We have a few namespace fixes to start with. webrtc::XWindowProperty<uint32_t> window_state in media/webrtc/trunk/webrtc/modules/desktop_capture/x11/window_list_utils.cc ::testing::StrictMock<MockTransportFeedbackSender> in media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc I'll attach other one liners as comments to this bug.

RTC_LOG(LS_ERROR) << deviceUniqueIdUTF8 << " Invalid deviceCapabilityNumber " in media/webrtc/trunk/webrtc/modules/video_capture/device_info_impl.cc

if (conversionResult != 0) { in media/webrtc/trunk/webrtc/modules/video_capture/video_capture_impl.cc.

num_cores_(0), in media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc

class H264SpropParameterSetsTest : public ::testing::Test {, in media/webrtc/trunk/webrtc/modules/video_coding/h264_sprop_parameter_sets_unittest.cc

timeval tv = {static_cast<time_t>(milliseconds) / 1000, static_cast<suseconds_t>((milliseconds % 1000) * 1000)}; in media/webrtc/trunk/webrtc/rtc_base/task_queue_libevent.cc.

x_display_->ProcessPendingXEvents(); in media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_x11.cc.

const char* p = ::strchr(Base64Table, ch); in media/webrtc/trunk/webrtc/rtc_base/base64.cc

enum SeiType : uint8_t { kSeiRecPt = 6, // Recovery Point SEI Payload in media/webrtc/trunk/webrtc/common_video/h264/h264_common.h

// add safety margin because STAP-A packets can cause it to expand by // ~two bytes per NAL uint32_t requiredSizeBytes = Length() + packet.sizeBytes + (packet.insertStartCode ? kH264StartCodeLengthBytes : 0) + kBufferSafetyMargin + in media/webrtc/trunk/webrtc/modules/video_coding/frame_buffer.cc

memset(_jumpBuf, 0, sizeof(_jumpBuf)); memset(_driftBuf, 0, sizeof(_driftBuf)); in media/webrtc/trunk/webrtc/modules/video_coding/rtt_filter.cc

::testing::StrictMock<MockRecoveredPacketReceiver> recovered_packet_receiver_; in media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc

next_time_to_send_rtcp_(clock->TimeInMilliseconds()), in media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc But only after we determine why we made this change and if it is still necessary.

(In reply to Dan Minor [:dminor] from comment #3) > num_cores_(0), in > media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc This member is gone upstream.

(In reply to Dan Minor [:dminor] from comment #6) > x_display_->ProcessPendingXEvents(); > > in media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_x11.cc. I think I have tracked this change to https://hg.mozilla.org/mozilla-central/rev/ffb95db2dc7e . This check was also added upstream, but after the IsWindowValid check. This may be sufficient.

(In reply to Dan Minor [:dminor] from comment #8) > enum SeiType : uint8_t { > kSeiRecPt = 6, // Recovery Point SEI Payload > > in media/webrtc/trunk/webrtc/common_video/h264/h264_common.h I think this need to be a larger patch that includes the changes here: https://hg.mozilla.org/mozilla-central/annotate/6c2bac3a1afc/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc#l74

(In reply to Dan Minor [:dminor] from comment #10) > memset(_jumpBuf, 0, sizeof(_jumpBuf)); > memset(_driftBuf, 0, sizeof(_driftBuf)); > > in media/webrtc/trunk/webrtc/modules/video_coding/rtt_filter.cc This has been fixed upstream. It is also fixed in 64.

The include change in https://phabricator.services.mozilla.com/D7465 can be upstreamed.

The namespacing change in https://phabricator.services.mozilla.com/D7470 can be upstreamed.