webrtc/api/audio_codecs
Alex Loiko 65438812ba 2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883
The difference to the original is new bitexactness strings.  The
reason for reland is breaking downstream projects.

Original CL description:

Tests for multi-stream Opus.

This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
tests are in audio_coding_unittest.cc. Some refactoring of
AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
possible. A few checks for "channels \in {1, 2}" are replaced with
"channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
other changes are made to be able to write and read multi-channel WAV
files.

The SDP changes are NOT included; as of this CL there is no way to set
up a multi-stream opus en/de-coder from SDP strings.

TBR=ossu@webrtc.org

Bug: webrtc:8649
Change-Id: I6261b18c69fd666d43ab34ed8f1bc9d5cc82b21f
Reviewed-on: https://webrtc-review.googlesource.com/c/123882
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26809}
2019-02-22 09:59:01 +00:00
..
g711 Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. 2018-10-23 09:24:15 +00:00
g722 Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. 2018-10-23 09:24:15 +00:00
ilbc Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. 2018-10-23 09:24:15 +00:00
isac Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. 2018-10-23 09:24:15 +00:00
L16 Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. 2018-10-23 09:24:15 +00:00
opus 2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883 2019-02-22 09:59:01 +00:00
test Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
audio_codec_pair_id.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
audio_codec_pair_id.h Audio codec factories: Pass a codec pair ID to new codecs 2018-03-01 12:23:28 +00:00
audio_decoder.cc Let NetEq use the PLC output from a decoder 2018-09-14 07:05:20 +00:00
audio_decoder.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
audio_decoder_factory.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
audio_decoder_factory_template.h Remove rtc_base/scoped_ref_ptr.h. 2019-01-25 20:29:58 +00:00
audio_encoder.cc Adds OnReceivedUplinkAllocation method to AudioEncoder. 2018-11-21 20:46:01 +00:00
audio_encoder.h Adds OnReceivedUplinkAllocation method to AudioEncoder. 2018-11-21 20:46:01 +00:00
audio_encoder_factory.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
audio_encoder_factory_template.h Remove rtc_base/scoped_ref_ptr.h. 2019-01-25 20:29:58 +00:00
audio_format.cc [clang-tidy] Apply performance-move-const-arg fixes (misc). 2019-02-05 15:12:20 +00:00
audio_format.h [clang-tidy] Apply performance-move-const-arg fixes (misc). 2019-02-05 15:12:20 +00:00
BUILD.gn Remove rtc_base/scoped_ref_ptr.h. 2019-01-25 20:29:58 +00:00
builtin_audio_decoder_factory.cc Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
builtin_audio_decoder_factory.h Remove rtc_base/scoped_ref_ptr.h. 2019-01-25 20:29:58 +00:00
builtin_audio_encoder_factory.cc Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
builtin_audio_encoder_factory.h Remove rtc_base/scoped_ref_ptr.h. 2019-01-25 20:29:58 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00