henrik.lundin@webrtc.org fcbe36a1d9 Add const qualifier to WebRtcPcm16b_Encode ...

BUG=909
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7831 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-12-08 18:26:49 +00:00

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audio_checksum.h

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audio_loop.cc

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audio_loop.h

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audio_sink.h

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constant_pcm_packet_source.cc

New ACM test to trigger audio glitch when switching output sample rate

2014-10-14 10:49:58 +00:00

constant_pcm_packet_source.h

New ACM test to trigger audio glitch when switching output sample rate

2014-10-14 10:49:58 +00:00

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_performance_test.cc

Add const qualifier to WebRtcPcm16b_Encode

2014-12-08 18:26:49 +00:00

neteq_performance_test.h

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neteq_quality_test.cc

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neteq_quality_test.h

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

neteq_rtpplay.cc

Adding a duration printout to neteq_rtpplay

2014-12-03 13:28:53 +00:00

output_audio_file.h

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output_wav_file.h

Add wav output capability to neteq_rtpplay

2014-11-24 14:50:53 +00:00

packet_source.h

Use RtpFileSource in NetEqDecodingTest

2014-11-17 09:08:38 +00:00

packet_unittest.cc

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packet.cc

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packet.h

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resample_input_audio_file.cc

Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile

2014-10-27 12:58:18 +00:00

resample_input_audio_file.h

Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile

2014-10-27 12:58:18 +00:00

rtp_analyze.cc

Let RtpFileSource use RtpFileReader

2014-10-02 08:19:38 +00:00

rtp_file_source.cc

Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader

2014-11-26 15:50:30 +00:00

rtp_file_source.h

Let RtpFileSource use RtpFileReader

2014-10-02 08:19:38 +00:00

rtp_generator.cc

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rtp_generator.h

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rtpcat.cc

Move and rename neteq/test/RTPcat to neteq/tools/rtpcat

2014-12-01 14:23:01 +00:00