webrtc/modules/rtp_rtcp/source/h264/pps_parser_unittest.cc - Issue 1979443004: Add H264 bitstream rewriting to limit frame reordering marker in header

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Unified Diff: webrtc/modules/rtp_rtcp/source/h264/pps_parser_unittest.cc

Issue 1979443004: Add H264 bitstream rewriting to limit frame reordering marker in header (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Rewriting on the receiver side as well Created 4 years, 7 months ago

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« webrtc/modules/rtp_rtcp/source/h264/pps_parser.h ('K') | « webrtc/modules/rtp_rtcp/source/h264/pps_parser.cc ('k') | webrtc/modules/rtp_rtcp/source/h264/sps_parser.h » ('j') | webrtc/modules/rtp_rtcp/source/h264/sps_parser.h » ('J')
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Index: webrtc/modules/rtp_rtcp/source/h264/pps_parser_unittest.cc

diff --git a/webrtc/modules/audio_coding/neteq/packet.cc b/webrtc/modules/rtp_rtcp/source/h264/pps_parser_unittest.cc

similarity index 72%

copy from webrtc/modules/audio_coding/neteq/packet.cc

copy to webrtc/modules/rtp_rtcp/source/h264/pps_parser_unittest.cc

index 8a19fe4d5923d5e22a5b485600c6da0c5f3856ba..1844fb5a3711044aa14ff7b620f472236040a803 100644

--- a/webrtc/modules/audio_coding/neteq/packet.cc

+++ b/webrtc/modules/rtp_rtcp/source/h264/pps_parser_unittest.cc

@@ -8,12 +8,6 @@

* be found in the AUTHORS file in the root of the source tree.

-#include "webrtc/modules/audio_coding/neteq/packet.h"

+#include "webrtc/modules/rtp_rtcp/source/h264/pps_parser.h"

-namespace webrtc {

-Packet::Packet() = default;

-Packet::~Packet() = default;

-} // namespace webrtc

+namespace webrtc {} // namespace webrtc