaryOutputDesc will be used for STRATEGY_PHONE and
      STRATEGY_ENFORCED_AUDIBLE, request a routing change only on primary
      output for these two strategy

    Change-Id: Ibb2c01f5d81886ad195662af2c5870ec9301cf30
    CRs-Fixed: 446946

commit 5efc941b46053fb8e1e2fd8aff03e2cdc0b06d15
Merge: e3471f4 187fde4
Author: Dhivya Subramanian <dthiru@codeaurora.org>
Date:   Wed Mar 6 14:38:23 2013 -0800

    Merge commit 'AU_LINUX_ANDROID_JB_2.5.04.02.02.40.173' into HEAD

    * commit 'AU_LINUX_ANDROID_JB_2.5.04.02.02.40.173':
      alsa_sound: Initialize the mPfdProxy in constructor
      alsa_sound: Fix to prevent additioal doRouting call
      alsa_sound:Fix for Audio leak in LPA while in voice call
      alsa_sound:Resample proxy ouput for other than 48KHz samplerate
      libalsa-intf: Fix build warning from ALOGE
      alsa_sound: Support for applying independent channel volume control
      libalsa-intf: Add mono headset support

    Change-Id: Ib2cec3df09df37beb9de66faaea9c933d7deef95

commit db974a18a70e34d0dc5c16eadeff51f18bbfe6bc
Merge: 00d162b b39e39b
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Mar 5 17:48:22 2013 -0800

    Merge "audio: fix input buffersize for AMR/EVRC formats"

commit 00d162b554bdad1f9983804f3d6de6187ec45a36
Merge: 34125e9 eae2d49
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Mar 5 12:48:08 2013 -0800

    Merge "libalsa-intf: Add support to handle numerical amix commands"

commit 34125e90cef9585b279dcb9e21eea4c37cd90e47
Merge: 2683272 061f10d
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Mar 5 11:05:58 2013 -0800

    Merge "alsa_sound: Fix for no ringtone heard while playing on WFD"

commit 2683272f7f964b52659f646605b0acaee69529ce
Merge: 187fde4 3093973
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Mar 5 01:18:06 2013 -0800

    Merge "alsa_sound: Add support for VoLTE over USB headset"

commit eae2d49cc7b8da7d948e91f40d7f4771f27126ad
Author: Damir Didjusto <damird@codeaurora.org>
Date:   Tue Feb 12 17:56:38 2013 -0800

    libalsa-intf: Add support to handle numerical amix commands

    The numerical amix commands from UCM files are not supported in
    the current code unless they are the amix commands to set volume.
    But there are other cases such as sample rate, which are defined
    as numerical commands as well.  The change is to add the support
    to cover rest of the use cases.

    The change includes update to amix commands for BT SCO sample
    rate from string type to numerical type.

    CRs-fixed: 394165
    Change-Id: I0120046e40a9b24ac19d10b0daf091d7407ba4f1

commit f61e98dd74dc9b1c93a55f2d2ce1cf953cf8b67a
Author: Sidipotu Ashok <sashok@codeaurora.org>
Date:   Fri Feb 22 21:07:26 2013 +0530

    alsa_sound: Patch to fix device switch in skype

    -During a skype VOIP call, device switch to speaker is not working
    -When a device switch is attempted in skype during VOIP call, skype closes
     the existing record/VOIP tx session, re-opens it and sets the tx device
     which is BUILT_IN_MIC, due to the limiation in HAL, for a BUILT_IN_MIC
     device handset_rx is picked up instead of speaker_rx, also the
     SetForce() value is not considered due to this.
    -Fix is to check the current rx device in audio HAL and picking up
     speaker_rx instead of handset_rx

    CRs-Fixed: 453817

    Change-Id: I8a5ab4fd7b5ec75958d5db0441285e319688185f

commit 8e43d9298e518c7e31d0f08f841b5b8a6266b420
Merge: 99dfda5 ef8a10a
Author: Giulio Cervera <giulio.cervera@cyanogenmod.org>
Date:   Sat Mar 2 17:11:50 2013 +0100

    Merge tag 'A8064AAAAANLYA161023' into HEAD

    "A8064AAAAANLYA161023"

commit 061f10d1cd742f67854d7fb51a4fa5e7f7b856b6
Author: Amal Paul <amal@codeaurora.org>
Date:   Thu Feb 28 22:10:24 2013 -0800

    alsa_sound: Fix for no ringtone heard while playing on WFD

    - When ringtone is played while WFD session is active, it is
      heard on WFD sink and not on speaker. Ringtone is expected to
      be played only on speaker device, when WFD session is active.
    - The cause for this is, device is made to PROXY device in WFD
      session, when strategy is SONIFICATION.
    - Fix is to select speaker device during WFD session when strategy is
      SONIFICATION.

    Change-Id: I57895df5ecfc090584969b3d6eba1e9431270839
    CRs-fixed: 455883

commit 187fde4a04b930bbd349287c137d3c0878fd0a7b
Merge: 13d144a 0ab86af
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Fri Mar 1 03:15:42 2013 -0800

    Merge "alsa_sound: Support for applying independent channel volume control"

commit 13d144ace02b37002b3fed33ec973f0801091865
Merge: 0c64b41 9c37bad
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Feb 28 15:33:22 2013 -0800

    Merge "alsa_sound: Initialize the mPfdProxy in constructor"

commit 0c64b414a026a0164b97a8c1e4d63bdbe023a10e
Merge: 1ea750a 1cfa1ac
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Feb 28 15:32:49 2013 -0800

    Merge "libalsa-intf: Fix build warning from ALOGE"

commit 1ea750a8e740a73b01e852248ac42ac0ae9c1331
Merge: 6a3357d eb56b2d
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Feb 28 15:32:46 2013 -0800

    Merge "alsa_sound:Resample proxy ouput for other than 48KHz samplerate"

commit 6a3357db4a18a5a7eae7048802e3e9b7e3edf3d8
Merge: e88ae7d 1b6657f
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Feb 28 03:48:04 2013 -0800

    Merge "alsa_sound:Fix for Audio leak in LPA while in voice call"

commit 722337ab637bd82c748f048a0763f8f914ba16c0
Author: Haynes Mathew George <hgeorge@codeaurora.org>
Date:   Wed Feb 27 16:13:26 2013 -0800

    alsa-sound: EOS notification if drain ioctl is interrupted

    If the drain ioctl returns -EWOULDBLOCK, post an EOS ack only if
    the client has signaled EOS

    Change-Id: I9a9bd1b3702f62451032e4b1a4c2113501f42b44

commit 9c37bad94b377023efcf57f33bb9e52ae524a73b
Author: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Date:   Mon Dec 10 19:45:07 2012 -0800

    alsa_sound: Initialize the mPfdProxy in constructor

    The uninitialised variable of the proxy parameter PfdProxy in the
    ALSADevice constructor is initialized to avoid unnecessary call to
    sys_close::lib_close without the proxy session enabled.

    Change-Id: I8ef6cdae95a4f6ecee5c68f3e5ab8c85f5fb04c0
    CRs-Fixed: 428656
    (cherry picked from commit 6d812e2cba18aba1c40159b5e5194d225534235a)

commit 99dfda5b9d480f2013889dfe07939c1060094235
Author: Giulio Cervera <giulio.cervera@cyanogenmod.org>
Date:   Thu Feb 14 12:26:00 2013 +0100

    fmradio: enable STE radio

    Change-Id: Ieb666f1f0cdd267f35157d9663491ccc55ba2be5

commit 1bac86c761664f85bdfae8a8243063d244a9798d
Author: Giulio Cervera <giulio.cervera@cyanogenmod.org>
Date:   Wed Feb 27 22:21:28 2013 +0100

    audio: cm build fix

    Change-Id: I532a152848b4497b6969a2a9f96117a806af8633

commit e88ae7de1e48831b5bf1ea34873186a516406aef
Merge: 4010138 a803776
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Feb 27 02:42:30 2013 -0800

    Merge "libalsa-intf: Add mono headset support"

commit b39e39b11cdd3c33fdc154181306b8025f05d85a
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Tue Feb 26 16:45:47 2013 -0800

    audio: fix input buffersize for AMR/EVRC formats

    - Compute input buffer size for AMRNB/AMRWB/EVRC/
      EVRCB/EVRCWB audio formats based on sample rate,
      and channel count
    - Use default input buffersize for invald audio
      formats

    Change-Id: I36fa35a2d128fd05d2a956357b89c1e3acba48d3

commit 3093973cd81443b52560bcc074c55aa472f01f84
Author: Vidyakumar Athota <vathota@codeaurora.org>
Date:   Tue Feb 26 16:44:59 2013 -0800

    alsa_sound: Add support for VoLTE over USB headset

    -Currently VoLTE calls are not routed to USB HEADSET
    -Start USB audio thread when VoLTE call is active

    Change-Id: I87615f5e0b6373e8364db99430fe887f4a3f23ee

commit e3471f4a1b7f4e8b717a65329a9b3b8177048c15
Merge: 3f6c449 614a4e5
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Feb 26 15:47:41 2013 -0800

    Merge "alsa_sound: Fix to prevent additioal doRouting call"

commit 614a4e57bf227b826afe0dbb55ea89471634b612
Author: Vicky Sehrawat <vickys@codeaurora.org>
Date:   Mon Feb 25 19:49:45 2013 -0800

    alsa_sound: Fix to prevent additioal doRouting call

    - During an MO VoLTE call if BT is connected then
      no sound was heard on either side.
    - When the Call state was updated for VoLTE,
      doRouting was called, which tried to set BT with
      A2DP profile as the device if BT was connected.
    - Added a check to call doRouting only when an
      ACTIVE or HELD call's state is being changed.

    Change-Id: I8422c024e52abd99836cadd15a2d5df9d36f2eea
    CRs-fixed: 446294

commit 4010138d09c21531a0d56805f541f666bf27027d
Merge: 3f6c449 1297a73
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Feb 26 07:29:29 2013 -0800

    Merge "alsa_sound: Fix to prevent additioal doRouting call"

commit 1297a7396bab26afb40b1267c6ef880d3965ebe4
Author: Vicky Sehrawat <vickys@codeaurora.org>
Date:   Mon Feb 25 19:49:45 2013 -0800

    alsa_sound: Fix to prevent additioal doRouting call

    - During an MO VoLTE call if BT is connected then
      no sound was heard on either side.
    - When the Call state was updated for VoLTE,
      doRouting was called, which tried to set BT with
      A2DP profile as the device if BT was connected.
    - Added a check to call doRouting only when an
      ACTIVE or HELD call's state is being changed.

    Change-Id: I8422c024e52abd99836cadd15a2d5df9d36f2eea
    CRs-fixed: 446294

commit 1b6657fb950c886ab2ec88d65603a4c378c0f77e
Author: Vidyakumar Athota <vathota@codeaurora.org>
Date:   Mon Feb 25 17:12:15 2013 -0800

    alsa_sound:Fix for Audio leak in LPA while in voice call

    Issue: When stereo headset inserted and voice call is in BT,
    LPA/Tunnel playback route to headset for a second before it
    routes to BT.
    RootCause: LPA/Tunnel player starts session with the device
    that has been selected from media strategy.
    Fix: Follow Phone strategy for Music also if in voice call.

    Change-Id: If30ed07b50c1201ec055b8c939605f2cd3b4d69d
    CRs-Fixed: 450628

commit 3f6c4495c060b86ba31cd3e501449b2bfb1d4ff2
Merge: ef8a10a 9e244fc
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Sat Feb 23 04:36:49 2013 -0800

    Merge "alsa-sound: Avoid use of extra bytes to signal seek processed"

commit eb56b2dc2c65adaf3bd06b80b8579f7b4c5d5b7e
Author: Vidyakumar Athota <vathota@codeaurora.org>
Date:   Thu Nov 29 13:14:34 2012 -0800

    alsa_sound:Resample proxy ouput for other than 48KHz samplerate

    -Issue:If USB samplerate is other than 48kHz then there
     is an issue of lower pitch observed. This is because proxy
     output is at 48kHz and there is a mismatch of sample rates.
    - Fixed by using resampler from 48kHz to requested samplerate

    Change-Id: Ia46b09f4856532ef4ca943f864769e5bcbcb8742

    Conflicts:
    	alsa_sound/Android.mk

commit ef8a10a4345b08a11bd8dd8579198573e9c86684
Merge: 8c9e621 6b52607
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Fri Feb 22 12:30:38 2013 -0800

    Merge "alsa-sound: Use errno to check for error code of ioctl()"

commit 9e244fcf91c73de822e23fde505862e91028a49d
Author: Haynes Mathew George <hgeorge@codeaurora.org>
Date:   Thu Feb 7 20:30:04 2013 -0800

    alsa-sound: Avoid use of extra bytes to signal seek processed

    - A few bytes were reserved in the buffer written by the AudioSessionOut
      client to indicate a seek has been processed and there
      is no need to skip it.
    - We won't need this method anymore and its expected the client
      take care of this.

    Change-Id: Idfdeb3d41320e3e332424c13638c66ed200c0fee

commit 8c9e62149879bb2964e6a877dcc6378c9b9d8218
Merge: 9d18f45 8ed7cc1
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Feb 21 21:10:27 2013 -0800

    Merge "alsa_sound: Fix usb headset record noise"

commit 1cfa1ac78d24c19d84c613511b18b8f1c43abb1f
Author: Phani Kumar Uppalapati <phanik@codeaurora.org>
Date:   Thu Feb 21 18:01:15 2013 -0800

    libalsa-intf: Fix build warning from ALOGE

    Fix build warning due to wrong use of ALOGE
    messaging.

    Change-Id: I6686b55037ec0c5551b14b66a26e6643fa01516b
    CRs-Fixed: 454730

commit 9d18f45180b8f1ea5a9b407001b5fc129a77f894
Merge: 78e0dde 9e5c3ef
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Feb 21 15:59:43 2013 -0800

    Merge "alsa-sound: AudioSessionOut optimization"

commit 78e0dde090747c3f37027440a0a2e75056d02cb0
Merge: 5b1a6c4 cb2fa2d
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Feb 21 08:08:12 2013 -0800

    Merge "alsa_sound: Fix for Audio routing to HDMI when BT connected"

commit 5b1a6c446276f6658ad104e252aa61039f18dd71
Merge: 60cc831 338813c
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Feb 20 21:37:38 2013 -0800

    Merge "alsa_sound: Set tunnel/lpa bit for playback over usb headset"

commit 9e5c3efe9e7c5cbe8f6f2100a9e4679e00263019
Author: Haynes Mathew George <hgeorge@codeaurora.org>
Date:   Wed Feb 13 11:54:56 2013 -0800

    alsa-sound: AudioSessionOut optimization

    - Skip memset(0) for tunnel mode playback in write()
    - For LPA playback, call memset(0) only for partial buffers

    Change-Id: Ia3cec99afa3bd9750298ebbe7aa6c5fc2ebed927
    CRs-Fixed: 444041

commit 60cc831a150eba183e07b0347272ac573a21e2ef
Merge: 3c31cec ae1de0f
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Feb 20 08:18:18 2013 -0800

    Merge "alsa_sound: Fix for no audio when switching between BT headsets"

commit 0ab86afb76cc01f15aa7f9485905684d9ba48b9a
Author: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Date:   Tue Feb 19 23:15:17 2013 -0800

    alsa_sound: Support for applying independent channel volume control

    Issue: The current volume sent to DSP is averaged between left and
    right channels. As a result with volume level as zero on one
    channel and non zero on the other channel results to playback on
    both channels
    Fix: send left and right volumes to DSP

    Change-Id: If0608adb5ca40730fb1a00cf5a08635a6eb48860
    CRs-Fixed: 449284

commit 8ed7cc178b54f3949dde857d0a58138f5c98a379
Author: Vidyakumar Athota <vathota@codeaurora.org>
Date:   Thu Jan 24 05:46:44 2013 -0800

    alsa_sound: Fix usb headset record noise

    -Issue: recording with USB headset has noise
    -Fix: Increase the period size to 2048 so that
     number of times that the poll event executed is
     reduced which inturn matches proxy writes to DSP
     in real time

    Change-Id: I5a566ce413f0c5c10cf30817b7b4095a2f45c6fa
    CRs-Fixed: 426421

commit cb2fa2dcdf579708d263f8ddc9a9cf3699c4a05d
Author: Vidyakumar Athota <vathota@codeaurora.org>
Date:   Thu Feb 14 18:36:19 2013 -0800

    alsa_sound: Fix for Audio routing to HDMI when BT connected

    Currently when BT is connected Audio routes to HDMI when
    HDMI is plugged in and expected behaviour is audio should
    route to BT only as BT has higher priority than HDMI.This is
    because route function is being called with HDMI device in
    openOutputStream. Removing the route call will fix the issue.

    Change-Id: I76933b7f19498023ad2739e6b0305d36a2c77bb6
    CRs-Fixed: 451268

commit 3c31cecc1815c5bb2e355d4f476360d7d40eec4c
Merge: d2279da 10b6ce4
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Feb 14 14:17:34 2013 -0800

    Merge "alsa_sound: Add Adaptive ANC devices"

commit d2279da103816ad828161e85a9c20b4f5f385d48
Merge: 66b6050 88b4903
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Feb 12 16:15:49 2013 -0800

    Merge "audio: add support for 8610"

commit 338813c7f25b5d00866ca4a2bc2ba6fc19167adc
Author: Krishnankutty Kolathappilly <kkolat@codeaurora.org>
Date:   Mon Feb 11 18:24:24 2013 -0800

    alsa_sound: Set tunnel/lpa bit for playback over usb headset

    -The tunnel/lpa bit was not being set in all scenarios when
     playback happens over usb headset.
    -This resulte in no audio when trying to play the next tunnel
     mode clip.
    -Fix by setting the lpa/tunnel bit if the device is usb and
     playback session is tunnel or lpa.

    Change-Id: I6721a508e034c836a4aae5cc98cebb6236ee14d0
    CRs-Fixed: 447677

commit 6b52607be19870db6c5d45da6eb2c0833a169bda
Author: Haynes Mathew George <hgeorge@codeaurora.org>
Date:   Fri Feb 8 15:08:24 2013 -0800

    alsa-sound: Use errno to check for error code of ioctl()

    - ioctl() calls by definition must return -1 on failure with
      the error type set in errno.
    - Therefore, the return value must not be used
      to check the error type for a failed drain ioctl().
    - Use errno instead.

    Change-Id: I4d6814bd0da678feaeef17bae83008d4c7376233

commit a803776711deb6a2e522438180cecefc9e353b8c
Author: Helen Zeng <xiaoyunz@codeaurora.org>
Date:   Tue Feb 12 10:28:46 2013 -0800

    libalsa-intf: Add mono headset support

    - Some customers use mono device such as mono headset or mono
      speaker in their platform
    - Add the mono headset acdb define

    CRs-Fixed: 443365
    Change-Id: I1ceba40e6329749bf73c366d8c4cbb76b7ca2830

commit 66b6050de693d6f792f1145318b107dd689f079f
Merge: 35802aa fa1cacb
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Sun Feb 10 20:46:31 2013 -0800

    Merge "hardware/qcom/audio: fix memory leak in alsa_ucm.c file"

commit 35802aaca2e2a3a0a721b969f087d5801045e94c
Merge: c260160 750a848
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Feb 7 21:08:01 2013 -0800

    Merge "audio: fix to play Skype calltone on earpiece instead of speaker"

commit 10b6ce4b1fdc158a4d6192bf358d52c62dc3edb8
Author: Damir Didjusto <damird@codeaurora.org>
Date:   Mon Feb 4 12:43:26 2013 -0800

    alsa_sound: Add Adaptive ANC devices

    Add support for AANC devices in Audio HAL.
    When AANC is needed, the HAL will select the AANC device
    in UCM.

    Change-Id: Ia92080664d0ac66467888e1f19e0d717579b76fd

commit c2601604135a823b61222362fe1dcb114a33f25c
Merge: 9393dbc 3184a32
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Feb 6 22:00:58 2013 -0800

    Merge "alsa-sound: Adjustment for tunnel mode buffer size"

commit 9393dbca076c9b0e98b30db44a789cefde22f8d7
Merge: 029fea4 6db504e
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Feb 6 02:26:42 2013 -0800

    Merge "AudioSessionOut: Fix to handle a single write() with an empty buffer"

commit 750a848ca368740b4e8b4018b9b6a045930fd1d7
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Tue Feb 5 16:03:21 2013 -0800

    audio: fix to play Skype calltone on earpiece instead of speaker

    - Google's default AudioPolicyManager uses earpiece for Skype
      call tones, remove the code to select speaker in phone
      strategy if not in call.

    CRs-Fixed: 421065

    Change-Id: I4c2421e081d3ab6f7a496bacff91732dfb3cbe9c

commit 029fea4a984ccfd21059e27d65b59438a1100d11
Merge: 10d6f1c 9cd649e
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Feb 4 17:59:42 2013 -0800

    Merge "libalsa-intf: Include the audcal header path"

commit 10d6f1c03e1c541b9e0add6339561655e9cfdc0f
Merge: 200b505 6c461b7
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Feb 4 17:59:35 2013 -0800

    Merge "libalsa: Fix voip voice calibration issue"

commit 200b505e348f5d5147285bdcd05193e86ecfbb1e
Merge: 7eb0ad3 9f3ab34
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Feb 4 17:58:43 2013 -0800

    Merge " Fix for MT call ringtone on USB headset"

commit 7eb0ad3a25adc486d622cd43485200507c16f8ea
Merge: d88ffa2 f131e62
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Feb 4 17:58:40 2013 -0800

    Merge "alsa-sound : Fix a deadlock"

commit 3184a324d6f4c8b37de2773e2997f0cfac887f43
Author: Haynes Mathew George <hgeorge@codeaurora.org>
Date:   Fri Feb 1 16:29:54 2013 -0800

    alsa-sound: Adjustment for tunnel mode buffer size

    - Make necessary adjustments to align the buffer size
      for tunnel mode playback to satisfy the following condition
    - Buffer size is a multiple of LCM(1,2,4,6,8)
    - Buffer size is aligned to 4k.
    CRs-Fixed: 447274, 442365

    Change-Id: Ieb188cba6945b96e99bad4ceba4b53112383fd57

commit ae1de0ff368573bcb45c0c719d4c0ca336b74b83
Author: Karthik Reddy Katta <akkatta@codeaurora.org>
Date:   Fri Feb 1 15:16:14 2013 +0530

    alsa_sound: Fix for no audio when switching between BT headsets

    - mExtOutStream is not initialized in openExtOutput() but set to
      NULL in closeExtOutput() while disconnecting BT device.
    - Initialize mExtOutStream in openExtOutput()

    Change-Id: I3ae5f20f0cce0b8a10878a574561ac32aa3e898b
    CRs-Fixed: 446922

commit 6db504e9c2883d8402eabfdd122bafdd84cd0f45
Author: Haynes Mathew George <hgeorge@codeaurora.org>
Date:   Mon Jan 28 18:04:45 2013 -0800

    AudioSessionOut: Fix to handle a single write() with an empty buffer

    - This fix is a workaround to exit cleanly in the case where the AudioSink
      write might give just a single zero length buffer to the driver.

    Change-Id: I68d023e6f0bd5b5e4f593a42b26c685e3405288b
    CRs-Fixed: 445182

commit d88ffa25abbadd2a95988dbf35741959dc4dfa15
Merge: 84ac67a f0792ac
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Jan 30 19:35:59 2013 -0800

    Merge "audio: Update sysfs node in audio daemon"

commit 84ac67ac846a41d62ec1feaf268cedc5ec254cdf
Merge: f0acbdf 7955ffc
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Jan 30 19:35:56 2013 -0800

    Merge "audio: update device for all outputs in setPhoneState"

commit 88b49034b6bc3002f4d85196ada6e6c9749b8f58
Author: Fred Oh <fred@codeaurora.org>
Date:   Wed Jan 30 15:43:19 2013 -0800

    audio: add support for 8610

    add alsa libraries into 8x10 intial build

    Change-Id: Icfe603a37ef03f7a6b9dcc013f2075185b228695
    Signed-off-by: Fred Oh <fred@codeaurora.org>

commit 6c461b749b83339bca16fbb0742133ec84df2a17
Author: Jayasena Sangaraboina <jsanga@codeaurora.org>
Date:   Tue Jan 29 11:09:34 2013 -0800

    libalsa: Fix voip voice calibration issue

    - check is added to send voice calibration for
      targets other than fusion, since acdb on APQ
      side is not used on fusion platform
    - For Voip uses case, this check is breaking
      the functionality for applying acdb tuning
      parameters
    - Fix the issue by allowing voice calibration
      for voip use case

    CRs-fixed: 446557

    Change-Id: I9662325a90b796fe0261f897920477ee8e709fc9

commit 9cd649e32b391da5a5016f31d755d89f14c01a27
Author: Jay Wang <jaywang@codeaurora.org>
Date:   Mon Oct 22 15:02:16 2012 -0700

    libalsa-intf: Include the audcal header path

    Change-Id: Ie2a80ed7a8d289d5e5b05cadb17b599342f075ca

commit f0acbdf253f7d46c6d3322ba13ca2aabfdd819f9
Merge: 58f8def 6f6620b
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Jan 29 16:55:01 2013 -0800

    Merge "audio: update voip input buffersize for 8974 target"

commit 58f8def102b4bfa5301a2c6ebfd7beef6315ac3e
Merge: d1b3519 3b3ffdc
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Jan 29 16:54:16 2013 -0800

    Merge "alsa-sound: AudioSessionOut synchronization fixes"

commit f131e621928f2643ef0a8e431cd8e998a908c4c5
Author: Haynes Mathew George <hgeorge@codeaurora.org>
Date:   Tue Jan 22 17:21:14 2013 -0800

    alsa-sound : Fix a deadlock

    - openOutputStream() deletes an object of type AudioSessionOut
      if the object initialization fails.
    - This leads to a deadlock as the AudioSessionOut::reset() tries
      to acquire the same lock held by openOutputStream.
    - Release the lock in openOutputStream before deleting the
      AudioSessionOut object.

    Change-Id: If763eded702a59828ae51adfa4dca23749bd7300
    CRs-Fixed: 442365

commit 6f6620bddc90c4d2137918636740d655b31df00d
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Mon Jan 28 13:01:25 2013 -0800

    audio: update voip input buffersize for 8974 target

    - use 320 for 8K voip input and 640 for 16K voip input

    Change-Id: I0d041a894456a4e1875f47eefc8430ce416d84ab

commit d1b3519d1d4ea3defad76f2d142c173ea6822a0c
Merge: 12bbc02 93601e9
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Sat Jan 26 11:17:29 2013 -0800

    Merge "alsa : check for valid verb index"

commit 12bbc02dc4905f7ebe297a93e07b9b2cac420481
Merge: 4678f31 6000eb5
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Sat Jan 26 11:17:11 2013 -0800

    Merge "alsa_sound: Add e2e multi-channel support on 8974."

commit 6000eb5776c3d335de9a0287194dde3af107dd8b
Author: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Date:   Sat Jan 5 18:02:44 2013 -0800

    alsa_sound: Add e2e multi-channel support on 8974.

    Add support for playing multi-channel content over HDMI
    if HDMI sink supports multi-channel.
    Get HDMI EDID from driver for Multichannel playback
    Set desired channel map for multi channel output

    Change-Id: I561c876af224b9a376d260d3fd14fa0cfa2a64b3
    CRs-Fixed: 433019

commit f0792ac8e1a1fc039a392e2f2778e6037cf4d3bd
Author: Ravishankar Sarawadi <crsaraw@codeaurora.org>
Date:   Fri Jan 25 10:34:46 2013 -0800

    audio: Update sysfs node in audio daemon

    Use newly created sysfs node for audio daemon to read ADSP
    subsystem state from. Re-using subsystem restart framework sysfs
    node for same purpose is not recommended as this sysfs node could
    be used for different purposes in further development. New node
    indicates ADSP state as ONLINE/OFFLINE.

    Change-Id: I8339960a6e041b629ef3a1ba3eb43cbfa7c3f7c2

commit 4678f31b62dd5396e853de1009a9913cd27fe1a6
Merge: a30da2e b3f7d30
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Fri Jan 25 04:28:21 2013 -0800

    Merge "audio: fix for setting fluence paramter issue"

commit 3b3ffdcb5e0498fb34720476870eb2411599106a
Author: Haynes Mathew George <hgeorge@codeaurora.org>
Date:   Sun Jan 13 23:21:21 2013 -0800

    alsa-sound: AudioSessionOut synchronization fixes

    - Modify synchronization between AudioSink and EventThread
    - Move the code block related to drain to a different function
    - Reset EOS flags in write()
    - Do not depend on the filled queue size to set mReachedEOS

    Change-Id: Ie8cd7b9ff9cab495f236f2cfa9c68a9f3a204fe4
    CRs-Fixed: 439892

commit a30da2eb554c452541cc9b72a7b75d38d08121d2
Merge: 70831d4 50938a2
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Jan 24 17:31:56 2013 -0800

    Merge "Merge commit 'android-4.2.1_r1' into Upgrade_branch"

commit 7955ffcee15698e15e39aebcfca246c70e5edf74
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Wed Jan 23 21:07:27 2013 -0800

    audio: update device for all outputs in setPhoneState

    - Change routing for all outputs is nessary in setPhoneState
      Update devices not only for primary outputs but also other
      outputs including direct, tunnel, HDMI and so on

    Change-Id: I4861e67c8aab9e39f83386c27b9a9e167befd347
    CRs-Fixed: 438651

commit fa1cacba7e389e38a00e428dcdc9cefe89b04f2d
Author: Preetam Singh Ranawat <apranawat@codeaurora.org>
Date:   Tue Nov 20 15:57:27 2012 +0530

    hardware/qcom/audio: fix memory leak in alsa_ucm.c file

     - Free the current device if no valid device is found in
       set_controls_of_usecase_for_all_devices function.

    (cherry picked from commit c03520e5e6c9fe7ec35ddcb510a8ab53ebb023b5)

    CRs-fixed: 421414
    Change-Id: I759baaa8b3257950aa712fc2aaa28aa38a064a21

commit 9f3ab34d8910d8f152a8e5c4bf7114e097653f4f
Author: Vidyakumar Athota <vathota@codeaurora.org>
Date:   Wed Jan 23 04:03:18 2013 -0800

     Fix for MT call ringtone on USB headset

    - Issue:when usb headset connected MT call ringtone heard
      only on speaker and expected behaviour is ringtone should
      be heard on both speaker and usb headset.
    - Fixed by defining combo device usb proxy rx speaker

    Change-Id: I0474651ed6fa40636038dcd7b25c7962cecc7239
    CRs-Fixed: 443227

commit 70831d4bcd090987eaf4f27395983e22d3c4449e
Merge: 07cdbb8 f158b28
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Jan 21 18:15:03 2013 -0800

    Merge "audio: Add support for audio recording using EC"

commit 07cdbb80206609e22758904cf89270ce5bc8f464
Merge: 1be95c0 eb33733
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Sun Jan 20 07:41:36 2013 -0800

    Merge "Update copyright to The Linux Foundation" into jb_2.5

commit 1be95c081b8532cc497e204f6403e97b8b40e3f4
Merge: 698d03d 5282ade
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Sat Jan 19 16:34:18 2013 -0800

    Merge "audio: fix for audio recording"

commit 5282ade2cb9483548f16e9ee0e2d6c602865bfce
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Thu Jan 17 17:08:17 2013 -0800

    audio: fix for audio recording

    - Read SSR system prop only for surround
      sound recording use case.
    - Change input buffer size for 8974 to 512 bytes

    Change-Id: I70064769e92149cb6675a92018fb35533900e65f

commit 50938a2c0cea7e9af329a2b44cce55807da627a8
Merge: 698d03d 014d19f
Author: Shruthi Krishna <skrish@codeaurora.org>
Date:   Thu Jan 17 11:46:05 2013 -0800

    Merge commit 'android-4.2.1_r1' into Upgrade_branch

    * commit 'android-4.2.1_r1':
      alsa_audio: route call voice to phone when on HDMI
      alsa_sound: fix for SIP call mute issue.
      Revert "alsa_sound: fix for SIP call mute issue."
      alsa_sound: fix for SIP call mute issue.
      alsa_sound: fix for output device routing during video chat
      audio: add the headset tx
      hardware/alsa_sound: Change the device disable sequence

commit eb33733b669d02a28ea1bdbcb5436fc3647ee03e
Author: Duy Truong <dtruong@codeaurora.org>
Date:   Thu Jan 17 10:33:22 2013 -0800

    Update copyright to The Linux Foundation

    Change-Id: I482b7fd6b5502b86ca095a1b7bce4f03c10c9304

commit 698d03ddf93e65c017a170639a7035359fcaea7a
Merge: ee1b40b 783a7b9
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Jan 16 23:47:28 2013 -0800

    Merge "audio: enabled SSR in HAL"

commit f158b2809a2ca0d8b0792f94ef63c6b10489f038
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Tue Jan 8 15:09:00 2013 -0800

    audio: Add support for audio recording using EC

    - Add Support for Echo cancellation with Fluence topolog
      when fluence option selection from QC settings and num
      channels equal to one
    - with Fluence option selection from QC settings and num
      of channels selected from sound recording is greater
      than one, apply Audio Topology without EC.

    CRs-fixed: 437475
    Change-Id: I583eec7eeba6b1acb1d35e249c2a19229c364088

commit 783a7b97816b9f59fc2cceb2da7196f15030ea09
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Sun Jan 6 17:04:08 2013 -0800

    audio: enabled SSR in HAL

    - Enable surround sound recording in HAL

    Change-Id: Ibcbda82166240d613e4302488580cfdb55827d1b

commit ee1b40bc8f91dd8d1e35c0d838117dfe4b268d83
Merge: 19e43a2 a309f94
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Sat Jan 12 03:16:27 2013 -0800

    Merge "alsa_sound: Decrease the playback buffer size and latency"

commit b3f7d3010a068496e1f8bc87ccc53071359c65a6
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Fri Jan 11 16:38:24 2013 -0800

    audio: fix for setting fluence paramter issue

    - allow changing fluence type to "fluence" if
      related system prop is set

    Change-Id: I35ec50c6dcc1a7e47df253969ce91ce841b21da2

commit a309f948e737fd87e9af15486f92a4e3bc7e6291
Author: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Date:   Tue Jan 8 10:05:33 2013 -0800

    alsa_sound: Decrease the playback buffer size and latency

    Increased latency and buffer size are required for deep buffer
    output. Since deep buffer is not supported reducing the playback
    buffer size and latency

    Change-Id: I240bb69bfdadb7362dfee2c1d2a6fa8e4459a34a
    CRs-Fixed: 438177

commit 19e43a21585bacbb42b92eae05ee8651bc5714e1
Merge: bd29f00 92abe66
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Fri Jan 11 01:11:54 2013 -0800

    Merge "alsa_sound: Subsystem Restart Changes"

commit bd29f0065d884a6962de7c9de40b1fb8701c391d
Merge: fc3addc 1dca4f2
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Jan 10 01:38:37 2013 -0800

    Merge "alsa_sound: Fix audio not routed from Speaker to HDMI device"

commit fc3addc559c275caf43e73ff84c7e1eed76dd679
Merge: 29cebf6 b7a8ca6
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Jan 10 01:38:32 2013 -0800

    Merge "alsa_sound: conditionally allow changing of fluence type"

commit 29cebf69f8e75f3e86c51e141a18fd9a81048892
Merge: 7050617 c410e12
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Jan 10 01:38:29 2013 -0800

    Merge "audio: Add new Audio daemon process"

commit 7050617faa0977d402e784dadad7d8d5831519d3
Merge: 694cf22 9b79f6d
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Jan 9 20:22:19 2013 -0800

    Merge "audio: invoke CSD client APIs only for Fusion3 platform"

commit 92abe66ff64a428bd30641f5957110b62cb2e42e
Author: Ravishankar Sarawadi <crsaraw@codeaurora.org>
Date:   Fri Dec 28 18:45:52 2012 -0800

    alsa_sound: Subsystem Restart Changes

     - Add new member variable to indicate Subsystem Restart(SSR)
       in HAL
     - For PCM playback/capture SSR usecase, wait for WRITE/READ to
       return EIO error and then re-open PCM handle as part of recovery.
     - For Tunnel mode playback SSR usecase, EOS is sent to Media
       player on SSR which in turn calls destructor for ongoing tunnel
       session.
     - Modify device switch to handle SSR case. Device switch is called
       for same device as previous for SSR case to re-apply AMIX
       controls. Reason for this is codec reset called as part ADSP SSR
       handling which invalidates AMIX controls for current usecase.

    Change-Id: I9e1727c481c57d0218b3fb3cf2fbfb42952203e7

commit 9b79f6d8d1e7b59314db15e4f4946f960b3e244c
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Wed Jan 9 15:43:18 2013 -0800

    audio: invoke CSD client APIs only for Fusion3 platform

    - CSD client APIs are only used for Fusion3 platform.
      Always check platform_is_Fusion3() before invoking CSD APIs
    - Miscellaneous fix for QCOM FM and ANC flags
      in AudioPolicyManagerALSA

    Change-Id: I549b602cd782b737bf75a69e5cabd776547aa0ca

commit b7a8ca6aa446c20370561569a5b5167a7fed4b42
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Thu Jan 3 16:28:05 2013 -0800

    alsa_sound: conditionally allow changing of fluence type

    - If fluencetype=fluencepro, allow changing fluence type
      to "fluencepro", "fluence", or "none"
    - If fluencetype=fluence, allow changing fluence type
      to "fluence", or "none"
    - If fluencetype=none, do not allow enabling fluence

    Change-Id: Ie7814de601feb2a9e7789b0406fea52413e63d40

commit 93601e9e20d8f94dc8b938f3d50d343d972319c4
Author: Haynes Mathew George <hgeorge@codeaurora.org>
Date:   Tue Jan 8 17:42:44 2013 -0800

    alsa : check for valid verb index

    - Check for valid verb index before indexing into the use case list
    - This is to prevent an invalid lookup with a verb_index value of -1

    Change-Id: I4676bd0c9b00364c185e77120744b39d8a675acb
    CRs-Fixed: 436857

commit 1dca4f2543e03dd60914ca427a167002f2b8aa33
Author: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Date:   Mon Jan 7 16:00:03 2013 -0800

    alsa_sound: Fix audio not routed from Speaker to HDMI device

    Issue: Audio is not routing to HD TV on Plug-out and Plug-in of
    HDMI cable during audio or a/v decode.
    Fix: Remove the routing to Speaker when HDMI is plugged out

    Change-Id: I400c839a486ccb28b6501622cbbfed038ce911aa
    CRs-Fixed: 429760

commit 694cf2255f234664bd8151fcd4e75a5bd54d3d49
Merge: a28e0c8 1ec163f
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Jan 8 18:08:19 2013 -0800

    Merge "alsa_sound: Miscellaneous fixes BT and USB playback"

commit a28e0c8d591d0fe53b0806b7a00b827e8cf59f90
Merge: b4173c4 b8851fa
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Tue Jan 8 18:08:05 2013 -0800

    Merge "audio: Fix delay with first voice call and MT voice call"

commit 1ec163fc1d7c6c6d9287d63159d4af5822b3cf30
Author: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Date:   Fri Dec 21 18:12:05 2012 -0800

    alsa_sound: Miscellaneous fixes BT and USB playback

    1. Fix for MO call delay during LPA playback
    Fixed this issue by setting dealy to zero device is same as
    previous device
    2. Fix for no audio over BT after BT ON/OFF
    Initilize ExtOutStream whenever doRouting is called
    with Device BT/USB and also in AudioSessionALSA and
    AudioStreamOutALSA
    3. Avoid dead-lock in A2DP/USB
    4. Fix for music mute after BT MO call
    A2DPsuspended=false keypair has been taken care in setparameters
    5. Fix for BT no audio while device switch
    Fix this by storing previous output device, stop and start a2dp
    playback only when device switch happens from combo device to
    normal or otherway
    6. Fix for no audio in a2dp playback
    Fixed by acquire common lock before mExtOutCv signal has been
    issued in startPlaybackOnExtOut function.
    7.Fix to avoid MT call ringtone to earpiece
    Fix this issue by selecting speaker in Phone strategy if not
    in call.
    8.Failed write deadlock in A2DB/BT_SCO
    Fix:Put the write thread to sleep if write fails so that A2DP
    stop get the lock

    Change-Id: I83c7a0849899bfad6a47a3e347a91122e86f82a2

commit b8851faf6f6ec27e87a845fce5160a9fad1d2688
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Sun Jan 6 20:50:34 2013 -0800

    audio: Fix delay with first voice call and MT voice call

    - Add csd init and deinit to HAL CTOR and DTOR
    - Add conditional check for voice acdb loader
    - Reducing the delay for playing the RINGTONE
      before voice call is accepted saves,appr 340ms.
    - on this platform reducing the delay wont affect
      the RINGTONE as the delay in setting up voice
      path after accepting the voice call will
      compensate RINGTONE buffers with kernel and
      firmware played on the device.

    Change-Id: Ic69bfa38cfbe659ce09aa62691eae45982a42a76

commit b4173c4d8dc0a0d529b6cdd2e2ec3fa63039f4e2
Merge: dcc0fc2 a5a6a80
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Jan 7 16:17:04 2013 -0800

    Merge "alsa_sound: Collection of fixes in HAL for EOS"

commit dcc0fc27c21aee82d548c76530ae247be17da373
Merge: b652d20 4526fde
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Jan 7 16:17:03 2013 -0800

    Merge "alsa_sound: Decrease the tunnel buffer size to 256Kbytes"

commit b652d209fd54208a2c7d7df4fa40b197834ece3d
Merge: ea44f15 fddaa94
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Jan 7 16:16:54 2013 -0800

    Merge "audio: add support for FM feature"

commit c410e12c9e7f9bab37f0a99460dad49c5613dacf
Author: Ravishankar Sarawadi <crsaraw@codeaurora.org>
Date:   Thu Jan 3 15:28:38 2013 -0800

    audio: Add new Audio daemon process

     - Add new audio daemon process.
     - Pupose of this daemon is to monitor and report changes ADSP
       processor state to audio framework/Audio flinger.
     - Specific sysfs node file on device indicates ADSP processor
       state as ONLINE or OFFLINE

    Change-Id: I747ff9b86296be3eec72309728b48cfab841acf8

commit a5a6a8038667f4ec3747c81c43c1ba910dc6eaef
Author: Haynes Mathew George <hgeorge@codeaurora.org>
Date:   Wed Jan 2 21:20:16 2013 -0800

    alsa_sound: Collection of fixes in HAL for EOS

    -Handling of EOS, and triggering EOS was wrong
    in TunnelPlayer. Seeking when EOS was posted
    to the HAL was wrong. EOS should
    not be posted till seek is complete
    -Also, EOS should not be posted to
    the app if we are seeking
    -Player should wake up when seeked,
    even after EOS was posted from player
    to HAL
    -Fixed this issue by cleaning up the code

    Change-Id: Ia5dc43e0482bb7c714a7269574b5eed34efd6637
    CRs-Fixed: 433346
    CRs-Fixed: 432233
    CRs-Fixed: 429868

commit 4526fdee325b36bfb36835410abbf9af0920aade
Author: Phani Kumar Uppalapati <phanik@codeaurora.org>
Date:   Fri Dec 14 14:14:31 2012 -0800

    alsa_sound: Decrease the tunnel buffer size to 256Kbytes

    Decrease Tunnel buffer size to 256Kbytes.

    Change-Id: Ib47d7551af8c3856e145d131010cc5fd2c218ce3

commit fddaa948a5ff7c74f34ab3cd55e04b55e2ccd609
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Thu Dec 27 17:42:46 2012 -0800

    audio: add support for FM feature

    - Enable FM flags in audio HAL
    - Use new FM device enums from system/core
    - Fix device_conv_table for new device and legacy
      device mapping
    - Update AudioPolicyManagerALSA to support FM

    Change-Id: I11d72035f0f52dd69eafe0a2a081cfc3b9ab26c5

commit ea44f15b043e0822719cc4932fe23fa4130358f0
Merge: 38fa643 ebe8557
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Jan 3 17:14:37 2013 -0800

    Merge "alsa_sound: Fix for no audio in voice call over USB"

commit 38fa64365ae047bf72ce5e3e32e0fcd0f3a3239b
Merge: a0c8418 9acf3d9
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Jan 3 17:14:37 2013 -0800

    Merge "alsa_sound:Fix for no audio while playback on USB headset"

commit a0c8418ccb688ca2a75d97d445afbaccdbf53225
Merge: 3af13f6 2759821
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Jan 3 17:14:36 2013 -0800

    Merge "alsa_sound: Fix for device switch crash"

commit ebe85572f2f91a831c531ccb0db6a031701d4893
Author: Vidyakumar Athota <vathota@codeaurora.org>
Date:   Thu Dec 27 19:17:12 2012 -0800

    alsa_sound: Fix for no audio in voice call over USB

    -Issue: No audio in voice call over USB headset
    -Rootcause:Phone strategy selects USB headset only
     if mPhonestate is not in IN_CALL mode
    -Fixed by update the USB headset device in phone strategy
     irrespective of mPhonestate value

    Change-Id: I111dc7e4225d10074ae55e841aebec8a7568cc64
    CRs-Fixed: 434752

commit 9acf3d96f4e904c6b8dbf1c2a30b4ee22b87d80f
Author: Vidyakumar Athota <vathota@codeaurora.org>
Date:   Fri Dec 7 11:05:01 2012 -0800

    alsa_sound:Fix for no audio while playback on USB headset

    - Currently LPA/Tunnel usecases have been reset in
      AudioStreamOutALSA file which is not correct.
    - Fixed by moving the reset logic to AudiosessionOut file
    - Also fixed no-audio issue when device switch happens between
      BT and USB headset

    Change-Id: I4873d50d4a6cb40a2c5ee9e3387cc63c5ac447aa

commit 27598218770793339bbf23b9c383589e8f937c75
Author: Vidyakumar Athota <vathota@codeaurora.org>
Date:   Thu Dec 27 15:15:42 2012 -0800

    alsa_sound: Fix for device switch crash

    -Issue:Native crash seen while device switch during
     LPA/Tunnel playback
    -RootCause: Close output is being called on direct output
     during device switch
    -Fixed by removing this call

    Change-Id: If3de7f8549abc718ae27641c88464676b5cb1779
    CRs-Fixed: 434708

commit 3af13f6ab94ed77aff2651f33d080abee3e45da8
Merge: 2f99066 27c9030
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Fri Dec 21 22:45:45 2012 -0800

    Merge "alsa_sound: Add support for 8 channel for HDMI device"

commit 2f990669f55226433874a824ae0d96567e9b6cb2
Merge: b144fd8 ebd7ad8
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Fri Dec 21 21:25:55 2012 -0800

    Merge "alsa_sound: Miscellaneous fixes for LPA and Tunnel Mode playback"

commit 27c9030c717e14155600755a0533da73588ea5e8
Author: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Date:   Wed Dec 19 20:14:54 2012 -0800

    alsa_sound: Add support for 8 channel for HDMI device

    Playback of upto 8 channels is supported over HDMI device
    Add support to enable the 8 channel on HDMI device

    Change-Id: I2314f2ab47fd47c8d22c3d0b29e60724711f43ff

commit ebd7ad8fb7b5836a42987e7f933894bd758b18aa
Author: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Date:   Tue Dec 18 21:50:46 2012 -0800

    alsa_sound: Miscellaneous fixes for LPA and Tunnel Mode playback

    Fix for, "Music getting played after onpausetimeout"
    flushing the buffers in case of onpausetimeout.

    Change-Id: Ibbe146cb83ce864c86fdd137c35ad6545e83cdf4

commit b144fd8fb572a260c79755c15a700bb6ac102a5b
Merge: 727610a 76aa4c9
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Dec 19 23:46:56 2012 -0800

    Merge "audio: Fix for usb headset recording issue"

commit 727610a0b93234ec37248828fecda6df30fe0e23
Merge: 8dacf2a 65860e4
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Dec 19 20:17:20 2012 -0800

    Merge "audio: compile mm-audio project"

commit 8dacf2a61ce780eadde7395471f6d4a9fc073ec2
Merge: 74f4b09 d3d2c86
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Dec 19 16:58:59 2012 -0800

    Merge "audio: Fix for TTY mode parameter values in HAL"

commit 76aa4c9792e584a02cf9080debaac712b0b04880
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Tue Dec 18 20:47:28 2012 -0800

    audio: Fix for usb headset recording issue

    - Add DEVICE_IN_ANLG_DOCK_HEADSET in device_conv_table
      so it can be converted from HAL API 2.0 to legacy
      device enumus

    CRs-fixed: 432592
    Change-Id: I627868305774267477489a12a4cebd7cef84bd7b

commit 65860e43c4cee5bc87baca284526ba263ef82513
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Mon Dec 17 17:40:30 2012 -0800

    audio: compile mm-audio project

    - Compile mm-audio project to use QCOM
      audio encoder/decoders

    CRs-fixed: 433215
    Change-Id: I03aea3d113c065c499031182d3730aa3ff012fe5

commit d3d2c86ba5edeeb5499a058b52568305fa49b622
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Mon Dec 17 15:56:43 2012 -0800

    audio: Fix for TTY mode parameter values in HAL

    - TTY audio parameter values in HAL must match
      the values set by PhoneApp
    - Support TTY mode parameters with and without
      "tty_" prefixes.

    CRs-Fixed: 432843
    Change-Id: I2441b41c9dabe670f86cc4485c11318739c71e86

commit 74f4b096cac3813bf53e2eea574185ebf18e4068
Merge: 2a079c7 4d33ffd
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Dec 13 23:29:25 2012 -0800

    Merge "alsa_sound: Re-design USB audio solution"

commit 2a079c7fd5ac0b411edb962fd4a39814cfc18431
Merge: b27edcd 979666c
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Thu Dec 13 23:29:23 2012 -0800

    Merge "alsa_sound: update AudioPolicyManagerALSA for QCOM audio features"

commit b27edcdcf612e0cab82a6d99704795aaa5ecb779
Merge: cab5ad8 9d40ec5
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Dec 12 21:41:23 2012 -0800

    Merge "alsa_sound: Modify playback and capture period size"

commit cab5ad81853cf4f057e136cc4380eda5786987db
Merge: 6ffb4f9 135a23d
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Wed Dec 12 21:41:21 2012 -0800

    Merge "HAL: make changes to compile audio hal for 8974."

commit 4d33ffdc2ba482839a994397056b3265cd2d281e
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Mon Dec 10 00:11:29 2012 -0800

    alsa_sound: Re-design USB audio solution

    - Add support for AOA USB audio feature
    - Currently there is no audio support for usb accessory/device
      modes. Added support for the same
    - Redesign proxy to support USB accessory/device also

    Change-Id: If9c975ed2d782adb0a0055fe41cfcc16fa64b59e

commit 979666c84b10d5008a2e2748a55594ba5165099b
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Sun Dec 9 23:22:36 2012 -0800

    alsa_sound: update AudioPolicyManagerALSA for QCOM audio features

    - Update USB audio, VOIP, a2dp support in AudioPolicyManagerALSA
      based on JB AU04.01.01.00.303
    - Move QCOM changes AudioPolicyMangerBase to AudioPolicyManagerALSA
      to minimize base class changes

    Change-Id: I9a43b11798f26385111d2ccdbff125b705525e8b

commit 9d40ec51d900dd343ecc13151ef1d9ce8ec55a61
Author: Phani Kumar Uppalapati <phanik@codeaurora.org>
Date:   Sat Dec 8 19:10:46 2012 -0800

    alsa_sound: Modify playback and capture period size

    Modify playback perid size to 2048 bytes and capture
    period size to 512 bytes.

    Change-Id: I2f76ba348ac6d53784641ddb5d9a27f9bdca12ce

commit 6ffb4f91882e4e69d36cfa215aabdc689839a2c0
Merge: 2acdb4f 908253a
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Mon Dec 10 17:12:16 2012 -0800

    Merge "alsa_sound: Support LPA Playback"

commit 908253afa7901e319c3a9f1be6327d746326fc90
Author: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Date:   Fri Dec 7 11:57:25 2012 -0800

    alsa_sound: Support LPA Playback

    Add support to enable Playback in LPA mode

    Change-Id: I19721738cf8a6c4c8df56b02deaaad7a2172d472

commit 2acdb4fcec1e027d49fb3eb814c7cc07bf25eb9f
Merge: 6502311 22d0d6f
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Sun Dec 9 18:53:50 2012 -0800

    Merge "alsa_sound: Support Tunnel mode of playback"

commit 6502311a86ee379af5d8ffdb419d4dc23daf3793
Merge: bbd94ad 868131e
Author: Linux Build Service Account <lnxbuild@localhost>
Date:   Sun Dec 9 01:53:58 2012 -0800

    Merge "mm-audio: intial commit of mm-audio"

commit 868131e11d7b0bfdce39e0b046bf623510136020
Author: Mingming Yin <mingming@codeaurora.org>
Date:   Wed Dec 5 16:54:01 2012 -0800

    mm-audio: intial commit of mm-audio

    - Add QCOM audio decoder and encoder
      components including: EVRC, QCELP,
      AAC, AMR, AMRWB, and MP3

    Change-Id: I33359c141c7a408425530171c6545e87383892ad

commit 135a23dce451f751100f3a075ddb102713fe837e
Author: Harmandeep Singh <hsingh@codeaurora.org>
Date:   Sat Dec 8 19:00:44 2012 -0800

    HAL: make changes to compile audio hal for 8974.

    Make changes to compile audio hal for 8974.

    Change-Id: Ibd80d4bab6342b6e5456eeced4f732e697fbd175

commit 22d0d6fe1c3ca2c24d74565fe55c69607d733378
Author: Subhash Chandra Bose Naripeddy <snariped@codeaurora.org>
Date:   Sat Dec 1 23:03:41 2012 -0800

    alsa_sound: Support Tunnel mode of playback

    Add support for Tunnel playback

    Change-Id: I161f4faf356f406fb6372bd1c28bf6b34dd1d8c0

commit 014d19fa3d557cfc931b3dd5ecf8cdedc591d6ca
Merge: 9eb1a4f 9769041
Author: The Android Automerger <android-build@android.com>
Date:   Wed Oct 17 16:07:18 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit 97690417fba0fd40996babab0349c2ea6d002f6e
Merge: 89c224e 4fea7c7
Author: The Android Automerger <android-build@android.com>
Date:   Wed Oct 10 06:59:32 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit 4fea7c7372bec27a4489e511dcdd7f24bdd1fc2a
Merge: 2c79891 b7c7b51
Author: The Android Automerger <android-build@android.com>
Date:   Sun Oct 7 18:58:08 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit b7c7b51c85554174ee64499399e28db27eb2c6ff
Merge: 5f24fd9 9f99e38
Author: The Android Automerger <android-build@android.com>
Date:   Wed Oct 3 06:59:50 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit 9f99e382ef161dda2ff137ea8be80f26f150399e
Author: Sungmin Choi <sungmin.choi@lge.com>
Date:   Fri Sep 28 16:09:12 2012 -0700

    alsa_audio: route call voice to phone when on HDMI

    Currently call voice is routed to HDMI rx/tx when HDMI cable is plugging.  So
    call voice is muted.  MO/MT call voice routes to speaker/back-mic and
    communication routes to HDMI rx/back-mic when HDMI cable is plugging.

    Bug: 7218296
    Change-Id: I53fc4d5b77ad6502a32c4e8d06e17141e2aa4be2
    Signed-off-by: Iliyan Malchev <malchev@google.com>

commit 1af98a6183584b3f7254a4ddff6ca629cd77e037
Author: SathishKumar Mani <smani@codeaurora.org>
Date:   Wed Sep 26 20:23:34 2012 -0700

    alsa_sound: fix for SIP call mute issue.

    -Mute VOIP through audio path is not supported by kernel,
     so calling setVoipMicMute() to disable ""Voip Tx Mute"" fails
    -Mute VOIP by cleanning buffers read if setMicMute is set
    -Remove mVoipMicMute and use mMicMute for voice and VOIP

    Bug: 7213748
    Change-Id: I62be8456447425cbd1521083782802effcb6d326
    Signed-off-by: Iliyan Malchev <malchev@google.com>

commit cda1a832f4aac97b14d788bc742308e9d039e2e1
Merge: 80baa42 604ccc9
Author: The Android Automerger <android-build@android.com>
Date:   Fri Sep 28 06:59:33 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit 604ccc983c49e360ab60427a65d507fbc7672300
Author: Iliyan Malchev <malchev@google.com>
Date:   Thu Sep 27 21:29:09 2012 -0700

    Revert "alsa_sound: fix for SIP call mute issue."

    This reverts commit f859d3e80b790d4b525515918f1c6889383cc495.

    bug: 7250052
    reopens-bug: 7213748
    Change-Id: I33312c0be196b327887edc93c0b766bcf86a94bc
    Signed-off-by: Iliyan Malchev <malchev@google.com>

commit 5481e7629e04e4d50ac5b471d0dafd720c0a5241
Author: SathishKumar Mani <smani@codeaurora.org>
Date:   Wed Sep 26 20:23:34 2012 -0700

    alsa_sound: fix for SIP call mute issue.

    -Mute VOIP through audio path is not supported by kernel,
     so calling setVoipMicMute() to disable ""Voip Tx Mute"" fails
    -Mute VOIP by cleanning buffers read if setMicMute is set
    -Remove mVoipMicMute and use mMicMute for voice and VOIP

    Bug: 7213748
    Change-Id: Ie2a200470c16da2e4ae991c0814ec8dfb4666833
    Signed-off-by: Iliyan Malchev <malchev@google.com>

commit 341c0602f27fe7e3cc629a0b6ac1c2b5df256309
Merge: 7778038 453d78c
Author: The Android Automerger <android-build@android.com>
Date:   Thu Sep 27 06:59:52 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit 453d78c3bd4bd706bf381a82788e0ed88d2f96d0
Author: SathishKumar Mani <smani@codeaurora.org>
Date:   Fri Sep 21 20:12:08 2012 -0700

    alsa_sound: fix for output device routing during video chat

    Bug: 7163358

    - Output device for MODE_IN_COMMUNICATION could be earpiece
      or speaker, do not update output device in switchDevice()
      if input device is BUILTIN_MIC

    Change-Id: I2a97a63f4cb57ef114695d3fd917282612b7a346
    Signed-off-by: Iliyan Malchev <malchev@google.com>

commit 85f3fc5fd4b2b6ae79502c172148b9a27fda6fa5
Merge: 925596f 9a0ca06
Author: The Android Automerger <android-build@android.com>
Date:   Tue Sep 25 06:59:58 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit 9a0ca062f87d50bef085d112292e11b8d790ffbb
Merge: 639ce75 98913ad
Author: The Android Automerger <android-build@android.com>
Date:   Fri Sep 21 06:59:56 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit 98913ad4e326628372f2c2ca3ba6b774c336ee44
Merge: 9efed76 8b9ec98
Author: The Android Automerger <android-build@android.com>
Date:   Wed Sep 19 07:01:39 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit 8b9ec98dfd49dad9c11ca4a208e8aca0572639cd
Merge: 2aa2507 f9e4091
Author: The Android Automerger <android-build@android.com>
Date:   Tue Sep 18 06:59:54 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit f9e4091e036cc653c0d1af97b7b5c7b44e00f6d6
Merge: 5ff7a02 9ea52b2
Author: The Android Automerger <android-build@android.com>
Date:   Sun Sep 16 06:59:52 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit 9ea52b2d34e3ce5d52073be595945e6212c444b7
Merge: d4f2d06 308dc9a
Author: The Android Automerger <android-build@android.com>
Date:   Fri Sep 14 07:11:58 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit 308dc9a1a8969a10f5600f697870cfb87c22146d
Merge: 94de74c 92ce81d
Author: The Android Automerger <android-build@android.com>
Date:   Thu Sep 13 06:59:56 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit 92ce81d8aeb4a3af6af27e3db020fb90c8eb9b85
Merge: 0a01991 f742751
Author: The Android Automerger <android-build@android.com>
Date:   Wed Sep 12 06:59:53 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit f742751fd3fe3fe2ed0dfbed6278351b3c50d53a
Merge: 7801df9 39a9790
Author: The Android Automerger <android-build@android.com>
Date:   Sun Sep 9 07:00:39 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit 39a97909720475e814f8e3aa280c1468f09a021d
Merge: a4e2367 13e8749
Author: The Android Automerger <android-build@android.com>
Date:   Wed Sep 5 06:34:05 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit 13e874988aaca5bae568b9b14e6f62de136e3b9e
Merge: 30f8dc2 15defba
Author: The Android Automerger <android-build@android.com>
Date:   Wed Aug 29 06:33:54 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit 15defbaaf797973200e6156a0d00982aa039da4e
Author: ehgrace.kim <ehgrace.kim@lge.com>
Date:   Sun Aug 12 20:40:32 2012 -0700

    audio: add the headset tx

    Change-Id: If15ef8362dd2800c41fb945b96488844db29e5db

commit a6ec2c4e605e0c50342aa543337853ebbc5f38f3
Merge: a97e6f6 fb65fc8
Author: The Android Automerger <android-build@android.com>
Date:   Tue Aug 28 06:33:56 2012 -0700

    merge in jb-mr1-release history after reset to jb-mr1-dev

commit fb65fc8a29c7eb9e3ede46168df60c55943821b5
Author: Ajay Dudani <adudani@codeaurora.org>
Date:   Mon Aug 20 18:48:13 2012 -0700

    hardware/alsa_sound: Change the device disable sequence

    Before tearing down codec, disable the MDM device first so that
    slimbus channel is disconnected from MDM to avoid overflow on
    codec slimbus interface

    Change-Id: I20ef6fe97b3e72b6fe17243a55e4e20ed30df93c

Change-Id: I29b1135213bff560a35705ae4a8cdc52a3c14fbb
audio-caf: fixed derp in AudioHardwareALSA.cpp

Change-Id: I52ce0ebf3ac099b18c7baa98a60243c6dc7cc643
audio-caf: Fixed a couple more derps with the AudioHardwareALSA.cpp file

Change-Id: Id971db465246b5fae791b75e588e7e8116b0b0a7
audio-caf: total replacement, too many issues to fix them individually

Change-Id: If161964bb7cc54af0532251e51a669d47e525475