Commit Graph

  • 73acc15c69 Revert 8538 "Reland "Fix CVO in androidvideocapturer.""" aluebs@webrtc.org 2015-02-27 23:27:26 +00:00
  • 3a93e33c56 Reland "Fix CVO in androidvideocapturer."" This reverts commit b8bcf8cbbf84971e2ae26d91659afdc58617b054. after I fixed a rebase mistake. The fix is the delta between patchset 1 and 2. perkj@webrtc.org 2015-02-27 20:18:38 +00:00
  • b8bcf8cbbf Revert "Fix CVO in androidvideocapturer." perkj@webrtc.org 2015-02-27 19:47:41 +00:00
  • 02ed57bf9d Fix CVO in androidvideocapturer. perkj@webrtc.org 2015-02-27 19:10:05 +00:00
  • 41d8fda12d VideoCapturerAndroid allocates direct buffers so that the frame buffers can be used in C++ without a copy. However byte[] array = ByteBuffer.array() seems to point to the beginning of the underlaying buffer and that is what the camera fills. But it turns out that ByteBuffer.arrayOffset() returns an offset and it seems like the pointer returned by jni->GetDirectBufferAddress(j_frame). This cl reverts back to pass the byte[] to c++ and use jni->GetByteArrayElements to get the address of the buffer. perkj@webrtc.org 2015-02-27 18:50:53 +00:00
  • 07dcf60ee0 Revert 8532 "Ensure only temporary IPv6 address is selected as t..." aluebs@webrtc.org 2015-02-27 18:42:22 +00:00
  • 21ad37528e Ensure we set the right attrib for correct shader guoweis@webrtc.org 2015-02-27 18:11:52 +00:00
  • 385a7ceb1f Ensure only temporary IPv6 address is selected as the best IP. guoweis@webrtc.org 2015-02-27 18:10:08 +00:00
  • fbef5c6293 Remove lock in ViEFrameProviderBase::IsFrameCallbackRegistered. tommi@webrtc.org 2015-02-27 15:42:37 +00:00
  • 7400e0b876 Revert "I420VideoFrame: Remove functions set_width, set_height, and ResetSize" magjed@webrtc.org 2015-02-27 15:18:26 +00:00
  • 4b3618c7f3 Remove TraceImpl logging thread. pbos@webrtc.org 2015-02-27 15:05:03 +00:00
  • 6c2e506cf4 Workaround Mac align bug for observer_ and crit_. pbos@webrtc.org 2015-02-27 14:28:15 +00:00
  • 3985f0151a ProcessThread improvements. tommi@webrtc.org 2015-02-27 13:36:34 +00:00
  • f296859c83 PeerConnectionClient.createPeerConnectionClient was calling new PeerConnectionParameters and PeerConnectionClient.createPeerConnectionFactory, .createPeerConnection with invalid arguments. hbos@webrtc.org 2015-02-27 12:42:48 +00:00
  • c68e0c9dfe Fix cpplint warning in the previous cl to peerconnection client example. braveyao@webrtc.org 2015-02-27 09:51:25 +00:00
  • abbdd520b0 AudioEncoder: documentation fix jmarusic@webrtc.org 2015-02-27 09:20:01 +00:00
  • ea89495786 Remove {Is,Set}BlackOutput from VideoAdapter. pbos@webrtc.org 2015-02-27 08:56:14 +00:00
  • 3aca0b0b31 Add 48kHz support to Beamformer aluebs@webrtc.org 2015-02-26 21:52:20 +00:00
  • 9650ab4d59 Fix case sensitivity of AppRTCDemo include dirs tkchin@webrtc.org 2015-02-26 20:58:08 +00:00
  • 2a72c6506a Keep feedback params in SetDefaultEncoderConfig. pbos@webrtc.org 2015-02-26 16:01:24 +00:00
  • b1f0de30be AudioEncoder: change Encode and EncodeInternal return type to void jmarusic@webrtc.org 2015-02-26 15:38:10 +00:00
  • 00b8f6b364 Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away kwiberg@webrtc.org 2015-02-26 14:34:55 +00:00
  • ac2d27d9ae Fix style violations in common_types.h and config.h kwiberg@webrtc.org 2015-02-26 13:59:22 +00:00
  • 891d48393e Wire up target_media_bitrate in VideoSendStream. pbos@webrtc.org 2015-02-26 13:15:22 +00:00
  • 9dd0ebc379 Remove the default RTP module. mflodman@webrtc.org 2015-02-26 12:57:47 +00:00
  • 3e6e271ec3 Implement CpuOveruseMetrics as callbacks. pbos@webrtc.org 2015-02-26 12:19:31 +00:00
  • 60f295fcb1 Remove lsan suppression.txt kjellander@webrtc.org 2015-02-26 11:13:38 +00:00
  • e723728992 Add p2p.gyp to rtc_base presubmit check exclusion. kjellander@webrtc.org 2015-02-26 11:12:17 +00:00
  • 9a4410e993 Implement adaptation stats in WebRtcVideoEngine2. pbos@webrtc.org 2015-02-26 10:03:39 +00:00
  • 38d9cc51d5 Add back return statement after FATAL() henrik.lundin@webrtc.org 2015-02-26 09:42:56 +00:00
  • b5e60b6ca7 Remove non necessary check from WebSocket send function. glaznev@webrtc.org 2015-02-25 19:18:29 +00:00
  • f09e7b8a4f WebRtcVideoFrame: DCHECK exclusive ownership for non-const pixel access magjed@webrtc.org 2015-02-25 14:49:48 +00:00
  • 6c66163567 Fix TestScaler PSNR tests magjed@webrtc.org 2015-02-25 14:10:22 +00:00
  • 96abda0316 Removing FEC functionality from the default RTP module. mflodman@webrtc.org 2015-02-25 13:50:10 +00:00
  • 9b969e167d AudioEncoderCopyRed: CHECK that encode call doesn't fail jmarusic@webrtc.org 2015-02-25 11:53:13 +00:00
  • 749c60217d Moved gypi to avoid presubmit warning about '..' when touching the files. andresp@webrtc.org 2015-02-25 11:50:17 +00:00
  • 5c928ebd1d Let first packet through to avoid getting key frame requests (and no nacks) for EndToEndTest.ReceivedFecPacketsNotNacked. asapersson@webrtc.org 2015-02-25 11:47:11 +00:00
  • 09c77b95bb Add decoder-timing stats to VideoReceiveStream. pbos@webrtc.org 2015-02-25 10:42:16 +00:00
  • c5558b7021 Remove AudioCodingModule's dependency on the Module interface henrik.lundin@webrtc.org 2015-02-25 10:37:20 +00:00
  • af82f75690 Let Add10MsData method do the encoding work as well henrik.lundin@webrtc.org 2015-02-25 10:33:10 +00:00
  • 4aef5fef18 Add thread checks to the CaptureManager. hbos@webrtc.org 2015-02-25 10:09:05 +00:00
  • 8d350d4bc4 Add new AcmGenericCodecTest and verify output from Encode function henrik.lundin@webrtc.org 2015-02-25 10:06:06 +00:00
  • 1eda4e3db6 Reland r8476 "Set decoder output frequency in AudioDecoder::Decode call" henrik.lundin@webrtc.org 2015-02-25 10:02:29 +00:00
  • 1e64263b90 Thread-safe ChannelManager.GetSupportedFormats, used by VideoSource hbos@webrtc.org 2015-02-25 09:49:41 +00:00
  • 0a3ff7976b New AudioTrack implementation now works on pre-Lollipop devices. henrika@webrtc.org 2015-02-25 09:27:51 +00:00
  • 112f127170 Refactor how VideoCapturerAndroid delivers frames and is stopped. With this cl, video buffers are now allocated using direct buffers. These buffers are guaranteed to live as long as the capturer is running. We can now post frames in c++ from the Java thread to the c++ worker thread and let c++ post the buffers back when it has finished processing them. perkj@webrtc.org 2015-02-25 09:20:07 +00:00
  • d4dfba8ea1 iSAC Decode: Prevent Memcheck from complaining about uninitialized value kwiberg@webrtc.org 2015-02-25 08:08:59 +00:00
  • 87a592dc50 Fix dependencies of media_file module and move gypi into the right dir to avoid submit warnings referencing files with '..'. andresp@webrtc.org 2015-02-25 03:18:23 +00:00
  • a4623d26d7 Fix H.264 HW decoding for Qualcomm KK devices. glaznev@webrtc.org 2015-02-25 00:02:50 +00:00
  • 49096de442 DCHECK send DataCountersUpdated for valid SSRCs. pbos@webrtc.org 2015-02-24 22:37:52 +00:00
  • 903182bd8e Revert r8476 "Set decoder output frequency in AudioDecoder::Decode call" henrik.lundin@webrtc.org 2015-02-24 21:17:50 +00:00
  • 348072845a Swap decl-terms from juberti@ review. lally@webrtc.org 2015-02-24 20:19:39 +00:00
  • 3630085df1 Tested equiv classes of DTLS/SCTP. lally@webrtc.org 2015-02-24 20:19:35 +00:00
  • 91d52305ac Renamed string and test. lally@webrtc.org 2015-02-24 20:19:30 +00:00
  • c7848b7fd1 Added a separate DTLS/SCTP test. lally@webrtc.org 2015-02-24 20:19:26 +00:00
  • a747093334 After another round of reviews. lally@webrtc.org 2015-02-24 20:19:21 +00:00
  • 9616196c38 Merging definitions of IsSctp. lally@webrtc.org 2015-02-24 20:19:16 +00:00
  • 12aa8a68f9 Post-rebase. lally@webrtc.org 2015-02-24 20:19:11 +00:00
  • 1730869596 Added raw SCTP to IsSctp. lally@webrtc.org 2015-02-24 20:19:06 +00:00
  • 871b1c373a Review comments -- added IsSctp() lally@webrtc.org 2015-02-24 20:18:59 +00:00
  • d7b6165483 Made DTLS/SCTP equivalent to UDP/DTLS/SCTP when comparing session descs in tests. lally@webrtc.org 2015-02-24 20:18:55 +00:00
  • ec97c6516f Attempt on read-only acceptance of -12. lally@webrtc.org 2015-02-24 20:18:48 +00:00
  • b9c18d5643 Set decoder output frequency in AudioDecoder::Decode call henrik.lundin@webrtc.org 2015-02-24 15:58:17 +00:00
  • f88791d783 AudioEncoderCng: CHECK that encode calls don't fail jmarusic@webrtc.org 2015-02-24 14:58:10 +00:00
  • 5e3fea1049 Fixing WebRTC engine demo JNI symbol export. phoglund@webrtc.org 2015-02-24 14:50:53 +00:00
  • a30f007e45 Fixing incorrect memset in mock class. phoglund@webrtc.org 2015-02-24 13:42:40 +00:00
  • a5de951b37 Make Options public and not package access in pc factory. phoglund@webrtc.org 2015-02-24 13:41:51 +00:00
  • db8e605c16 Break out BWE test models to separate files sprang@webrtc.org 2015-02-24 13:24:19 +00:00
  • ccd7c7c45d Remove more unused code in ACM henrik.lundin@webrtc.org 2015-02-24 12:01:34 +00:00
  • 13ca5f6db2 AudioEncoderOpus: CHECK that encode call doesn't fail jmarusic@webrtc.org 2015-02-24 09:56:58 +00:00
  • e3fccd4268 Merge changes from internal repo to AppRTCDemo. glaznev@webrtc.org 2015-02-24 00:53:04 +00:00
  • d324546ced Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ : * Move constants into the files/functions that use them * Declare variables in the narrowest scope possible * Use correct (expected, actual) order for gtest macros * Remove unused functions * Untabify * 80-column limit * Avoid C-style casts * Prefer true typed constants to "enum hack" constants * Print size_t using the right format macro * Shorten and simplify code * Other random cleanup bits and style fixes pkasting@chromium.org 2015-02-23 21:28:22 +00:00
  • 722739108a Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530) kjellander@webrtc.org 2015-02-23 19:08:31 +00:00
  • b28474c7a0 Add H.264 HW encoder and decoder support for Android. glaznev@webrtc.org 2015-02-23 17:44:27 +00:00
  • 77e11bbe83 Wire up preferred/nominal_bitrate to stats. pbos@webrtc.org 2015-02-23 16:39:07 +00:00
  • 829a6f4ac2 Merge ACMGenericCodec and ACMGenericCodecWrapper henrik.lundin@webrtc.org 2015-02-23 16:33:05 +00:00
  • f3a306b5bc g722: Enhanced documentation. Added CHECK. jmarusic@webrtc.org 2015-02-23 15:41:30 +00:00
  • 2acec4cc32 Enhanced documentation. Replaced DCHECK with CHECK. jmarusic@webrtc.org 2015-02-23 15:27:52 +00:00
  • 962c62475e Refactoring WebRTC Java/JNI audio track in C++ and Java. henrika@webrtc.org 2015-02-23 11:54:05 +00:00
  • 2ad3bb17a7 Reland patch for Switch default color format to YV12 on Android. The new since the previous patch is that we ignore all resolutions with width % 16 != 0 since they are not tightly packed. perkj@webrtc.org 2015-02-23 11:14:57 +00:00
  • 8278c072b6 Enable NACK under SendsAndReceivesH264. pbos@webrtc.org 2015-02-23 11:11:49 +00:00
  • fa58745445 Delete all codec-specific subclasses of ACMGenericCodec henrik.lundin@webrtc.org 2015-02-23 09:25:40 +00:00
  • 2a5cfc2167 Replaced unnecessary check with an explicit CHECK. WebRtcIlbcfix_Encode method that is called returns an error code only if a packet with more than 3 frames is passed, which is illegal. jmarusic@webrtc.org 2015-02-23 08:52:37 +00:00
  • 343096ac03 Fix incorrect rtx config in full_stack tests. sprang@webrtc.org 2015-02-23 08:34:17 +00:00
  • 1467421646 Fix for flaky test: VideoSendStreamTest.RtcpSenderReportContainsMediaBytesSent. asapersson@webrtc.org 2015-02-23 08:14:07 +00:00
  • 50e28166af Move SetTargetSendBitrates logic from default module to payload router. mflodman@webrtc.org 2015-02-23 07:45:11 +00:00
  • a43fce6e02 Add functions rtc::AtomicOps::Load and rtc::RefCountedObject::HasOneRef magjed@webrtc.org 2015-02-21 13:23:27 +00:00
  • 2af3057b24 Revert "When clearing the priority message queue, don't copy an item to itself." decurtis@webrtc.org 2015-02-21 01:59:50 +00:00
  • 2bffc3cb72 When clearing the priority message queue, don't copy an item to itself. decurtis@webrtc.org 2015-02-21 01:45:04 +00:00
  • d3a487c28b Exclude end-to-end test RestartingSendStreamPreservesRtpStatesWithRt on memcheck. marpan@webrtc.org 2015-02-20 19:57:04 +00:00
  • 3c4668e27d Amend CpuMonitor fix. torbjorng@webrtc.org 2015-02-20 14:16:54 +00:00
  • f906e55de1 Add CpuMonitor to Android ApprtcDemo torbjorng@webrtc.org 2015-02-20 13:15:09 +00:00
  • 7ac374abd7 Fix shutdown race for ViEEncoder when there is a frame in the encoder. mflodman@webrtc.org 2015-02-20 12:45:40 +00:00
  • dc77d7447e Disable FullStackTest.ForemanCifPlr5 temporarily while investigating flakiness. sprang@webrtc.org 2015-02-20 10:40:25 +00:00
  • ec45e3b290 Fix test race in GetStatsMultipleSendStreams. pbos@webrtc.org 2015-02-20 10:24:53 +00:00
  • 804eb46806 Change default from GICE to ICE5245 for SDP offers jlmiller@webrtc.org 2015-02-20 02:20:03 +00:00
  • d3d3baaa8e Copy SetThreadName from webrtc/base/thread.cc into thread_win.cc (webrtc/system_wrappers/source/thread_win.cc). It would be good to consolidate these helpers at some point. tommi@webrtc.org 2015-02-19 19:18:32 +00:00
  • 661af50dd5 Small Beamformer optimization aluebs@webrtc.org 2015-02-19 19:02:17 +00:00
  • cce874b8d2 Fix libjingle_media_unittest codec comparison issue guoweis@webrtc.org 2015-02-19 18:14:36 +00:00
  • bc6961fe32 Make webrtc 50 KB smaller by not inlining Codec. guoweis@webrtc.org 2015-02-19 17:55:18 +00:00