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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
eb166972591504a230baae416bc37b674bb20c2a
/
modules
/
audio_coding
/
codecs
/
opus
/
audio_encoder_opus.cc
eb16697
AudioEncoderOpus: Don't mix up sample rate and RTP timestamp rate
by Karl Wiberg
· 5 years ago
e5b9416
Decoder for multistream Opus.
by Alex Loiko
· 5 years ago
e45c688
Remove webrtc::ProtoString.
by Mirko Bonadei
· 6 years ago
9dac02d
Adding text log on actual opus bitrate.
by Minyue Li
· 6 years ago
10542f2
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
by Steve Anton
· 6 years ago
41f3a43
Remove CodecInst pt.3
by Fredrik Solenberg
· 6 years ago
8ac05cc
Adds trial to use link capacity estimate in Opus encoder.
by Sebastian Jansson
· 6 years ago
988cc08
[Cleanup] Add missing #include. Remove useless ones.
by Yves Gerey
· 6 years ago
2edab4c
Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase.
by Niels Möller
· 6 years ago
83bd37c
Add field trials for configuring Opus encoder packet loss rate.
by Jakob Ivarsson
· 6 years ago
88b68ac
Create field trial for setting a minimum value for Opus encoder packet loss rate
by Jakob Ivarsson
· 6 years ago
918f50c
Use absl::make_unique and absl::WrapUnique directly
by Karl Wiberg
· 6 years ago
665174f
Reformat the WebRTC code base
by Yves Gerey
· 6 years ago
b602123
Replace rtc::Optional with absl::optional in modules/audio_coding
by Danil Chapovalov
· 6 years ago
e40468b
Move some numeric utility code from rtc_base/ to rtc_base/numerics/
by Karl Wiberg
· 7 years ago
eeb2765
Implement Opus bandwidth adjustment behind a FieldTrial
by Alex Luebs
· 7 years ago
36de62e
Avoid flagging Opus DTX frames as speech.
by Gustaf Ullberg
· 7 years ago
12ab00b
Optional: Use nullopt and implicit construction in /modules/audio_coding
by Oskar Sundbom
· 7 years ago
675513b
Stop using LOG macros in favor of RTC_ prefixed macros.
by Mirko Bonadei
· 7 years ago
7275e18
Hide the internal AudioEncoderOpus class by giving it an "Impl" suffix
by Karl Wiberg
· 7 years ago
7120742
Adding NOLINT for typedefs.h and common_types.h
by Mirko Bonadei
· 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/audio_encoder_opus.cc]
e1198e0
Add new ANA stats to the old GetStats() to count the number of actions taken by each controller.
by ivoc
· 7 years ago
bf94fda
Renaming probing_interval to bwe_period globally.
by minyue-webrtc
· 7 years ago
5d68910
Don't use rvalue reference function arguments in the audio coding module
by minyue-webrtc
· 7 years ago
ee89e78
Replace CHECK(x && y) with two separate CHECK() calls
by kwiberg
· 7 years ago
54348fb
Removed an obsolete DCHECK in AudioEncoderOpus.
by tschumim
· 7 years ago
c20978e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 7 years ago
dca1e09
Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)"
by Henrik Kjellander
· 7 years ago
c8fa692
Update includes for webrtc/{base => rtc_base} rename (1/3)
by kjellander
· 7 years ago
96da011
Opus implementation of the AudioEncoderFactoryTemplate API
by kwiberg
· 7 years ago
1a610f1
Revert of Opus implementation of the AudioEncoderFactoryTemplate API (patchset #4 id:80001 of https://codereview.webrtc.org/2930243003/ )
by charujain
· 7 years ago
fe1aa82
Opus implementation of the AudioEncoderFactoryTemplate API
by kwiberg
· 7 years ago
0703856
Add SafeClamp(), which accepts args of different types
by kwiberg
· 7 years ago
0d6195d
Hooked up Opus CBR support when configured manually or through an SdpAudioFormat.
by ossu
· 7 years ago
92aef17
Replace Clock with timeutils in AudioEncoder.
by michaelt
· 7 years ago
7c2c843
Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ )
by mbonadei
· 7 years ago
a1a040a
Injectable audio encoders: BuiltinAudioEncoderFactory
by ossu
· 7 years ago
d00aad5
Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )
by mbonadei
· 7 years ago
16ab93b
To accommodate some downstream WebRTC users we need to loosen
by mbonadei
· 7 years ago
dadb4dc
Allow ANA to receive RPLR (recoverable packet loss rate) indications
by elad.alon
· 7 years ago
cfd88bb
Fix AudioEncoderOpus::RecreateEncoderInstance() referring to old config_
by elad.alon
· 8 years ago
c1b57a1
Test field trial group with startswith rather than equals.
by sprang
· 8 years ago
6f08d7d
Change frame length on very low bandwidth.
by michaelt
· 8 years ago
a55f021
Add 120ms frame ability to ANA
by michaelt
· 8 years ago
54340d8
Change opus min bitrate.
by michaelt
· 8 years ago
0fe1216
Move more calls to webrtc::field_trial::FindFullName into ctor, thereby minimizing the non-trivial cost of repeated string comparisons.
by elad.alon
· 8 years ago
bc5d921
Rename base/analytics/ to base/numerics/
by terelius
· 8 years ago
bf279fc
Pass event log to ANA.
by michaelt
· 8 years ago
566d820
Update smoothed bitrate.
by michaelt
· 8 years ago
584c35a
Make a the decisions of ANA optional for the opus encoder.
by michaelt
· 8 years ago
eca373f
Adding OnReceivedOverhead to AudioEncoder.
by minyue
· 8 years ago
4b9a2cb
Reland "Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate."
by minyue
· 8 years ago
e69b468
Revert of Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. (patchset #5 id:240001 of https://codereview.webrtc.org/2411613002/ )
by minyue
· 8 years ago
84e56d5
Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate.
by minyue
· 8 years ago
af476c7
RTC_[D]CHECK_op: Remove "u" suffix on integer constants
by kwiberg
· 8 years ago
69b627d
Move smoothing filter to common audio and exp_filter to base/analytics.
by minyue
· 8 years ago
3c3aef4
Revert of Reland "Move smoothing filter to common audio". (patchset #5 id:100001 of https://codereview.webrtc.org/2520003005/ )
by minyue
· 8 years ago
223641f
Reland "Move smoothing filter to common audio".
by minyue
· 8 years ago
875862c
Let Opus increase complexity for low bitrates
by henrik.lundin
· 8 years ago
a6f495c
Simplifying audio network adaptor by moving receiver frame length range to ctor.
by minyue
· 8 years ago
7e30432
Hooking up audio network adaptor to VoE.
by minyue
· 8 years ago
41b9c80
Adding audio network adaptor to AudioEncoderOpus.
by minyue
· 8 years ago
85228d6
Regression test for issue where Opus DTX status was being forgotten.
by ivoc
· 8 years ago
2036135
AudioEncoderOpus: Default to 32 kbit/s for mono, 64 kbit/s for stereo
by kwiberg
· 8 years ago
8bce67b
Added UMA logging for audio codec usage. A histogram statistic named "WebRTC.Audio.Encoder.CodecType" has been created.
by aleloi
· 8 years ago
264087f
A few small cleanups of stuff caught by lint
by ossu
· 8 years ago
2903ba5
Reland Remove the deprecated EncodeInternal interface from AudioEncoder
by ossu
· 8 years ago
164bc4b
Revert of Remove the deprecated EncodeInternal interface from AudioEncoder (patchset #4 id:60001 of https://codereview.webrtc.org/1864993002/ )
by ossu
· 8 years ago
5222d31
Remove the deprecated EncodeInternal interface from AudioEncoder
by ossu
· 8 years ago
d44c077
Revert of Safe numeric library: base/numerics (copied from Chromium) (patchset #11 id:250001 of https://codereview.webrtc.org/1753293002/ )
by Tommi
· 9 years ago
de1c81b
Safe numeric library added: base/numerics (copied from Chromium)
by hbos
· 9 years ago
4f43fcf
Renamed new EncodeInternal to EncodeImpl to ensure proper backwards compatibility.
by ossu
· 9 years ago
10a029e
Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size.
by ossu
· 9 years ago
6955870
Convert channel counts to size_t.
by Peter Kasting
· 9 years ago
25702cb
Misc. small cleanups.
by pkasting
· 9 years ago
3c652b6
modules/audio_coding: Remove some codec include dirs
by kjellander@webrtc.org
· 9 years ago
288886b
Pass audio to AudioEncoder::Encode() in an ArrayView
by kwiberg
· 9 years ago
7464089
audio_coding: rename interface -> include
by Henrik Kjellander
· 9 years ago
91d6ede
Add RTC_ prefix to (D)CHECKs and related macros.
by henrikg
· 9 years ago
3f5f1c2
Change return type of AudioEncoder::SetMaxPlaybackRate to void
by kwiberg
· 9 years ago
12cfc9b
Fold AudioEncoderMutable into AudioEncoder
by kwiberg
· 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
b297c5a
Miscellaneous changes split from https://codereview.webrtc.org/1230503003 .
by pkasting
· 9 years ago
3e89dbf
Add AudioEncoder::GetTargetBitrate
by Henrik Lundin
· 9 years ago
bba7807
Reland "Upconvert various types to int.", misc. codecs portion.
by Peter Kasting
· 9 years ago
728d903
Reformat existing code. There should be no functional effects.
by Peter Kasting
· 9 years ago
b7e5054
Match existing type usage better.
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
dcccab3
New interface: AudioEncoderMutable
by Karl Wiberg
· 9 years ago
9afaee7
Reland 8749: AudioEncoder: return EncodedInfo from Encode() and EncodeInternal()
by jmarusic@webrtc.org
· 10 years ago
019955d
Revert 8749 "We changed Encode() and EncodeInternal() return typ..."
by tommi@webrtc.org
· 10 years ago
0cb612b
We changed Encode() and EncodeInternal() return type from bool to void in this issue:
by jmarusic@webrtc.org
· 10 years ago
e16bfde
Adding flag to force Opus application and DTX while toggling.
by minyue@webrtc.org
· 10 years ago
51ccf37
AudioEncoder: add method MaxEncodedBytes
by jmarusic@webrtc.org
· 10 years ago
c86bbba
Add speech flag to EncodedInfo
by henrik.lundin@webrtc.org
· 10 years ago
0561716
Adding Opus DTX support in ACM.
by minyue@webrtc.org
· 10 years ago
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