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Unified Diff: webrtc/video/video_send_stream.cc

Issue 1507643004: Lint clean video/ and add lint presubmit check. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: ostringstream Created 5 years ago
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Index: webrtc/video/video_send_stream.cc
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index 08b0f1902a3406ea892335a051a35d6fa4afdb22..d73ef9cbd36f216ec43637b0b1aa3bbe9b1c13ad 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -400,8 +400,8 @@ bool VideoSendStream::ReconfigureVideoEncoder(
RTC_DCHECK_GE(streams[i].max_bitrate_bps, streams[i].target_bitrate_bps);
RTC_DCHECK_GE(streams[i].max_qp, 0);
- sim_stream->width = static_cast<unsigned short>(streams[i].width);
- sim_stream->height = static_cast<unsigned short>(streams[i].height);
+ sim_stream->width = static_cast<uint16_t>(streams[i].width);
+ sim_stream->height = static_cast<uint16_t>(streams[i].height);
sim_stream->minBitrate = streams[i].min_bitrate_bps / 1000;
sim_stream->targetBitrate = streams[i].target_bitrate_bps / 1000;
sim_stream->maxBitrate = streams[i].max_bitrate_bps / 1000;
@@ -410,12 +410,12 @@ bool VideoSendStream::ReconfigureVideoEncoder(
streams[i].temporal_layer_thresholds_bps.size() + 1);
video_codec.width = std::max(video_codec.width,
- static_cast<unsigned short>(streams[i].width));
+ static_cast<uint16_t>(streams[i].width));
video_codec.height = std::max(
- video_codec.height, static_cast<unsigned short>(streams[i].height));
+ video_codec.height, static_cast<uint16_t>(streams[i].height));
video_codec.minBitrate =
- std::min(video_codec.minBitrate,
- static_cast<unsigned int>(streams[i].min_bitrate_bps / 1000));
+ std::min(static_cast<uint16_t>(video_codec.minBitrate),
+ static_cast<uint16_t>(streams[i].min_bitrate_bps / 1000));
video_codec.maxBitrate += streams[i].max_bitrate_bps / 1000;
video_codec.qpMax = std::max(video_codec.qpMax,
static_cast<unsigned int>(streams[i].max_qp));
@@ -500,7 +500,7 @@ std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const {
std::map<uint32_t, RtpState> rtp_states;
for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
uint32_t ssrc = config_.rtp.ssrcs[i];
- rtp_states[ssrc] = vie_channel_->GetRtpStateForSsrc( ssrc);
+ rtp_states[ssrc] = vie_channel_->GetRtpStateForSsrc(ssrc);
}
for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
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