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26bddb92f0f76c08bd0dbf5efe4909128ef0a90b
platform-external-webrtc/webrtc/modules/audio_coding/acm2
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ehmaldonado 26bddb92f0 Replace test_support_main by test_main and get rid of test_support_main_threaded_mac 
test_support_main_threaded_mac doesn't seem to be used. It looks like it was
last used about a year and a half ago, and was removed in
https://webrtc-codereview.appspot.com/55379004

BUG=webrtc:6424
R=kjellander@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2540693002
Cr-Commit-Position: refs/heads/master@{#15332}
2016-11-30 14:12:10 +00:00
..
acm_codec_database.cc
Support receiving DTMF for multiple RTP clock rates.
2016-11-17 12:45:25 +00:00
acm_codec_database.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
acm_common_defs.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
acm_receive_test.cc
Support receiving DTMF for multiple RTP clock rates.
2016-11-17 12:45:25 +00:00
acm_receive_test.h
Stop using old AudioCodingModule::RegisterReceiveCodec overloads
2016-10-24 20:47:16 +00:00
acm_receiver_unittest.cc
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac
2016-11-30 14:12:10 +00:00
acm_receiver.cc
Support receiving DTMF for multiple RTP clock rates.
2016-11-17 12:45:25 +00:00
acm_receiver.h
Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.
2016-10-12 18:04:16 +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.cc
Drop _oldapi from ACM test file names
2016-10-06 08:31:38 +00:00
acm_send_test.h
Drop _oldapi from ACM test file names
2016-10-06 08:31:38 +00:00
audio_coding_module_unittest.cc
Stop using old AudioCodingModule::RegisterReceiveCodec overloads
2016-10-24 20:47:16 +00:00
audio_coding_module.cc
Revert of Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. (patchset #5 id:240001 of https://codereview.webrtc.org/2411613002/ )
2016-11-30 09:19:06 +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
RTC_[D]CHECK_op: Remove "u" suffix on integer constants
2016-11-28 23:21:51 +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
Support receiving DTMF for multiple RTP clock rates.
2016-11-17 12:45:25 +00:00
rent_a_codec.h
Support receiving DTMF for multiple RTP clock rates.
2016-11-17 12:45:25 +00:00