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webrtc_m130/webrtc/modules/audio_coding/acm2
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minyue 2e03c66119 Adding build switch for Opus that supports 120ms ptime. 
BUG=webrtc:7097

TEST=Set "ptime=120", try WebRTC calls over custom build Chromium with and without Opus 120ms. Try both Chromium w <-> Chromium w and Chromium w <-> Chromium w/o

Review-Url: https://codereview.webrtc.org/2668633004
Cr-Commit-Position: refs/heads/master@{#16408}
2017-02-02 01:31:11 +00:00
..
acm_codec_database.cc
Adding build switch for Opus that supports 120ms ptime.
2017-02-02 01:31:11 +00:00
acm_codec_database.h
Delete voice_engine_configurations.h
2016-12-12 13:03:08 +00:00
acm_common_defs.h
Delete voice_engine_configurations.h
2016-12-12 13:03:08 +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
Delete voice_engine_configurations.h
2016-12-12 13:03:08 +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
Pass SdpAudioFormat through Channel, without converting to CodecInst
2017-01-19 15:03:59 +00:00
audio_coding_module.cc
Update smoothed bitrate.
2017-01-12 18:17:38 +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
Delete voice_engine_configurations.h
2016-12-12 13:03:08 +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
Pass SdpAudioFormat through Channel, without converting to CodecInst
2017-01-19 15:03:59 +00:00
rent_a_codec.h
Support receiving DTMF for multiple RTP clock rates.
2016-11-17 12:45:25 +00:00