Log - 86eb7d94f3a3facfefef7a59ee549f1935a396d3 - src/webrtc

86eb7d9 Move isacfix.gypi and isac.gypi by kjellander@webrtc.org · 10 years ago
121d8e7 Set webrtc_rtp category to be disabled by default. by sprang@webrtc.org · 10 years ago
c6688d2 Break out code from bloated files in the BWE simulator. by stefan@webrtc.org · 10 years ago
e9aaba8 Add audio_coding module OWNERS file. by kjellander@webrtc.org · 10 years ago
21b6665 Revert r8342 "Switch to using AudioEncoderIsac instead of ACMISAC" by henrik.lundin@webrtc.org · 10 years ago
8a687f9 Simplify mask calculation by aluebs@webrtc.org · 10 years ago
ee1189e Add support for bi-directional simulations by having both an uplink and a downlink. by stefan@webrtc.org · 10 years ago
fae177c Remove EventWrapper::Reset(). by pbos@webrtc.org · 10 years ago
acabb06 This is a code clean up. No functional change intended. by guoweis@webrtc.org · 10 years ago
994ded1 Allowing RED decoding for Opus. by minyue@webrtc.org · 10 years ago
c8b6b97 Re-enable BWE tests using baseline files. by solenberg@webrtc.org · 10 years ago
ac525a3 WebRTC now compiles for enable_android_opensl=1. by henrika@webrtc.org · 10 years ago
b5a63a0 audio_coding/codec/ilbc: Removed usage of macro WEBRTC_SPL_MUL_16_16 by bjornv@webrtc.org · 10 years ago
45da874 Remove call to RtpRtcp::RegisterSendPayload for the default RTP module. by mflodman@webrtc.org · 10 years ago
1e7cc15 Add a unit test for callbacks with empty frames and fix bug in code by henrik.lundin@webrtc.org · 10 years ago
3702141 Remove temporary GYP targets by kjellander@webrtc.org · 10 years ago
2991d5f Fix problem where Android VoE can not record on multiple channels. by perkj@webrtc.org · 10 years ago
41cb873 Remove potential deadlock in RTPSenderAudio. by pbos@webrtc.org · 10 years ago
82730e0 Remove getting max payload length from default module. by mflodman@webrtc.org · 10 years ago
7773624 Switch to using AudioEncoderIsac instead of ACMISAC by henrik.lundin@webrtc.org · 10 years ago
cbcc5a9 Switch to using AudioEncoderOpus instead of ACMOpus by henrik.lundin@webrtc.org · 10 years ago
fd7343a CVO capturer feature: allow unrotated frame flows through the capture pipeline. by guoweis@webrtc.org · 10 years ago
27bd438 CVO capturer feature: allow unrotated frame flows through the capture pipeline. by guoweis@webrtc.org · 10 years ago
42892a6 CVO capturer feature: allow unrotated frame flows through the capture pipeline. by guoweis@webrtc.org · 10 years ago
3fc15a6 Remove default RTP module functionality for setting CSRC. by mflodman@webrtc.org · 10 years ago
d25ffba Don't rely on webrtc/base/scoped_ptr.h to include stuff for you by kwiberg@webrtc.org · 10 years ago
a2b007b Switch to using AudioEncoderG722 instead of ACMG722 by henrik.lundin@webrtc.org · 10 years ago
579727c Refactoring WebRTC Java/JNI audio recording in C++ and Java. by henrika@webrtc.org · 10 years ago
d3c7c06 Switch to using AudioEncoderPcm16B instead of ACMPCM16B by henrik.lundin@webrtc.org · 10 years ago
12971d4 Rename GYP and GN targets for video capture+render. by kjellander@webrtc.org · 10 years ago
0d1749a Fix bug when there are no blocks in a chunk in Beamformer by aluebs@webrtc.org · 10 years ago
ebbb0c2 Make ChannelBuffer aware of frequency bands by aluebs@webrtc.org · 10 years ago
6b3f73d Make sure that the norms are positive in Beamformer by aluebs@webrtc.org · 10 years ago
e231c2b Apply mask smoothing in Beamformer by aluebs@webrtc.org · 10 years ago
8e21021 Switch to using AudioEncoderIlbc instead of ACMILBC by henrik.lundin@webrtc.org · 10 years ago
1f5b96b Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig. by stefan@webrtc.org · 10 years ago
e6bf778 pcm16b: Make input arrays const and use uint8_t[] for byte arrays by kwiberg@webrtc.org · 10 years ago
c4fb2c7 Fixing a bug in expand_rate calculation for stereo signal. by minyue@webrtc.org · 10 years ago
ada4696 Revert r8297 "Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig." by stefan@webrtc.org · 10 years ago
55f173b Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig. by stefan@webrtc.org · 10 years ago
89ded13 Fixing a nit by henrik.lundin@webrtc.org · 10 years ago
9c01449 G711: Make input arrays const and use uint8_t[] for byte arrays by kwiberg@webrtc.org · 10 years ago
3d8c87f Restructure GYP for vp9, opus and direct trace by kjellander@webrtc.org · 10 years ago
972ccb8 Remove default arguments in EncodedImageCallback. by changbin.shao@webrtc.org · 10 years ago
e898b13 Fix the binary layout of ProcessThreadImpl. by tommi@webrtc.org · 10 years ago
81b6b6a Disable ProcessThread tests that are dependent on timing. by tommi@webrtc.org · 10 years ago
8684a45 Normalize delay-and-sum mask in Beamformer by aluebs@webrtc.org · 10 years ago
a417d39 Add high frequency correction to Beamformer by aluebs@webrtc.org · 10 years ago
5009286 audio_processing: Now records mic volume level also when using new AGC by bjornv@webrtc.org · 10 years ago
bd7e1f3 Switch to using AudioEncoderPcmU/A instead of ACMPCMU/A by henrik.lundin@webrtc.org · 10 years ago
c51222c Implementing a packet router class, used to route RTP packets to the by mflodman@webrtc.org · 10 years ago
21a8036 Fix delete of stack allocated object causing test crashes. by stefan@webrtc.org · 10 years ago
235633e Wire up new feedback format by introducing a FeedbackPacket type. by stefan@webrtc.org · 10 years ago
b46f229 audio_processing/agc: Changed to correct include path in agc_unittests by bjornv@webrtc.org · 10 years ago
0776556 Update ProcessCallAfterXms to better match the performance of our faster bots. Previously I had made sure these tests didn't flake out on our slow trybots, but apparently I need to do the same for the fast bots :) by tommi@webrtc.org · 10 years ago
4de6286 Revamp the ProcessThreadImpl implementation. by tommi@webrtc.org · 10 years ago
c2b6512 Remove defined(__cplusplus) tests in C++ code. by jan.skoglund@webrtc.org · 10 years ago
97ed521 Reland r8248 "Introduce ACMGenericCodecWrapper" by henrik.lundin@webrtc.org · 10 years ago
a3a54be Clean up Beamformer initialization by aluebs@webrtc.org · 10 years ago
49da516 voice_engine: Updates GetEcDelayMetrics() w.r.t. new metric by bjornv@webrtc.org · 10 years ago
684f85d Revert r8248 "Introduce ACMGenericCodecWrapper" by henrik.lundin@webrtc.org · 10 years ago
2b9479f Introduce ACMGenericCodecWrapper by henrik.lundin@webrtc.org · 10 years ago
7260796 Add new AudioEncoderOpusTest by henrik.lundin@webrtc.org · 10 years ago
22b04fa Remove SetNotAlive method from the thread class. by tommi@webrtc.org · 10 years ago
1a8794b Add method for incrementing RtpPacketCounter. Removes duplicate code. by asapersson@webrtc.org · 10 years ago
1514b94 Rewrite ThreadPosix. by tommi@webrtc.org · 10 years ago
80b9d48 Remove temp files in audio_processing_unittest.cc. by pbos@webrtc.org · 10 years ago
1c39714 Enable bitrate probing by default. by stefan@webrtc.org · 10 years ago
411650c audio_processing: Added a new AEC delay metric value that gives the amount of poor delays by bjornv@webrtc.org · 10 years ago
e00a87e Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting. by pkasting@chromium.org · 10 years ago
fb7023d Switched lists of packets to lists of packet pointers. Allows Packet polymorphism. by stefan@webrtc.org · 10 years ago
d5612b9 Fixed potential crash if rtp packet history is completely full. by sprang@webrtc.org · 10 years ago
ef7fec7 Change name for local CriticalSectionScoped variable by henrik.lundin@webrtc.org · 10 years ago
6850ee8 WebRtcG722_Decode: Input array should be const uint8_t[] by kwiberg@webrtc.org · 10 years ago
b25aac2 Using << on an int8_t or uint8_t will output a character rather than a number. by pkasting@chromium.org · 10 years ago
185eee3 Refactor senders into senders and sources in the simulation framework. by stefan@webrtc.org · 10 years ago
3fc7958 Fix a bug in ACM test channel by henrik.lundin@webrtc.org · 10 years ago
6cb1592 Reland r8210 "Add a new parameter to ACMGenericCodec constructor"" by henrik.lundin@webrtc.org · 10 years ago
29882bb WebRtcIsacfix_Time2SpecNeon and _Spec2TimeNeon: Fix stack alignment by henrik.lundin@webrtc.org · 10 years ago
42b18c5 Revert r8210 "Add a new parameter to ACMGenericCodec constructor" by henrik.lundin@webrtc.org · 10 years ago
800659a Add a new parameter to ACMGenericCodec constructor by henrik.lundin@webrtc.org · 10 years ago
4567cac Add arbitrary microphone geometry input to audioproc_f test utility. by mgraczyk@chromium.org · 10 years ago
1a48d11 Add new members to AudioEncoderOpus::Config by henrik.lundin@webrtc.org · 10 years ago
6ae75ba Re-land "Remove <(webrtc_root) from source file entries." by kjellander@webrtc.org · 10 years ago
e2a4d47 Fix a number of things in AudioEncoderDecoderIsac* by henrik.lundin@webrtc.org · 10 years ago
7b17831 Remove ChangeUniqueID. by tommi@webrtc.org · 10 years ago
8a13084 Revert "Remove <(webrtc_root) from source file entries." by kjellander@webrtc.org · 10 years ago
0224142 Move audio_codec_speed_tests into include_tests==1 condition. by kjellander@webrtc.org · 10 years ago
2645d28 Remove <(webrtc_root) from source file entries. by kjellander@webrtc.org · 10 years ago
bbf5cad Allow rtp packet history to dynamically expand in size. by sprang@webrtc.org · 10 years ago
f98fb80 Add case to ApmTest.Process to test the extended filter mode by aluebs@webrtc.org · 10 years ago
d40d97b Move channel_buffer.{h,cc} to common_audio. by kjellander@webrtc.org · 10 years ago
2234f41 Enable Clang warning implicit-fallthrough and annotate the code. by kjellander@webrtc.org · 10 years ago
f924d43 Reland r8125: Modify some tests to never use DTX disable mode by henrik.lundin@webrtc.org · 10 years ago
413575d Notify jitter buffer about received FEC packets (to avoid sending NACK request for these packets). by asapersson@webrtc.org · 10 years ago
3f4c99c Add support for 40 and 60 ms frames to AudioEncoderIlbc by henrik.lundin@webrtc.org · 10 years ago
cce642c Make sure ByteReader<T>::Read* is properly constified. by sprang@webrtc.org · 10 years ago
f132be8 Adjust parameter in videoprocessor_integrationtest for VP9. by marpan@webrtc.org · 10 years ago
4154c79 Adjust qp-max settinhg in VP9 wrapper. by marpan@webrtc.org · 10 years ago
846183f Minor updates to AudioEncoderCng by henrik.lundin@webrtc.org · 10 years ago