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webrtc_m130/webrtc/modules/audio_coding/neteq/tools
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terelius d5c1a0bd5d New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs. 
Changed rtpdump converter and neteq tool to use new parser, but still aborting if the file is corrupt.

Review-Url: https://codereview.webrtc.org/1768773002
Cr-Commit-Position: refs/heads/master@{#12714}
2016-05-13 07:43:04 +00:00
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audio_checksum.h
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audio_loop.cc
Reset indexer upon initialization in AudioLoop.
2016-02-24 11:49:23 +00:00
audio_loop.h
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audio_sink.h
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constant_pcm_packet_source.cc
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constant_pcm_packet_source.h
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input_audio_file_unittest.cc
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input_audio_file.cc
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input_audio_file.h
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neteq_external_decoder_test.cc
NetEq: Implement muted output
2016-05-12 20:51:37 +00:00
neteq_external_decoder_test.h
Remove the type parameter to NetEq::GetAudio
2016-03-08 10:38:02 +00:00
neteq_performance_test.cc
NetEq: Implement muted output
2016-05-12 20:51:37 +00:00
neteq_performance_test.h
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neteq_quality_test.cc
NetEq: Implement muted output
2016-05-12 20:51:37 +00:00
neteq_quality_test.h
Change NetEq::GetAudio to use AudioFrame
2016-03-04 18:34:26 +00:00
neteq_rtpplay.cc
NetEq: Implement muted output
2016-05-12 20:51:37 +00:00
output_audio_file.h
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output_wav_file.h
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packet_source.h
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packet_unittest.cc
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packet.cc
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packet.h
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resample_input_audio_file.cc
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resample_input_audio_file.h
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rtc_event_log_source.cc
New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs.
2016-05-13 07:43:04 +00:00
rtc_event_log_source.h
New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs.
2016-05-13 07:43:04 +00:00
rtp_analyze.cc
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rtp_file_source.cc
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rtp_file_source.h
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rtp_generator.cc
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rtp_generator.h
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rtpcat.cc
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