webrtc/modules/audio_coding/codecs/opus
Sebastian Jansson c3eb9fd49f Reland "Reland "Only include overhead if using send side bandwidth estimation.""
This is a reland of 086055d0fd

ANA was accitendly disabled even when transport sequence numbers were
negotiated due to a bug in how the audio send stream is configured. To
solve this we simply continue to always allow enabling ANA and leave it
up to the application to ensure that it's not used together with receive
side estimation.

Original change's description:
> Reland "Only include overhead if using send side bandwidth estimation."
>
> This is a reland of 8c79c6e1af
>
> Original change's description:
> > Only include overhead if using send side bandwidth estimation.
> >
> > Bug: webrtc:11298
> > Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30382}
>
> Bug: webrtc:11298
> Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30390}

Bug: webrtc:11298
Change-Id: If2ad91e17ebfc85dc51edcd9607996e18c5d1f13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167883
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30413}
2020-01-29 18:42:34 +00:00
..
test Use source_sets in component builds and static_library in release builds. 2019-10-17 21:17:18 +00:00
audio_coder_opus_common.cc Decoder for multistream Opus. 2019-04-08 16:15:37 +00:00
audio_coder_opus_common.h Decoder for multistream Opus. 2019-04-08 16:15:37 +00:00
audio_decoder_multi_channel_opus_impl.cc WebRTC Opus C interface: Add support for non-48 kHz decode sample rate 2019-05-29 10:33:03 +00:00
audio_decoder_multi_channel_opus_impl.h Format almost everything. 2019-07-08 13:45:15 +00:00
audio_decoder_multi_channel_opus_unittest.cc Fully qualify googletest symbols. 2019-04-09 17:18:20 +00:00
audio_decoder_opus.cc AudioDecoderOpus: Add support for 16 kHz output sample rate 2019-05-29 12:42:38 +00:00
audio_decoder_opus.h Format almost everything. 2019-07-08 13:45:15 +00:00
audio_encoder_multi_channel_opus_impl.cc Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
audio_encoder_multi_channel_opus_impl.h Encoder side of Multistream Opus. 2019-04-25 15:07:38 +00:00
audio_encoder_multi_channel_opus_unittest.cc Encoder side of Multistream Opus. 2019-04-25 15:07:38 +00:00
audio_encoder_opus_unittest.cc Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
audio_encoder_opus.cc Reland "Reland "Only include overhead if using send side bandwidth estimation."" 2020-01-29 18:42:34 +00:00
audio_encoder_opus.h Reland "Reland "Only include overhead if using send side bandwidth estimation."" 2020-01-29 18:42:34 +00:00
opus_bandwidth_unittest.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
opus_complexity_unittest.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
opus_fec_test.cc Make Opus PLC always output 10ms audio. 2019-11-07 21:15:58 +00:00
opus_inst.h Make Opus PLC always output 10ms audio. 2019-11-07 21:15:58 +00:00
opus_interface.cc Add include of <cstdlib> 2020-01-07 14:46:03 +00:00
opus_interface.h Adding GetInDtx to WebRTC Opus Interface. 2019-11-19 14:14:06 +00:00
opus_speed_test.cc WebRTC Opus C interface: Add support for non-48 kHz decode sample rate 2019-05-29 10:33:03 +00:00
opus_unittest.cc Expose NetEqDecodingTest for re-use in chromium tests. 2019-11-11 17:45:46 +00:00