Log - a4d873786f10eedd72de25ad0d94ad7c53c1f68a - platform/external/webrtc

e8fbc5d Refactor WebRtcOpus_PacketHasFec. by Minyue Li · 5 years ago
a1d1a1e WebRTC Opus C interface: Add support for non-48 kHz decode sample rate by Karl Wiberg · 5 years ago
7e7c5c3 WebRTC Opus C interface: Add support for non-48 kHz encode sample rate by Karl Wiberg · 5 years ago
e5b9416 Decoder for multistream Opus. by Alex Loiko · 5 years ago
50b8c39 Generalize the C-language Opus interface. by Alex Loiko · 5 years ago
7a3e43a Reland of Opus multistream. by Alex Loiko · 5 years ago
1fa51d6 Revert "Opus multistream." by Amit Hilbuch · 6 years ago
83ed89a Opus multistream. by Alex Loiko · 6 years ago
eeb2765 Implement Opus bandwidth adjustment behind a FieldTrial by Alex Luebs · 7 years ago
92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/modules/audio_coding/codecs/opus/opus_interface.c]
eaaae9e base->rtc_base: Update .c, .mm and .java files. by ehmaldonado · 7 years ago
28dc285 Adding cbr support for Opus by soren · 7 years ago
deaf6fb Opus: Let the decoder interpret 2-byte payloads as DTX/CNG packets by henrik.lundin · 7 years ago
9238245 Fix nr of bytes sent to Opus decoder in DTX mode by flim · 7 years ago
2e03c66 Adding build switch for Opus that supports 120ms ptime. by minyue · 7 years ago
41b9c80 Adding audio network adaptor to AudioEncoderOpus. by minyue · 8 years ago
c8299f9 Posting Opus's set-force-channels functionality to WebRTC. by minyue · 8 years ago
2e48646 RTC_CHECK and RTC_DCHECK macros for C by kwiberg · 8 years ago
64a7eab Update tests and DTX check for Opus 1.1.3. by flim · 8 years ago
7e937e9 Remove workaround for Opus DTX noise pumping issue. by minyuel · 8 years ago
6955870 Convert channel counts to size_t. by Peter Kasting · 9 years ago
3c652b6 modules/audio_coding: Remove some codec include dirs by kjellander@webrtc.org · 9 years ago
3cea256 Reland "Prevent Opus DTX from generating intermittent noise during silence" by minyue · 9 years ago
b4a753f Revert of Prevent Opus DTX from generating intermittent noise during silence (patchset #10 id:250001 of https://codereview.webrtc.org/1415173005/ ) by kjellander · 9 years ago
f475add Prevent Opus DTX from generating intermittent noise during silence. by minyue · 9 years ago
7464089 audio_coding: rename interface -> include by Henrik Kjellander · 9 years ago
6d92bf5 Returning correct duration estimate on Opus DTX packets. by minyuel · 9 years ago
4376648 AudioDecoder: Replace Init() with Reset() by Karl Wiberg · 9 years ago
dce40cf Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 9 years ago
bba7807 Reland "Upconvert various types to int.", misc. codecs portion. by Peter Kasting · 9 years ago
cb18097 Revert "Upconvert various types to int." by Peter Kasting · 9 years ago
83ad33a Upconvert various types to int. by Peter Kasting · 9 years ago
092041c Setting OPUS_SIGNAL_VOICE when enable DTX. by Minyue Li · 9 years ago
7dba786 Setting Opus target application. by minyue@webrtc.org · 10 years ago
0ca768b Adding DTX to WebRTC Opus wrapper (relanding). by minyue@webrtc.org · 10 years ago
19dd129 Revert 7846 "Adding DTX to WebRTC Opus wrapper" by minyue@webrtc.org · 10 years ago
4321f17 Adding DTX to WebRTC Opus wrapper by minyue@webrtc.org · 10 years ago
33ccdfa Relanding r7807. by minyue@webrtc.org · 10 years ago
52bc4f4 Revert 7807 "Removing unused opus wrapper APIs." by minyue@webrtc.org · 10 years ago
e54a634 Removing unused opus wrapper APIs. by minyue@webrtc.org · 10 years ago
4bd2db9 Opus wrapper: Use const for inputs and uint8[] for byte streams by kwiberg@webrtc.org · 10 years ago
adee8f9 Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate by minyue@webrtc.org · 10 years ago
0040a6e This is a setup to solve https://code.google.com/p/webrtc/issues/detail?id=1906 by minyue@webrtc.org · 10 years ago
f563e85 This is to re-open an earlier CL by minyue@webrtc.org · 10 years ago
d42da54 Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..." by minyue@webrtc.org · 10 years ago
8f8503d Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling. by minyue@webrtc.org · 10 years ago
46509c8 adding FEC support to WebRTC Opus wrapper and tests. by minyue@webrtc.org · 10 years ago
0454688 This CL is to add Opus complexity knob and to test it. by minyue@webrtc.org · 10 years ago
bd21fb5 Adding call to Opus PLC by tina.legrand@webrtc.org · 11 years ago
45426ea In call to Opus decoder: frame length too large by tina.legrand@webrtc.org · 11 years ago
db11fab Adding Opus unit test by tina.legrand@webrtc.org · 11 years ago
46d90dc Adding three frame sizes to Opus by tina.legrand@webrtc.org · 11 years ago
5dfb1f2 Bug fix in WebRtcOpus_DurationEst by henrik.lundin@webrtc.org · 12 years ago
d0d4149 Adding AUDIO application as default for Opus stereo by tina.legrand@webrtc.org · 12 years ago
4275ab1 Implement NetEq duration estimation for Opus. by tina.legrand@webrtc.org · 12 years ago
c459058 Opus mono/stereo on the same payloadtype, and fix of memory bug by tina.legrand@webrtc.org · 12 years ago
0ad3c1a Adding Opus stereo support to WebRTC by tina.legrand@webrtc.org · 12 years ago
14b43be Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago[Renamed from src/modules/audio_coding/codecs/opus/opus_interface.c]
a7d8387 Opus integration by tina.legrand@webrtc.org · 12 years ago