1497614 - Upstream small webrtc.org fixes

Status: NEW

(Core :: WebRTC: Audio/Video, enhancement, P3)

Description

[Dan Minor [:dminor]]

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.

Comment 1

[Dan Minor [:dminor]]

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

Comment 2

[Dan Minor [:dminor]]

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

Comment 3

[Dan Minor [:dminor]]

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

Comment 4

[Dan Minor [:dminor]]

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

Comment 5

[Dan Minor [:dminor]]

    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.

Comment 6

[Dan Minor [:dminor]]

  x_display_->ProcessPendingXEvents();

in media/webrtc/trunk/webrtc/modules/desktop_capture/window_capturer_x11.cc.

Comment 7

[Dan Minor [:dminor]]

  const char* p = ::strchr(Base64Table, ch);

in media/webrtc/trunk/webrtc/rtc_base/base64.cc

Comment 8

[Dan Minor [:dminor]]

enum SeiType : uint8_t {
  kSeiRecPt = 6, // Recovery Point SEI Payload

in media/webrtc/trunk/webrtc/common_video/h264/h264_common.h

Comment 9

[Dan Minor [:dminor]]

 // 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

Comment 10

[Dan Minor [:dminor]]

  memset(_jumpBuf, 0, sizeof(_jumpBuf));
  memset(_driftBuf, 0, sizeof(_driftBuf));

in media/webrtc/trunk/webrtc/modules/video_coding/rtt_filter.cc

Comment 11

[Dan Minor [:dminor]]

::testing::StrictMock<MockRecoveredPacketReceiver> recovered_packet_receiver_;

in media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc

Comment 12

[Dan Minor [:dminor]]

      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.

Comment 13

[Nico Grunbaum [:ng, @chew:mozilla.org]]

(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.

Comment 14

[Nico Grunbaum [:ng, @chew:mozilla.org]]

(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.

Comment 15

[Nico Grunbaum [:ng, @chew:mozilla.org]]

(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

Comment 16

[Nico Grunbaum [:ng, @chew:mozilla.org]]

(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.

Comment 17

[Dan Minor [:dminor]]

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

Comment 18

[Dan Minor [:dminor]]

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