| Index: webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
|
| index e896c65f97269db765d112ce6965b903f4d378bd..15a1d8b1520f094a60611b7469bdf4b2da33fdfe 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
|
| @@ -81,7 +81,11 @@ bool ParseStapAStartOffsets(const uint8_t* nalu_ptr,
|
| RtpPacketizerH264::RtpPacketizerH264(size_t max_payload_len,
|
| H264PacketizationMode packetization_mode)
|
| : max_payload_len_(max_payload_len),
|
| - packetization_mode_(packetization_mode) {}
|
| + packetization_mode_(packetization_mode) {
|
| + // Guard against uninitialized memory in packetization_mode.
|
| + RTC_CHECK(packetization_mode == H264PacketizationMode::NonInterleaved ||
|
| + packetization_mode == H264PacketizationMode::SingleNalUnit);
|
| +}
|
|
|
| RtpPacketizerH264::~RtpPacketizerH264() {
|
| }
|
|
|