webrtc_m130

History

Henrik Lundin b0b54259c3 Let rtp_analyze parse absolute sender time

Also change to use virtual_packet_length_bytes in order to print the
actual packet size of the complete packet even when the RTP file only
contains RTP headers.

BUG=2692
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51559004

Cr-Commit-Position: refs/heads/master@{#9025}

2015-04-17 09:46:56 +00:00

codecs

AudioEncoderDecoderIsac: Merge the two config structs

2015-04-16 04:00:18 +00:00

main

AudioEncoderDecoderIsac: Merge the two config structs

2015-04-16 04:00:18 +00:00

neteq

Let rtp_analyze parse absolute sender time

2015-04-17 09:46:56 +00:00

audio_codec_speed_tests.isolate

Re-land "Remove <(webrtc_root) from source file entries."