index
:
external/chromium_org/third_party/webrtc.git
idea133-weekly-release
l-preview
lollipop-cts-release
lollipop-dev
lollipop-mr1-cts-release
lollipop-mr1-dev
lollipop-mr1-fi-release
lollipop-mr1-release
lollipop-mr1-wfc-release
lollipop-release
lollipop-wear-release
main
master
master-soong
ub-webview-m40-release
summary
refs
log
tree
commit
diff
log msg
author
committer
range
path:
root
/
modules
/
audio_coding
Age
Commit message (
Expand
)
Author
2014-06-26
Receiver bit-exactness test for AudioCoding Module
henrik.lundin@webrtc.org
2014-06-25
This is to compare NetEq with various codecs under a shared packet loss pattern.
minyue@webrtc.org
2014-06-25
Remove payload duplication in AudioDecoderTest
henrik.lundin@webrtc.org
2014-06-24
Removing neteq decode lock and friends
henrik.lundin@webrtc.org
2014-06-24
Annotating the rest of AcmGenericCodec
henrik.lundin@webrtc.org
2014-06-23
Annotating the rest of AudioCodingModuleImpl
henrik.lundin@webrtc.org
2014-06-23
GN: Add BUILD.gn files + kjellander to OWNERS
kjellander@webrtc.org
2014-06-21
Roll chromium_revision 272489:277350 + fix sanitizer options
kjellander@webrtc.org
2014-06-19
Maintain constantness of the input to iSAC-fix decoder, and prevent heap-buff...
turaj@webrtc.org
2014-06-19
Adding an empty constructor implementation to the AudioSink class
henrik.lundin@webrtc.org
2014-06-19
Adding test::AudioSink interface and derived classes
henrik.lundin@webrtc.org
2014-06-18
Update PacketSource and RtpFileSource
henrik.lundin@webrtc.org
2014-06-18
Revert "Restore ptypes.txt file"
henrik.lundin@webrtc.org
2014-06-17
Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..."
minyue@webrtc.org
2014-06-17
Restore ptypes.txt file
henrik.lundin@webrtc.org
2014-06-17
Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-...
minyue@webrtc.org
2014-06-16
common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16
bjornv@webrtc.org
2014-06-16
Add thread annotations to parts of ACMGenericCodec
henrik.lundin@webrtc.org
2014-06-13
Pass GYP DEPTH variable to isolate.
kjellander@webrtc.org
2014-06-11
Re-land "Create a joint encoder/decoder wrapper for iSAC in ACM"
henrik.lundin@webrtc.org
2014-06-11
Revert r6377 "Create a joint encoder/decoder wrapper for iSAC in ACM"
henrik.lundin@webrtc.org
2014-06-11
common_audio/signal_processing: Removes macro WEBRTC_SPL_RSHIFT_U16
bjornv@webrtc.org
2014-06-11
common_audio/signal_processing: Moves WEBRTC_SPL_UMUL_16_16_RSFT16 to iSAC fix
bjornv@webrtc.org
2014-06-10
Delete last file in neteq4 folder
henrik.lundin@webrtc.org
2014-06-10
MIPS optimizations for ISAC (patch #1)
andrew@webrtc.org
2014-06-10
Add kjellander@webrtc.org as OWNER for *.isolate
kjellander@webrtc.org
2014-06-09
Create a joint encoder/decoder wrapper for iSAC in ACM
henrik.lundin@webrtc.org
2014-06-09
Add thread annotations to AcmReceiver
henrik.lundin@webrtc.org
2014-06-09
Multi-threaded unit test for Audio Coding Module using iSAC
henrik.lundin@webrtc.org
2014-06-09
Rename neteq4 folder to neteq
henrik.lundin@webrtc.org
2014-06-08
Re-enable AudioCodingModuleMtTest again
henrik.lundin@webrtc.org
2014-06-05
Fix the chain that propagates the audio frame's rtp and ntp timestamp including:
wu@webrtc.org
2014-06-05
Opus send rate overflows if over 65 kbps
tina.legrand@webrtc.org
2014-06-05
NetEq: Add thread annotation to const scoped_ptrs
henrik.lundin@webrtc.org
2014-06-05
The correct fix of workaround in r6261.
bjornv@webrtc.org
2014-06-05
common_audio/signal_processing: Removed macro WEBRTC_SPL_MUL_16_16_RSFT_WITH_...
bjornv@webrtc.org
2014-06-04
Android: cleanup gtest_target_type conditions.
henrike@webrtc.org
2014-06-03
Revert 6312 "Re-enable AudioCodingModuleMtTest"
turaj@webrtc.org
2014-06-03
Re-enable AudioCodingModuleMtTest
henrik.lundin@webrtc.org
2014-06-02
- Get rid of 'using' from .h
solenberg@webrtc.org
2014-06-02
Adding thread annotations to parts of Audio Coding Module
henrik.lundin@webrtc.org
2014-05-30
Disable AudioCodingModuleMtTest due to memcheck and tsan failures.
andrew@webrtc.org
2014-05-30
Multi-threaded test for Audio Coding Module
henrik.lundin@webrtc.org
2014-05-30
Fixing a bug regarding VOE packet loss rate feedback to ACM
minyue@webrtc.org
2014-05-28
Better buffer size estimation in NetEq for redundant packets
minyue@webrtc.org
2014-05-28
Revert 6257 "Rename neteq4 folder to neteq"
henrik.lundin@webrtc.org
2014-05-28
Rename neteq4 folder to neteq
henrik.lundin@webrtc.org
2014-05-26
Fix a bug preventing FilePlayer from playing encoded wav files
henrik.lundin@webrtc.org
2014-05-23
Avoid reading uninitialized values (outside baundary) in DFT arithmatic decod...
turaj@webrtc.org
2014-05-23
1. Make a clear distinction between codec internal FEC and RED, confusing men...
minyue@webrtc.org
[next]