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webrtc_m130/webrtc/modules/audio_coding/acm2
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kwiberg 5377bc77cc Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere 
The former is always defined (by webrtc/base/checks.h) to either 0 or
1, whereas the latter isn't necessarily defined.

This is a re-land of https://codereview.webrtc.org/2384693002, which
broke Chromium. We re-land without changing this CL at all, because
the thing that needed fixing was in Chromium:
https://codereview.chromium.org/2384263004.

NOTRY=true
TBR=ossu@webrtc.org
BUG=webrtc:6451

Review-Url: https://codereview.webrtc.org/2389943003
Cr-Commit-Position: refs/heads/master@{#14508}
2016-10-04 20:47:02 +00:00
..
acm_codec_database.cc
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
acm_codec_database.h
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
acm_common_defs.h
audio_coding: remove "main" directory
2015-11-26 12:45:01 +00:00
acm_receive_test_oldapi.cc
AudioCodingModule: Specify decoders using SdpAudioFormat
2016-10-04 16:33:33 +00:00
acm_receive_test_oldapi.h
AudioCodingModule: Specify decoders using SdpAudioFormat
2016-10-04 16:33:33 +00:00
acm_receiver_unittest_oldapi.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
acm_receiver.cc
AudioCodingModule: Specify decoders using SdpAudioFormat
2016-10-04 16:33:33 +00:00
acm_receiver.h
AudioCodingModule: Specify decoders using SdpAudioFormat
2016-10-04 16:33:33 +00:00
acm_resampler.cc
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
acm_resampler.h
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
acm_send_test_oldapi.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
acm_send_test_oldapi.h
Fix Chromium clang plugin warnings
2016-08-29 17:05:34 +00:00
audio_coding_module_unittest_oldapi.cc
Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere
2016-10-04 20:47:02 +00:00
audio_coding_module.cc
AudioCodingModule: Specify decoders using SdpAudioFormat
2016-10-04 16:33:33 +00:00
call_statistics_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
call_statistics.cc
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
call_statistics.h
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
codec_manager_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
codec_manager.cc
Don't recreate the speech encoder if we don't have to
2016-06-23 10:58:45 +00:00
codec_manager.h
Don't recreate the speech encoder if we don't have to
2016-06-23 10:58:45 +00:00
rent_a_codec_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
rent_a_codec.cc
Stopped using the NetEqDecoder enum internally in NetEq.
2016-09-23 09:19:49 +00:00
rent_a_codec.h
AudioDecoderIsacT: Require caller to always specify sample rate
2016-06-02 09:59:02 +00:00