| 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) {
|
|
|