pkasting@chromium.org 16825b1a82 Use int64_t more consistently for times, in particular for RTT values. ...

Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t.  Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d

2015-01-12 21:51:21 +00:00

..

codec_database_test.cc

Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798

2014-08-11 21:06:30 +00:00

codec_database_test.h

Include files from webrtc/.. paths in video_coding/.

2013-07-16 12:32:05 +00:00

generic_codec_test.cc

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

generic_codec_test.h

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

media_opt_test.cc

Use size_t more consistently for packet/payload lengths.

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

media_opt_test.h

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

mt_rx_tx_test.cc

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

mt_test_common.cc

Use size_t more consistently for packet/payload lengths.

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

mt_test_common.h

Use size_t more consistently for packet/payload lengths.

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

normal_test.cc

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

normal_test.h

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

plotJitterEstimate.m

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

plotReceiveTrace.m

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

plotTimingTest.m

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

quality_modes_test.cc

Use size_t more consistently for packet/payload lengths.

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

quality_modes_test.h

Use size_t more consistently for packet/payload lengths.

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

receiver_tests.h

Use size_t more consistently for packet/payload lengths.

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

release_test.h

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

rtp_player.cc

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

rtp_player.h

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

subfigure.m

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

test_callbacks.cc

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

test_callbacks.h

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

test_macros.h

Switch C++-style C headers with their C equivalents.

2013-08-05 16:22:53 +00:00

test_util.cc

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +00:00

test_util.h

Use size_t more consistently for packet/payload lengths.

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

tester_main.cc

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +00:00

vcm_payload_sink_factory.cc

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

vcm_payload_sink_factory.h

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

video_rtp_play_mt.cc

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

video_rtp_play.cc

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

video_source.cc

Use size_t more consistently for packet/payload lengths.

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

video_source.h

Use size_t more consistently for packet/payload lengths.

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