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) { |