Files
platform-external-webrtc/modules/audio_coding/neteq/tools
Tommi 25eb47ccf1 Make the RtpHeaderParserImpl available to tests and tools only.
There are a few reasons for making this test only:
* The code is only used by tests and utilities.
* The pure interface has only a single implementation so an interface isn't really needed.
  (a followup change could remove it altogether)
* The implementation always incorporates locking regardless of how the class gets used.
  See e.g. previous use in the Packet class.
* The implementation is a layer on top of RtpUtility::RtpHeaderParser which is
  sufficient for most production cases.

Change-Id: Ide6d50567cf8ae5127a2eb04cceeb10cf317ec36
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150658
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29010}
2019-08-29 15:56:40 +00:00
..
2019-07-08 13:45:15 +00:00
2019-07-08 13:45:15 +00:00
2019-07-08 13:45:15 +00:00

NetEQ RTP Play tool

Testing of the command line arguments

The command line tool neteq_rtpplay can be tested by running neteq_rtpplay_test.sh, which is not use on try bots, but it can be used before submitting any CLs that may break the behavior of the command line arguments of neteq_rtpplay.

Run neteq_rtpplay_test.sh as follows from the src/ folder:

src$ ./modules/audio_coding/neteq/tools/neteq_rtpplay_test.sh  \
  out/Default/neteq_rtpplay  \
  resources/audio_coding/neteq_opus.rtp  \
  resources/short_mixed_mono_48.pcm

You can replace the RTP and PCM files with any other compatible files. If you get an error using the files indicated above, try running gclient sync.

Requirements: awk and md5sum.