ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
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04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames This changes the following module directories: * webrtc/modules/audio_conference_mixer/interface * webrtc/modules/interface * webrtc/modules/media_file/interface * webrtc/modules/rtp_rtcp/interface * webrtc/modules/utility/interface To avoid breaking downstream, I followed this recipe: 1. Copy the interface dir to a new sibling directory: include 2. Update the header guards in the include directory to match the style guide. 3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code. 4. Add a pragma warning in the header files in the interface dir. Example: #pragma message("WARNING: webrtc/modules/interface is DEPRECATED; " "use webrtc/modules/include") 5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S) 6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*) BUG=5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417683006 . Cr-Commit-Position: refs/heads/master@{#10500}
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
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5eb9d578839e502e989615186b800d9d1074ea99 |
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03-Nov-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
Re-enable PCAP reading in neteq_rtpplay Reading of PCAP (Wireshark) files was not possible due to a bug in the parsing of files. This change fixes that by adding new validator methods to RtpFileSource that can be used to determine the input file type. R=ivoc@webrtc.org Review URL: https://codereview.webrtc.org/1427923003 Cr-Commit-Position: refs/heads/master@{#10490}
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
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11d583f41484913fd1e7b3e283966eb7b7e11ed2 |
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18-Sep-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
Fix a bug in RtpFileSource related to RTCP packets in rtpdump files According to http://www.cs.columbia.edu/irt/software/rtptools/#rtpdump, RTCP packets are marked with plen==0. In this class, plen is mapped to original_length, not length. Review URL: https://codereview.webrtc.org/1356543002 Cr-Commit-Position: refs/heads/master@{#9981}
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
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91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
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17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros. We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition. Alternative solutions: * Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable. * Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce. * Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable. * Changes in Chromium for this is obviously not an option. BUG=chromium:468375 NOTRY=true Review URL: https://codereview.webrtc.org/1335923002 Cr-Commit-Position: refs/heads/master@{#9964}
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
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00b8f6b3643332cce1ee711715f7fbb824d793ca |
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26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away BUG= R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/36229004 Cr-Commit-Position: refs/heads/master@{#8517} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
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91d928e737732f7ad71c335da9a1c8b58f3a7701 |
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26-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader This is in preparation for creating a new class RtpFileWriter which will use the same RtpPacket struct. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7749 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
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4b133da5fd37a93de2f191ef340fd105e6f83672 |
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02-Oct-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Let RtpFileSource use RtpFileReader RtpFileSource used to implement it's own RTP dump file reader, but with this change it simply uses RtpFileReader. One benefit is that pcap files are now also supported. All NetEq test tools that use RtpFileSource are updated. BUG=2692 R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22839004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7367 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
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c8e98187d1751a0ed31a0e76ea80564c4e4a4c04 |
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26-Jun-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Receiver bit-exactness test for AudioCoding Module This CL introduces a bit-exactness test for the receive-side of the AudioCoding Module. The main part of the test is done in the helper class AcmReceiveTest. The test is executed from the test fixture AcmReceiverBitExactness. The test inserts packets from a pre-encoded RTP file. The output is summed up into a checksum, which is verified versus a reference at the end of the test. Alternatively, if the flag --generate_output is given, the output is written to a file for subjective verification. R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6549 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
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12396aba4262a03dbdaf9fce3e6bedbfaad7e86d |
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18-Jun-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update PacketSource and RtpFileSource The NextPacket method should now return NULL when the end of the source was reached. In the RtpFileSource, this means that when the end of file is reached, NULL is returned. Also, when an RTCP packet is encountered, the next packet will be read from file immediately. R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20699004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6479 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
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9c55f0f957534144d2b8a64154f0a479249b34be |
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09-Jun-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename neteq4 folder to neteq Keep the old neteq4/audio_decoder_unittests.isolate while waiting for a hard-coded reference to change. This CL effectively reverts r6257 "Rename neteq4 folder to neteq". BUG=2996 TBR=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
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1b9df05c8521d1d807b08d7c00eb2f7e5b097fdf |
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28-May-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6257 "Rename neteq4 folder to neteq" > Rename neteq4 folder to neteq > > BUG=2996 > R=turaj@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/12569005 TBR=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13549004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6259 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
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a90f6d67f72359cf63b59480fa87a13aae808c03 |
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28-May-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename neteq4 folder to neteq BUG=2996 R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12569005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6257 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
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