)]}'
{
  "log": [
    {
      "commit": "55773033ed1f23c4e8f3d2704002fbd0cedbf7a5",
      "tree": "54bc894243fa3c52e611c85aed75ee6677056c13",
      "parents": [
        "7aed66e788c8955ddd22a1235856bb9dac34d82a"
      ],
      "author": {
        "name": "Mikhail Naganov",
        "email": "mnaganov@google.com",
        "time": "Thu Oct 01 15:08:13 2020 -0700"
      },
      "committer": {
        "name": "Mikhail Naganov",
        "email": "mnaganov@google.com",
        "time": "Thu Oct 08 18:12:55 2020 -0700"
      },
      "message": "Convert mask types from uint32_t to enum type\n\nThis applies to the following types:\n\n- audio_gain_mode_t;\n- audio_flags_mask_t;\n- audio_channel_representation_t;\n- audio_channel_mask_t;\n- audio_devices_t.\n\nEnum types are distinct thus proper overloading on the type\nis possible in C++. Also, assignments to enum types are\nless prone to errors.\n\nBug: 169889714\nTest: basic audio functionality\n      m stagefright\n      atest audiopolicy_tests\nChange-Id: I42506484b1d0b0482618b7314fad4c8012a06fb4\n"
    },
    {
      "commit": "f14e77fef40420ac18d066f0de0486803b941742",
      "tree": "1d2605f264d85476362829da612769e591ce2c8f",
      "parents": [
        "87025d5658d1a3d78c1a634035f7126f0ba8a4f1"
      ],
      "author": {
        "name": "Dean Wheatley",
        "email": "dwhea@dolby.com",
        "time": "Thu Mar 23 15:52:01 2017 +1100"
      },
      "committer": {
        "name": "Dean Wheatley",
        "email": "dwhea@dolby.com",
        "time": "Tue Oct 01 10:57:17 2019 +1000"
      },
      "message": "SpdifStreamOut: Support AUDIO_FORMAT_E_AC3_JOC\n\nTest: build\nChange-Id: Iac3dc90e136d625ecbd95eea5c2eea13fc938657\n"
    },
    {
      "commit": "cbc8f617c1aebef5d041fa40dcd38a5466690b99",
      "tree": "525470607c3e6436c5a73caac90830104fcde9ac",
      "parents": [
        "0c739c59dd1a3d52beb82a353c13b6c853d93f5f"
      ],
      "author": {
        "name": "Mikhail Naganov",
        "email": "mnaganov@google.com",
        "time": "Tue Oct 11 18:05:13 2016 -0700"
      },
      "committer": {
        "name": "Mikhail Naganov",
        "email": "mnaganov@google.com",
        "time": "Tue Oct 11 18:05:13 2016 -0700"
      },
      "message": "Eliminate dependencies on hardware/audio.h (trivial cases)\n\nframeworks/av must not depend on hardware/ (except for the code\nfrom libaudiohal that actually calls into HAL).\n\nThis CL deals with simple cases where depending on system/audio.h\nis enough.\n\nChange-Id: Ia2cb66cc8c92316ce5ab884a008d5e531263c2e4\nTest: make\n"
    },
    {
      "commit": "1dc98674f701dada94143b4d31b7221c58346c6c",
      "tree": "44f80a66fdff774d2229ca3aaabdd614ed1f8013",
      "parents": [
        "9debf8f6d26f7aa12c68ddfb6cdb1323d4b5adbf"
      ],
      "author": {
        "name": "Mikhail Naganov",
        "email": "mnaganov@google.com",
        "time": "Thu Aug 18 17:50:29 2016 -0700"
      },
      "committer": {
        "name": "Mikhail Naganov",
        "email": "mnaganov@google.com",
        "time": "Wed Sep 21 15:58:24 2016 -0700"
      },
      "message": "Abstract away access to audio streams HAL in AudioFlinger\n\nIn this CL all direct access to audio_stream_t, audio_stream_out_t, and\naudio_stream_in_t their functions is encapsulated within the new\nhierarchy of Stream[In|Out]HalLocal classes.  AudioFlinger uses\ninterface classes Stream[In|Out]HalInterface to access these functions.\n\nNote that NBAIO still receives raw HAL stream handles and needs to be\nconverted separately.\n\nBug: 30222631\nTest: manual with Loopback app\n\nChange-Id: I6388cfa2006791c9c0aa7bb186719209726a2d48\n"
    },
    {
      "commit": "41ae1d663d896a9ec22299d95c01940313a9b276",
      "tree": "0d6246404b453fd7ebb7d134d3b02ce6e01c5869",
      "parents": [
        "6134ad6fdeab91e54a1abc8f00eafc956e42fb3d"
      ],
      "author": {
        "name": "Phil Burk",
        "email": "philburk@google.com",
        "time": "Tue Aug 04 11:11:28 2015 -0700"
      },
      "committer": {
        "name": "Phil Burk",
        "email": "philburk@google.com",
        "time": "Tue Aug 04 11:11:28 2015 -0700"
      },
      "message": "AudioFlinger: tell HAL the PCM is non-audio\n\nAdd a bit to tell the HAL that the PCM data is really encoded audio\nwrapped in a data burst.\nOtherwise the device may try to play the encoded data directly\nwhich will sound like modulated white noise.\n\nBug: 22576112\nChange-Id: Ib140da96876e849023858fe2510612310501d1ee\nSigned-off-by: Phil Burk \u003cphilburk@google.com\u003e\n"
    },
    {
      "commit": "ca5e6143740299c877d69e97f7968cd04476d32c",
      "tree": "50101b57406c5abce066ec90120d91d3dfd0152b",
      "parents": [
        "bc24bb8a552097e7975d0c16fad80158b542ba62"
      ],
      "author": {
        "name": "Phil Burk",
        "email": "philburk@google.com",
        "time": "Tue Jul 14 09:42:29 2015 -0700"
      },
      "committer": {
        "name": "Lajos Molnar",
        "email": "lajos@google.com",
        "time": "Fri Jul 17 20:56:04 2015 +0000"
      },
      "message": "AudioFlinger: fix repeated underruns for compressed audio\n\nThe AudioFlinger kept pausing the audio when playing compressed AC3 or DTS.\nThis caused pause/resume loops that were hard to break out of.\n\nThe AudioFlinger was thinking that the compressed audio was PCM\nbecause the HAL was in PCM mode playing SPDIF data bursts.\nIt also thought that EAC3 was at 192000 Hz instead of 48000\nHz because the data bursts are played at a higher rate.\n\nThis CL adds more calls to the shim that separates the AudioFlinger.\nNow the AudioFlinger gets information about the HAL sample rate,\nchannel masks and format from the shim instead of calling the HAL directly.\n\nThe AudioFlinger now uses a different threshold for detecting\nunderruns when the audio is compressed.\n\nBug: 19938315\nBug: 20891646\nChange-Id: Ib16f539346d1c7a273ea4feb3d3afcc3dc60237d\nSigned-off-by: Phil Burk \u003cphilburk@google.com\u003e\n"
    },
    {
      "commit": "90eea7631b07117e46ae8b84889a2baa3eee7aea",
      "tree": "e31c07b33a68919b011a031f08eb785bba0c066d",
      "parents": [
        "29b7cec9a248828d2371bc8758d819308d0e84a4"
      ],
      "author": {
        "name": "Phil Burk",
        "email": "philburk@google.com",
        "time": "Mon Jul 06 16:24:14 2015 -0700"
      },
      "committer": {
        "name": "Phil Burk",
        "email": "philburk@google.com",
        "time": "Tue Jul 07 16:40:00 2015 -0700"
      },
      "message": "AudioFlinger: reset frame position on standby\n\nThe HAL does not reset the frame position on standby().\nBut applications expect the frame position to be reset.\nSo we subtract the position at standby from the current position.\n\nBug: 21724210\nBug: 21930805\nChange-Id: I0c4520ba1c6c06a580f45f6bafc8cf1d56969f07\nSigned-off-by: Phil Burk \u003cphilburk@google.com\u003e\n"
    },
    {
      "commit": "48e6ea95dc23e1c84911f1f907403ac1729c6b78",
      "tree": "d02ec907f02e33d5cd6d9d870898d461557e3305",
      "parents": [
        "91b0018d658a509aacceddf9cde84afca591d014"
      ],
      "author": {
        "name": "Phil Burk",
        "email": "philburk@google.com",
        "time": "Thu Jun 18 15:37:08 2015 -0700"
      },
      "committer": {
        "name": "Phil Burk",
        "email": "philburk@google.com",
        "time": "Fri Jun 19 17:11:30 2015 -0700"
      },
      "message": "AudioFlinger: reset SPDIF encoder on flush\n\nThis may prevent corrupt data from reaching the AVR.\n\nBug: 19604395\nChange-Id: I3dbdd002326917b9f14dda46a3e5170e1ec799bb\nSigned-off-by: Phil Burk \u003cphilburk@google.com\u003e\n"
    },
    {
      "commit": "23d8997f58bb9c59fa3a1b9a6b2edbf1b2b0f4c6",
      "tree": "eed231c9bd5dc2523b7a0cf7c8b142810079dabc",
      "parents": [
        "31ffe45a22b72ee5d25bb4932d0525e90dbc19fe"
      ],
      "author": {
        "name": "Phil Burk",
        "email": "philburk@google.com",
        "time": "Mon Apr 06 16:22:23 2015 -0700"
      },
      "committer": {
        "name": "Phil Burk",
        "email": "philburk@google.com",
        "time": "Thu Apr 09 16:18:34 2015 -0700"
      },
      "message": "AudioFlinger: more DTS passthrough support\n\nPass format for AC3 or DTS to SPDIF encoder.\n\nBug: 18292317\nChange-Id: I6d80eb548cb431bac3390ebb3235146171344291\nSigned-off-by: Phil Burk \u003cphilburk@google.com\u003e\n"
    },
    {
      "commit": "062e67a26e0553dd142be622821f493df541f0c6",
      "tree": "125d28264adfc5b7bd993bb343569eea63bfb95d",
      "parents": [
        "21b51b61ee52e6aa74d98b138d3dd4f0e17b1441"
      ],
      "author": {
        "name": "Phil Burk",
        "email": "philburk@google.com",
        "time": "Wed Feb 11 13:40:50 2015 -0800"
      },
      "committer": {
        "name": "Phil Burk",
        "email": "philburk@google.com",
        "time": "Tue Mar 24 13:24:18 2015 -0700"
      },
      "message": "AudioFlinger: call SPDIF wrapper from AudioFlinger\n\nCreate an interface layer between the AudioFlinger and the HAL\nthat manages the wrapping and format conversion.\n\nRemoved unnecessary includes.\nHandle rate conversion in getRenderPosition().\nTry to open HAL with encoded format before wrapping with SPDIF.\n\nBug: 17566660\nChange-Id: I00ad888ca15ff0f85b85efb8167c7f5ea761a244\nSigned-off-by: Phil Burk \u003cphilburk@google.com\u003e\n"
    }
  ]
}