)]}'
{
  "log": [
    {
      "commit": "a636433cbd09c0708b85f337ef45c0cdef3bcb4d",
      "tree": "8a629f33d648d1ddd8b79305f02763c539fef7cd",
      "parents": [
        "9adb12830ac2da72c91b842d24725ced6d236b16"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Thu Apr 19 09:35:04 2012 -0700"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Mon Apr 23 16:54:41 2012 -0700"
      },
      "message": "Use C APIs instead of C++ APIs for policy\n\nThe C++ APIs are going away.\n\nNote: we use tid \u003d\u003d 0 which is not supported yet by the C APIs,\ndo not submit this until that is added.\n\nChange-Id: I0e90789e6c81c69f2544e899c52421ea5d1342be\n"
    },
    {
      "commit": "3acbd053c842e76e1a40fc8a0bf62de87eebf00f",
      "tree": "1d6dcc01039a1af770131c6b427efa27a2e5755f",
      "parents": [
        "f853499c6a088ef8506cacbb3ccdffa87e0cec3e"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Tue Feb 28 10:39:56 2012 -0800"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Sun Apr 22 14:50:01 2012 -0700"
      },
      "message": "Configure policy of mediaserver threads\n\nChange-Id: Ifd825590ba36996064a458f64453a94b84722cb0\n"
    },
    {
      "commit": "0ca3cf94c0dfc173ad7886ae162c4b67067539f6",
      "tree": "bc156e7822e663626b78ec2ad09ef3970fbb48c5",
      "parents": [
        "ee685f6d19a6cc3b0bf02e69eb5dea11a2aeb601"
      ],
      "author": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Wed Apr 18 09:24:29 2012 -0700"
      },
      "committer": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Wed Apr 18 15:04:46 2012 -0700"
      },
      "message": "rename audio policy output flags\n\nChange-Id: I27c46bd1d1b2b5f96b87af7d05b951fef18a1312\n"
    },
    {
      "commit": "4a4a0959bca78e03e3c3f486ba17829c28314d8c",
      "tree": "5fbfe5302ca28741483ae64d0ccf2630830f167e",
      "parents": [
        "a1472d9883e35edd280201c8be3191695007dfd4"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Mon Mar 19 11:38:14 2012 -0700"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Mon Apr 02 07:34:39 2012 -0700"
      },
      "message": "AudioTrack client fast policy\n\nChange-Id: I42ce691df3f586ac061b62237f35a263a0116f1f\n"
    },
    {
      "commit": "291f4d505aff81969e6666049d3cc3446f836af2",
      "tree": "f2ad96c1acfb08c190148f8f12a7f86825cd7728",
      "parents": [
        "ccf89b54f973f11082150d02ed957f7e967fbc8b"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Mon Mar 19 12:16:56 2012 -0700"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Wed Mar 21 15:41:02 2012 -0700"
      },
      "message": "Remove enforceFrameCount\n\nIt was only used to decide whether to issue a warning.\nThe warning was issued the first time track was created but\nnot at re-creation.  Now it is a verbose message every time,\nnot a warning since it happens all the time with key clicks on A2DP.\n\nChange-Id: I9d39f53c0a7eb84b666e55b1b76ff830cf8f37ba\n"
    },
    {
      "commit": "ea7939a079b3600cab955760839b021326f8cfc3",
      "tree": "570fb56838735a46a28e28a4f083d763f758ea24",
      "parents": [
        "eb76ff27c13386eb5a2dad73b1a8366f80eb0ff7"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Wed Mar 14 12:56:26 2012 -0700"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Mon Mar 19 17:53:33 2012 -0700"
      },
      "message": "Whitespace\n\nFix indentation, and add blank lines in key places for clarity\n\nChange-Id: I57a0a8142394f83203161aa9b8aa9276abf3ed7c\n"
    },
    {
      "commit": "18868c5db2f90309c6d11e5837822135e4a0c0fa",
      "tree": "58c7505f59a1a93ab729fc039aa5cb1eaa49652e",
      "parents": [
        "f2015b625cd6bea86da7401e725332c35b90e7cf"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Wed Mar 07 09:15:37 2012 -0800"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Fri Mar 09 15:32:22 2012 -0800"
      },
      "message": "Use audio_policy_output_flags_t consistently\n\nThis affects:\n - IAudioFlinger::openOutput\n - AudioTrack::AudioTrack\n - AudioTrack::set\n - apps that call these\n\nChange-Id: I26fb281bac6cb87593d17697bc9cb37a835af205\n"
    },
    {
      "commit": "d8d6185c978c9b27ca69e7791785e0983ed9e8b8",
      "tree": "cda49b71e787cf1bcf68671e237389f70706cff3",
      "parents": [
        "985ed9a1a22ec7e6e245d3fb8e93d3a23bdc539b"
      ],
      "author": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Mon Mar 05 17:06:40 2012 -0800"
      },
      "committer": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Mon Mar 05 17:18:36 2012 -0800"
      },
      "message": "AudioTrack: relax check on minimum buffer size\n\nCurrent AudioTrack implementation enforces that the requested audio\nbuffer size is at least corresponding the audio latency.\nThis requirement is too strong and leads to problems with current\nstagefright and AudioSink implementations when playing over output\nstreams with long latency.\n\nUltimately, the AudioSink design should be changed to specify a minimum\nbuffer size in time or frames units but not in buffer count units.\n\nChange-Id: I8ba603956f92ac49143a8249572665aa548f2f0f\n"
    },
    {
      "commit": "4ff14bae91075eb274eb1c2975982358946e7e63",
      "tree": "e9e54fddb9832d30b69c2a11c9ed2884397f2f95",
      "parents": [
        "951bd8d1ad9581a414e171ad8605a9515d0ad667"
      ],
      "author": {
        "name": "John Grossman",
        "email": "johngro@google.com",
        "time": "Wed Feb 08 16:37:41 2012 -0800"
      },
      "committer": {
        "name": "John Grossman",
        "email": "johngro@google.com",
        "time": "Thu Feb 16 13:45:11 2012 -0800"
      },
      "message": "Upintegrate Audio Flinger changes from ICS_AAH\n\nBring in changes to audio flinger made to support timed audio tracks\nand HW master volume control.\n\nChange-Id: Ide52d48809bdbed13acf35fd59b24637e35064ae\nSigned-off-by: John Grossman \u003cjohngro@google.com\u003e\n"
    },
    {
      "commit": "d9b9b8d09e7471b0ffa21cfa9f944ef4ad300a71",
      "tree": "e2b8666cc7ca81254ce264129ddd70c076a63ed1",
      "parents": [
        "ed15977476a3d53103866e6d527fa3fb65d4166c",
        "99e53b86eebb605b70dd7591b89bf61a9414ed0e"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Tue Feb 14 09:42:32 2012 -0800"
      },
      "committer": {
        "name": "Android (Google) Code Review",
        "email": "android-gerrit@google.com",
        "time": "Tue Feb 14 09:42:32 2012 -0800"
      },
      "message": "Merge \"Update comments\""
    },
    {
      "commit": "99e53b86eebb605b70dd7591b89bf61a9414ed0e",
      "tree": "8939550ada02fe21121550a83952ebf7638c0dac",
      "parents": [
        "6dbc1359f778575d09d6da722b060a6d72c2e7c5"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Thu Jan 19 08:59:58 2012 -0800"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Tue Feb 14 09:17:59 2012 -0800"
      },
      "message": "Update comments\n\nWe no longer put the filename at start of file.\n\nChange-Id: Ic435b159a23105681e3d4a6cb1ac097bc853302e\n"
    },
    {
      "commit": "38f5d71e72f3b76c5b519614d27f051d53cd2712",
      "tree": "a27ac85c22bc55b10bdeeaa3bddaae473325caaf",
      "parents": [
        "98ec94c5854daccc3474758524e7f4adfe535ce0"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Wed Feb 08 14:12:12 2012 -0800"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Wed Feb 08 14:12:12 2012 -0800"
      },
      "message": "Remove dead code AudioTrack::getLoop\n\nChange-Id: I868329c52f31bc20125f068500d8f892b4ec9796\n"
    },
    {
      "commit": "4f9b0c54011eb8fd2ccfb393c2dcd51cd07800e0",
      "tree": "318b28bb04ce0b4ce39f75f6a8dbac27cee1455c",
      "parents": [
        "86feafe15b3f9609e1e9f64184688c2b6f2e4834",
        "0e265cf36d201a7ccc0238b5c60b50f43d1dc450"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Wed Feb 08 08:34:33 2012 -0800"
      },
      "committer": {
        "name": "Android (Google) Code Review",
        "email": "android-gerrit@google.com",
        "time": "Wed Feb 08 08:34:33 2012 -0800"
      },
      "message": "Merge \"Remove dead mutex in AudioTrack/AudioRecord thread\""
    },
    {
      "commit": "a5224f319e2ba4b51ddb4287705ccf8d4b8ecc51",
      "tree": "eb36f63045d044c840072ad95398d49b743cd6bd",
      "parents": [
        "787bae0578fbaab6219ebf23494866b224d01438"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Wed Jan 04 12:41:44 2012 -0800"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Fri Feb 03 15:48:25 2012 -0800"
      },
      "message": "AudioTrack declare more methods const\n\nChange-Id: I4999e984460893961d0d8092cff17f3cf07d7214\n"
    },
    {
      "commit": "a0d68338a88c2ddb4502f95017b546d603ef1ec7",
      "tree": "f052125f4720fde00179d41d4ee04e02934052e4",
      "parents": [
        "87f155d6655b2d3b27e69281a29e85c6407e4d26"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Fri Jan 27 16:47:15 2012 -0800"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Fri Feb 03 07:57:01 2012 -0800"
      },
      "message": "Use NULL not 0 for raw pointers\n\nUse if (p !\u003d NULL) instead of if (ptr)\n\nChange-Id: Iaec3413a59ccbf233c98fcd918cc7d70ac5da9fa\n"
    },
    {
      "commit": "0e265cf36d201a7ccc0238b5c60b50f43d1dc450",
      "tree": "395a2c8f3685c1d1140f3ad12d2dff884d7db6ad",
      "parents": [
        "bf6954598193147cac980b8b5d7ad5215f1dec16"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Thu Feb 02 10:52:34 2012 -0800"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Thu Feb 02 11:23:30 2012 -0800"
      },
      "message": "Remove dead mutex in AudioTrack/AudioRecord thread\n\nThe client callback threads had mutexes called AudioTrackThread::mLock\nand ClientRecordThread::mLock.  These mutexes were only used by start()\nand stop(), and were unused by the thread itself.  But start() and\nstop() already have their own protection provided by AudioTrack::mLock\nand AudioRecord::mLock.  So the thread mutexes can be removed.\n\nChange-Id: I098406d381645d77fba06a15511e179a327848ef\n"
    },
    {
      "commit": "335787fe43596f38ea2fa50b24c54d0823a3fb1d",
      "tree": "8efb65da6c538ee243e4ac63541f8f815591aa7b",
      "parents": [
        "58f30210ea540b6ce5aa6a46330cd3499483cb97"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Fri Jan 20 17:00:00 2012 -0800"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Fri Jan 20 17:12:59 2012 -0800"
      },
      "message": "Remove AudioFlinger dependencies on client\n\nChange-Id: Ibb591e41a3ca5d7015e2b66b98b8fef5f415fb37\n"
    },
    {
      "commit": "c813985abd8ba61e999b3505f6a332574f87a1be",
      "tree": "d87c4ec564d4f3d19db4c1032661799cce847617",
      "parents": [
        "04eaf3e5f20d7faa956c0d672024ffb5117f4c26"
      ],
      "author": {
        "name": "Andreas Huber",
        "email": "andih@google.com",
        "time": "Wed Jan 18 10:51:55 2012 -0800"
      },
      "committer": {
        "name": "Andreas Huber",
        "email": "andih@google.com",
        "time": "Wed Jan 18 10:51:55 2012 -0800"
      },
      "message": "Temporarily restore AudioSystem/AudioTrack APIs with their former signatures\n\nuntil we get updated prebuilts from vendor.\n\nChange-Id: I8aae81d2513edca0ab268053a11c8c4206879e61\n"
    },
    {
      "commit": "613882293184e575a44bff681a3decaefe889e69",
      "tree": "052f84c8fe6e7b201f03117f617c3ad77b8b0dd0",
      "parents": [
        "0107954f72153db747a3727dc1157e9236dfed90",
        "b9980659501d0428d65d8292f3c32da69d37fbd2"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Tue Jan 17 07:27:46 2012 -0800"
      },
      "committer": {
        "name": "Android (Google) Code Review",
        "email": "android-gerrit@google.com",
        "time": "Tue Jan 17 07:27:46 2012 -0800"
      },
      "message": "Merge \"Use size_t for frame size\""
    },
    {
      "commit": "b9980659501d0428d65d8292f3c32da69d37fbd2",
      "tree": "8b703aca299f4f715a65d07cc37b753e9753a941",
      "parents": [
        "f4e76c538f81068bc2e125f3f9ef8b8dd40f2d2a"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Wed Jan 11 09:48:27 2012 -0800"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Fri Jan 13 14:58:44 2012 -0800"
      },
      "message": "Use size_t for frame size\n\nexcept in the control block, where we don\u0027t have room.\n\nIn AudioFlinger::ThreadBase::TrackBase::getBuffer,\nread the frame size from control block only once.\n\nChange-Id: Id6c4bccd4ed3e07d91df6bbea43bae45524f9f4e\n"
    },
    {
      "commit": "fff6d715a8db0daf08a50634f242c40268de3d49",
      "tree": "f5331971b746b2e0d878e1127003a0dab72e5fbb",
      "parents": [
        "09192653e836b21689f004bf8dee375356641181"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Thu Jan 12 16:38:12 2012 -0800"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Fri Jan 13 10:20:14 2012 -0800"
      },
      "message": "Use audio_stream_type_t consistently\n\nAt native level it was a mixture of audio_stream_type_t, int, uint32_t,\nand uint8_t.  Java is still int.  Also fixed a couple of hard-coded -1\ninstead of AUDIO_STREAM_DEFAULT, and in startToneCommand a hard-coded 0\ninstead of AUDIO_STREAM_VOICE_CALL.\n\nChange-Id: Ia33bfd70edca8c2daec9052984b369cd8eee2a83\n"
    },
    {
      "commit": "4cc55d53d542c4f4ed645738cebb65b9e7eb6c44",
      "tree": "417efab1bc349bfaa4bdf7fccdff0e53fb76e7c0",
      "parents": [
        "d5a8bf0bca459389cdb81fdcc30893516bdce951",
        "e1c3962e268ffc12bfd1bd9ea84da1f135f36960"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Thu Jan 12 10:30:28 2012 -0800"
      },
      "committer": {
        "name": "Android (Google) Code Review",
        "email": "android-gerrit@google.com",
        "time": "Thu Jan 12 10:30:28 2012 -0800"
      },
      "message": "Merge \"Use audio_format_t consistently\""
    },
    {
      "commit": "e1c3962e268ffc12bfd1bd9ea84da1f135f36960",
      "tree": "d24fac9434dbe5808a6c31a50414da7b5ce47c9a",
      "parents": [
        "a2a0a5d7d56baa831870f4bf2a0d942a477d92ef"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Wed Jan 04 09:36:37 2012 -0800"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Thu Jan 05 13:34:54 2012 -0800"
      },
      "message": "Use audio_format_t consistently\n\nWas int, uint32_t, uint16_t, and uint8_t with 2-bit bitfield.\nAlso replace 0 by AUDIO_FORMAT_DEFAULT and replace 1 by\nAUDIO_FORMAT_PCM_16_BIT.\n\nChange-Id: Ia8804f53f1725669e368857d5bb2044917e17975\n"
    },
    {
      "commit": "9a2aaf927e56a4b4acab23ef16b3f133a9f48a63",
      "tree": "c4bbdbe2a7e0292105736849e28faf12d298e47c",
      "parents": [
        "a2a0a5d7d56baa831870f4bf2a0d942a477d92ef"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Tue Jan 03 09:42:47 2012 -0800"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Thu Jan 05 11:55:24 2012 -0800"
      },
      "message": "Clean up AudioTrack::mActive and stopped()\n\nmActive is protected by mLock; volatile is meaningless on SMP.\n\nFixed a couple of places where mActive was accessed without a lock:\n - stopped()\n - processAudioBuffer()\n\nAdded stopped_l() for cases where we already hold the lock.\n\nMade mActive a bool not int.\n\nMoved down a lock in setPosition that was being acquired too early.\n\nChange-Id: I73ff368e991c0db9f9472df0b3f96fd33fcc7311\n"
    },
    {
      "commit": "362c4e697d8e9c034e964ac7b40227e054491547",
      "tree": "dde3ca3f11e00cc82f1b6a6bd7ebe7b1c24421c5",
      "parents": [
        "eebeceec684a36222b4559e3157b0db04c0a67ed"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Wed Dec 14 10:28:06 2011 -0800"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Wed Dec 14 14:38:26 2011 -0800"
      },
      "message": "Audio C++ comments\n\nChange-Id: I84906ebb9dfcfa5b96b287d18364b407f02a30c1\n"
    },
    {
      "commit": "c9260c15c1e7e2743039c964b8769379a3fff332",
      "tree": "cd829dd370a04344ec40f73c20dff6e8cc8b7793",
      "parents": [
        "926798f8c21ab002d9797ef8973852a2612c1f75",
        "b68a91a70bc8d0d18e7404e14443d4e4020b3635"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Mon Nov 28 14:54:28 2011 -0800"
      },
      "committer": {
        "name": "Android (Google) Code Review",
        "email": "android-gerrit@google.com",
        "time": "Mon Nov 28 14:54:28 2011 -0800"
      },
      "message": "Merge \"Make AudioTrack a RefBase so wrappers not needed\""
    },
    {
      "commit": "879135196fd1c97deefc538c888037c56c2879a7",
      "tree": "3ee39136a951bf1fbb25e0b484c3c1b6709c5868",
      "parents": [
        "5c052218128013d630b4d9a84393ebfbff11f3e8"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Wed Jun 22 16:15:25 2011 -0700"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Wed Nov 16 14:52:55 2011 -0800"
      },
      "message": "Bug 4903178 Restore priority and cgroup on stop\n\nOn AudioTrack and AudioRecord stop or failed start, restore the priority\nand cgroup of the caller to their previous values, rather than forcing\nto NORMAL.  Dependent on new thread APIs.\n\nAlso fixes bug where priority was set to AUDIO but cgroup not set.\n\nChange-Id: Ib83893918fb4fdf57c6b87884b51038997a631d8\n"
    },
    {
      "commit": "b68a91a70bc8d0d18e7404e14443d4e4020b3635",
      "tree": "e885c21714d54383f2fad17ac4d98cd6c822550e",
      "parents": [
        "5c052218128013d630b4d9a84393ebfbff11f3e8"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Tue Nov 15 13:55:13 2011 -0800"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Tue Nov 15 14:26:09 2011 -0800"
      },
      "message": "Make AudioTrack a RefBase so wrappers not needed\n\nChange-Id: I2305abe586ac41251af04a194bc818c110a1b293\n"
    },
    {
      "commit": "cfe2ba6b01a258e39f9c215ffc7b750e0b68f708",
      "tree": "c4a213cfc30955cc61a8c9673e032caff41a0947",
      "parents": [
        "98afd841e8f0803a00d1970e4a28ea8999e9c498"
      ],
      "author": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Tue Sep 13 15:04:17 2011 -0700"
      },
      "committer": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Tue Sep 13 17:33:29 2011 -0700"
      },
      "message": "Issue 5298399: Lost speech after a crash in gTalk.\n\nFixed problem in AudioTrack::restoreTrack_l() causing a permanent\nfailure if the IAudioTrack interface to AudioFlinger could not be\nrestored at the first attempt.\n\nChange-Id: I039d4fe2dca8d3baf71f1a6c51119f27a67b6611\n"
    },
    {
      "commit": "cd07594333cbe8b2c86c6609cce01a74d6cc33f8",
      "tree": "70b1b94cb811f564e8fe0c6ec348469adaae5f3c",
      "parents": [
        "539a735a82fefb1944aa6135df73dfe9f541566c"
      ],
      "author": {
        "name": "Jean-Michel Trivi",
        "email": "jmtrivi@google.com",
        "time": "Thu Aug 25 16:47:23 2011 -0700"
      },
      "committer": {
        "name": "Jean-Michel Trivi",
        "email": "jmtrivi@google.com",
        "time": "Thu Aug 25 17:33:49 2011 -0700"
      },
      "message": "Bug 4364249 Play position is 0 after flushing AudioTrack\n\nAudioTrack::stop() is not synchronous, so a stop() followed\n by flush(), which is synchronous, will not always report\n a playhead position of 0 after being called.\nThis CL adds a flag to mark a track as flushed, and report the\n correct playhead position in this state.\nBug 5217011 has been created to address the real issue in the\n future, where flush could be made synchronous, to properly\n address bug 4364249.\n\nChange-Id: Icf989d41a6bcd5985bb87764c287f3edb7e26d12\n"
    },
    {
      "commit": "d217a8c4632b3e3065f8c2a26b9ce4dc4c97171f",
      "tree": "36ea6300b9a3e5412d45ddaa5e5d6a38117c30c0",
      "parents": [
        "013b5a67d6f8e12a1b06e74863daec1bee4d99c1"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Wed Jun 01 15:20:35 2011 -0700"
      },
      "committer": {
        "name": "Android (Google) Code Review",
        "email": "android-gerrit@google.com",
        "time": "Wed Jun 01 15:20:35 2011 -0700"
      },
      "message": "Revert \"Use AudioTrack::event_type not int in callback\"\n\nThis reverts commit 2225e4b7049fa3fb9d39a068b8268b63c952d7c1"
    },
    {
      "commit": "013b5a67d6f8e12a1b06e74863daec1bee4d99c1",
      "tree": "cb10033d696d75aa6b609f39c19e8b5640f9c2fc",
      "parents": [
        "8a72c54e74eb365971288a81687e5c2aa47963e7"
      ],
      "author": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Wed May 04 17:18:16 2011 -0700"
      },
      "committer": {
        "name": "Glenn Kasten",
        "email": "gkasten@google.com",
        "time": "Wed Jun 01 13:50:23 2011 -0700"
      },
      "message": "Use AudioTrack::event_type not int in callback\n\nChange-Id: Ifefc708d46874e04fd0d01cb6e2d43b987ee796c\n"
    },
    {
      "commit": "0d255b2d9061ba31f13ada3fc0f7e51916407176",
      "tree": "4df26e48710542386bddab7a0d1cbfa5d6835ac0",
      "parents": [
        "65580f9adf6c4d98449ad0716488f9fe3869aa5a"
      ],
      "author": {
        "name": "Jean-Michel Trivi",
        "email": "jmtrivi@google.com",
        "time": "Tue May 24 15:53:33 2011 -0700"
      },
      "committer": {
        "name": "Jean-Michel Trivi",
        "email": "jmtrivi@google.com",
        "time": "Wed Jun 01 10:55:29 2011 -0700"
      },
      "message": "Use channel mask instead of channel count for track creation\n\nRecord and playback objects (resp AudioRecord and AudioTrack)\nare created using a channel mask, but this information is lost\nin the mixer because only the channel count is known to\nAudioFlinger. A channel count can always be derived from a\nchannel mask.\n\nThe change consists in:\n- disambiguiting variable names for channel masks and counts\n- passing the mask information from the client to AudioFlinger\n and the mixer.\n- when using the DIRECT ouput, only verifying the format of\n the track is compatible with the output\u0027s for PCM.\n\nChange-Id: I50d87bfb7d7afcabdf5f12d4ab75ef3a54132c0e\n"
    },
    {
      "commit": "fce7a473248381cc83a01855f92581077d3c9ee2",
      "tree": "e002e1edd8a11f2be56ce9538ab1b13aa289bc9e",
      "parents": [
        "db5cb14318bb24cd6ea14ff7ceea0d5e1f83d903"
      ],
      "author": {
        "name": "Dima Zavin",
        "email": "dima@android.com",
        "time": "Tue Apr 19 22:30:36 2011 -0700"
      },
      "committer": {
        "name": "Dima Zavin",
        "email": "dima@android.com",
        "time": "Wed Apr 27 13:10:10 2011 -0700"
      },
      "message": "audio/media: convert to using the audio HAL and new audio defs\n\nChange-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5\nSigned-off-by: Dima Zavin \u003cdima@android.com\u003e\n"
    },
    {
      "commit": "1703cdfee717b1b312bf8979816a9e2f16a82e5d",
      "tree": "eb9d3cfe8ffebe45679de91af6f9ffd676ca5067",
      "parents": [
        "fbb1909036ba7de3d9fb8738daba60b357881153"
      ],
      "author": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Mon Mar 07 14:52:59 2011 -0800"
      },
      "committer": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Tue Mar 08 16:33:15 2011 -0800"
      },
      "message": "Fix issue 3439872: video chat and bluetooth SCO\n\nThis change fixes the stability problems experienced when using\na bluetooth headset supporting both A2DP and SCO. Problems occur\nwhen starting the video chat at which time the A2DP output is being\nstopped to start SCO. At that time, active AudioTracks are invalidated\nby AudioFlinger so that a new AudioTrack binder interface can be\nrecreated by the client process on the new mixer thread with correct parameters.\nThe problem was that the process to restore the binder interface was not\nprotected against concurrent requests which caused 2 binder interfaces\nto be created sometimes. This could lead to permanent client deadlock\nif one of the client threads was waiting for a condition of the first\ncreated binder interface while the second one was created (as the AudioFlinger\nwould only signal conditions on the last one created).\nThis concurrent request situation is more likely to happen when a client\nuses the JAVA AudioTrack as the JNI implementation uses simultaneously the\nnative AudioTrack callback and write push mechanisms. By doing so, the code\nthat checks if the binder interface should be restored (in obtainBuffer()) is\nmuch more likely to be called concurrently from two different threads.\n\nThe fix consists in protecting the critical binder interface restore phase\nwith a flag in the AudioTrack control block. The first thread acting upon the binder\ninterface restore request will raise the flag and the second thread will just wait for\na condition to be signaled when the restore process is complete.\n\nAlso protected all accesses to the AudioTrack control block by a mutex to prevent\naccess while the track is being destroyed and restored. If a mutex cannot be held\n(e.g because we call a callback function), acquire a strong reference on the IAudioTrack\nto prevent its destruction while the cblk is being accessed.\n\nModified AudioTrack JNI to use GetByteArrayElements() instead of\nGetPrimitiveArrayCritical() when writing audio buffers. Entering a critical section would\ncause the JNI to abort if a mediaserver crash occurs during a write due to the AudioSystem\ncallback being called during the critical section when media server process restarts.\nAnyway with current JNI implementation, either versions do not copy data most of the times\nand the criticial version does not guaranty no data copy.\n\nThe same modifications have been made to AudioRecord.\n\nChange-Id: Idc5aa711a04c3eee180cdd03f44fe17f3c4dcb52\n"
    },
    {
      "commit": "f5aafb209d01ba2ab6cb55d1a12cfc653e2b4be0",
      "tree": "75b6429a6bb8b78943b770cad6924a7aaa94dc32",
      "parents": [
        "b9ff444a7eaf7ffd43970c0477110c6808bd4a7c"
      ],
      "author": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Thu Nov 18 08:40:16 2010 -0800"
      },
      "committer": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Fri Nov 19 15:49:42 2010 -0800"
      },
      "message": "Fix issue 3157123.\n\nUse a Mutex wherever atomic operations were used in AudioTrack,\nAudioRecord, AudioFlinger and AudioEffect classes.\n\nChange-Id: I6f55b2cabdcd93d64ef19446735b8f33720f8dbc\n"
    },
    {
      "commit": "2beeb50b1bba9e92f6cacfeca37fe9fa9d36ead1",
      "tree": "ce966d62f6fbb94178763ec53760f6961ffffd58",
      "parents": [
        "3108231d90e8aa324923fd8864ca2477948c5d25"
      ],
      "author": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Fri Jul 16 07:43:46 2010 -0700"
      },
      "committer": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Wed Jul 21 06:28:01 2010 -0700"
      },
      "message": "Added support for auxiliary audio effects to AudioTrack and MediaPlayer.\n\nAdded methods to AudioTrack and MediaPlayer java classes to enable use of\nauxiliary audio effects. The effect can be attached and detached by specifying its\nID and the send level controlled.\n\nChange-Id: Ie74ff54a453096a742688476f612ce355543b6f3\n"
    },
    {
      "commit": "33005a932c60a0780fe9b7307d5988df3d9f6c26",
      "tree": "2519156995c4fabaae1dfdaa091e70b973a41547",
      "parents": [
        "1616fa4a567bae928dfd28ebf1ec45cd701f27f1"
      ],
      "author": {
        "name": "Chia-chi Yeh",
        "email": "chiachi@android.com",
        "time": "Wed Jun 16 06:33:13 2010 +0800"
      },
      "committer": {
        "name": "Chia-chi Yeh",
        "email": "chiachi@android.com",
        "time": "Mon Jun 21 09:51:18 2010 +0800"
      },
      "message": "media: add AudioTrack::getMinFrameCount().\n\nChange-Id: If15162583d1d16d89f59be0793106afe77417f35\n"
    },
    {
      "commit": "be916aa1267e2e6b1c148f51d11bcbbc79cb864c",
      "tree": "7e2be6182cde7d023ae9afec28463d69d231d7bc",
      "parents": [
        "c282e3eee921453fc9188705b4879d6289b71f9c"
      ],
      "author": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Tue Jun 01 23:49:17 2010 -0700"
      },
      "committer": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Thu Jun 03 03:21:53 2010 -0700"
      },
      "message": "Issue 2667801: [Audio Effect Framework] AudioFlinger, AudioMixer AudioTrack modifications.\n\nFirst drop of audio framework modifications for audio effects support.\n\n- AudioTrack/AudioRecord:\nAdded support for auxiliary effects in AudioTrack\nAdded support for audio sessions\nFixed left right channel inversion in setVolume()\n\n- IAudioFlinger:\nAdded interface methods for effect enumeraiton and instantiation\nAdded support for audio sessions.\n\n- IAudioTrack:\nAdded method to attach auxiliary effect.\n\n- AudioFlinger\nCreated new classes to control effect engines in effect library and manage effect connections to tracks or\noutput mix:\n  EffectModule: wrapper object controlling the effect engine implementation in the effect library. There\n\tis one EffectModule per instance of an effect in a given audio session\n  EffectChain: group of effects associated to one audio session. There is one EffectChain per audio session.\n\tEffectChain for session 0 is for output mix effects, other chains are attached to audio tracks\n\twith same session ID. Each chain contains a variable number of EffectModules\n  EffectHandle: implements the IEffect interface. There is one EffectHandle object for each application\n\tcontrolling (or using) an effect module. THe EffectModule maintians a list of EffectHandles.\n\nAdded support for effect modules and effect chains creation in PlaybackThread.\nmodified mixer thread loop to allow track volume control by effect modules and call effect processing.\n\n-AudioMixer\nEach track now specifies its output buffer used by mixer for accumulation\nModified mixer process functions to process tracks by groups of tracks with same buffer\nModified track process functions to support accumulation to auxiliary channel\n\nChange-Id: I26d5f7c9e070a89bdd383e1a659f8b7ca150379c\n"
    },
    {
      "commit": "d1b449aad6c087a69f5ec66b7facb2845b73f1cb",
      "tree": "b16f7d28fdac76c90d66be56161116fe09822992",
      "parents": [
        "ae77ffa16bda593fb3751e41d45327d867f3c8e1"
      ],
      "author": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Fri May 14 03:26:45 2010 -0700"
      },
      "committer": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Mon May 17 02:23:47 2010 -0700"
      },
      "message": "Fix issue 2553359: Pandora does not work well with Passion deskdock / Cardock.\n\nThe problem is due to a too big difference between the buffer size used at the hardware interface and at the A2DP interface.\nWhen no resampling occurs we don\u0027t notice problems but the timing is very tight. As soon as resampling is activated, the AudioTrack underruns.\nThis is because the AudioTrack buffers are not resized when moving the AudioTrack from hardware to A2DP output.\nThe AudioTrack buffers are calculated based on a hardware output buffer size of 3072 bytes. Which is much less than the A2DP output buffer size (10240).\n\nThe solution consists in creating new tracks with new buffers in AudioFlinger when the A2DP output is opened\ninstead of just transfering active tracks from hardware output mixer thread to the new A2DP output mixer thread.\nTo avoid synchronization issues between mixer threads and client processes, this is done by invalidating tracks\nby setting a flag in their control block and having AudioTrack release the handle on this track (IAudioTrack)\nand create a new IAudioTrack when this flag is detected next time obtainBuffer() or start() is executed.\n\nAudioFlinger modifications:\n- invalidate the tracks when setStreamOutput() is called\n- make sure that notifications of output opening/closing and change of stream type to output mapping are sent synchronously to client process.\nThis is necessary so that AudioSystem has the new stream to output mapping when the AudioTrack detects the invalidate flag in the client process.\nPreviously their were sent when the corresponding thread loop was executed.\n\nAudioTrack modifications:\n- move frame count calculation and verification from set() to createTrack() so that is is updated every time a new IAudioTrack is created.\n- detect track invalidate flag in obtainBuffer() and start() and create a new IAudioTrack.\n\nAudioTrackShared modifications\n- group all flags (out, flowControlFlag, forceReady...) into a single bit filed to save space.\n\nChange-Id: I9ac26b6192230627d35084e1449640caaf7d56ee\n"
    },
    {
      "commit": "34f1d8ecd23169a5f299937e3aaf1bd7937578a0",
      "tree": "04f5504f74df6dde00b495a95bf30766be12821a",
      "parents": [
        "c6c6237a67661c7b2d8822b7360cdf01e64d2c7c"
      ],
      "author": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Wed Nov 04 08:27:26 2009 -0800"
      },
      "committer": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Wed Nov 04 23:47:21 2009 -0800"
      },
      "message": "Fix issue 2203561: Sholes: audio playing out of earpiece.\n\nCreate a new IAudioTrack interface to AudioFlinger when start() fails due to a broken pipe error.\nDo the same if start fails due to the same error after time out in obtainBuffer().\nDo not indicate that the AudioTrack is started to AudioPolicyManager if IAudioTrack start fails.\nThis avoids that an AudioTrack keeps a dead IAudioTrack after a media server crash.\n\nSame modifications for AudioRecord.\n\nAdd a flag to ToneGenerator indicating that the callback thread can call Java. Without it, when the media server crashes and restarts, the AudioSystem error callback will crash in JNI if the IAudiotrack is created from AudioTrack callback thread.\n"
    },
    {
      "commit": "c2f1f07084818942352c6bbfb36af9b6b330eb4e",
      "tree": "88ac93be41edadd8cbfe6448e1421d5165883f59",
      "parents": [
        "a64c8c79af1a15911c55306d83a797fa50969f77"
      ],
      "author": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Fri Jul 17 12:17:14 2009 -0700"
      },
      "committer": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Thu Jul 23 06:03:39 2009 -0700"
      },
      "message": "Fix issue 1795088       Improve audio routing code\n\nInitial commit for review.\nIntegrated comments after patch set 1 review.\nFixed lockup in AudioFlinger::ThreadBase::exit()\nFixed lockup when playing tone with AudioPlocyService startTone()\n"
    },
    {
      "commit": "349dba337e07e129f6ba49a132999f0b73fedbe3",
      "tree": "8bd3262593f56c5fce1d6bced18f9f26a8220934",
      "parents": [
        "82827d70378961f30077d3a35a06aab58c368d52",
        "573266210fb2b2e7d86fbd46d0dfe16763611d91"
      ],
      "author": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Tue Jul 07 09:36:14 2009 -0700"
      },
      "committer": {
        "name": "The Android Open Source Project",
        "email": "initial-contribution@android.com",
        "time": "Tue Jul 07 09:36:14 2009 -0700"
      },
      "message": "am 88e209dc: Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR\n\nMerge commit \u002788e209dcf8c2ebddda5c272f46d1bd5478bc639c\u0027\n\n* commit \u002788e209dcf8c2ebddda5c272f46d1bd5478bc639c\u0027:\n  Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR\n"
    },
    {
      "commit": "573266210fb2b2e7d86fbd46d0dfe16763611d91",
      "tree": "f7565b32ee2d59d5098dd165658c0aefa65f1fc4",
      "parents": [
        "76aa28371e93829927c66f1a48e39e18e01d704d"
      ],
      "author": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Tue Jul 07 07:10:45 2009 -0700"
      },
      "committer": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Tue Jul 07 07:10:45 2009 -0700"
      },
      "message": "Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR\n\nStore sample rate on 32 bits instead of 16 bits in audio_track_cblk_t.\nRemoved sampleRate() methods from AudioTrack and AudioRecord: replaced by getSampleRate().\nAudioTrack::setSampleRate() no returns a status.\n"
    },
    {
      "commit": "7562408b2261d38415453378b6188f74fda99d88",
      "tree": "6b2ac284c1b721385d695045357d9e1d60ece03a",
      "parents": [
        "615a9bfbd98b90e09d868c128db84836feb658e4"
      ],
      "author": {
        "name": "Mathias Agopian",
        "email": "mathias@google.com",
        "time": "Tue May 19 19:08:10 2009 -0700"
      },
      "committer": {
        "name": "Mathias Agopian",
        "email": "mathias@google.com",
        "time": "Wed May 20 12:55:03 2009 -0700"
      },
      "message": "move libbinder\u0027s header files under includes/binder\n"
    },
    {
      "commit": "1dd70b9f04961a06fcb73a97fca10a53b3245d3c",
      "tree": "a2704fd75e6de04107a7d40bd3e56f6f7acfa51b",
      "parents": [
        "ce80c82f58378da72d8a36f520c781cc5cac18a1"
      ],
      "author": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Tue Apr 21 07:56:33 2009 -0700"
      },
      "committer": {
        "name": "Eric Laurent",
        "email": "elaurent@google.com",
        "time": "Tue Apr 21 07:56:33 2009 -0700"
      },
      "message": "Fix issue 1745312: Various cleanups in media framework\n\nAudioTrack, AudioRecord:\n  - remove useless mAudioFlinger member of AudioTrack and AudioRecord.\n  - signal cblk.cv condition in stop() method to speed up stop completion.\n  - extend wait condition timeout in obtainBuffer() when waitCount is -1 to avoid waking up callback thread unnecessarily\n\nAudioFlinger:\n  - remove some warnings in AudioFlinger.cpp.\n  - remove function AudioFlinger::MixerThread::removetrack_l()  as its content is never executed.\n  - remove useless call to setMasterVolume in AudioFlinger::handleForcedSpeakerRoute().\n  - Offset VOICE_CALL stream volume to reflect actual volume that is never 0 in hardware (this fix has been made in the open source): 0.01 + v * 0.99.\n\nAudioSystem.java:\n  - correct typo in comment\n\nIAudioflinger, IAudioFlingerClient:\n  - make AudioFlinger binder interfaces used for callbacks ONEWAY.\n\nAudioHardwareInterface:\n  - correct routeStrings[] table in AudioHardwareInteface.cpp\n"
    },
    {
      "commit": "2c22aeb65e801f663a754d043062f85e49f77739",
      "tree": "56016c49e2ce13a911a21b25cac543695e45aadc",
      "parents": [
        "1179bc9b0e3d17c984e8f4ad38561c049dd102fa"
      ],
      "author": {
        "name": "Jean-Michel Trivi",
        "email": "",
        "time": "Tue Mar 24 18:11:07 2009 -0700"
      },
      "committer": {
        "name": "The Android Open Source Project",
        "email": "initial-contribution@android.com",
        "time": "Tue Mar 24 18:11:07 2009 -0700"
      },
      "message": "Automated import from //branches/donutburger/...@140663,140663\n"
    },
    {
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