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2779bab02a3c9e576cdb9927ef86d3fc40772bdc
platform-external-webrtc/webrtc/modules/audio_coding/acm2
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solenberg 2779bab02a Support receiving DTMF for multiple RTP clock rates. 
BUG=webrtc:2795

Review-Url: https://codereview.webrtc.org/2337473002
Cr-Commit-Position: refs/heads/master@{#15128}
2016-11-17 12:45:25 +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
Drop _oldapi from ACM test file names
2016-10-06 08:31:38 +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
Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.
2016-10-12 18:04:16 +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 old video defines in engine config.
2016-10-07 05:07:36 +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