| Index: webrtc/video/payload_router.cc
|
| diff --git a/webrtc/video/payload_router.cc b/webrtc/video/payload_router.cc
|
| index abe476f276b8a83532446dab87be9f27a6ae7ceb..3be5882cdbf2172928fe312caf2a9b3401d1ea9b 100644
|
| --- a/webrtc/video/payload_router.cc
|
| +++ b/webrtc/video/payload_router.cc
|
| @@ -84,6 +84,7 @@ void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader* rtp) {
|
| return;
|
| }
|
| }
|
| +
|
| } // namespace
|
|
|
| PayloadRouter::PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
|
| @@ -115,10 +116,12 @@ bool PayloadRouter::active() {
|
| return active_ && !rtp_modules_.empty();
|
| }
|
|
|
| -void PayloadRouter::SetSendingRtpModules(size_t num_sending_modules) {
|
| - RTC_DCHECK_LE(num_sending_modules, rtp_modules_.size());
|
| +void PayloadRouter::SetSendStreams(const std::vector<VideoStream>& streams) {
|
| + RTC_DCHECK_LE(streams.size(), rtp_modules_.size());
|
| rtc::CritScope lock(&crit_);
|
| - num_sending_modules_ = num_sending_modules;
|
| + num_sending_modules_ = streams.size();
|
| + streams_ = streams;
|
| + // TODO(perkj): Should SetSendStreams also call SetTargetSendBitrate?
|
| UpdateModuleSendingState();
|
| }
|
|
|
| @@ -163,12 +166,22 @@ int32_t PayloadRouter::Encoded(const EncodedImage& encoded_image,
|
| encoded_image._length, fragmentation, &rtp_video_header);
|
| }
|
|
|
| -void PayloadRouter::SetTargetSendBitrates(
|
| - const std::vector<uint32_t>& stream_bitrates) {
|
| +void PayloadRouter::SetTargetSendBitrate(uint32_t bitrate_bps) {
|
| rtc::CritScope lock(&crit_);
|
| - RTC_DCHECK_LE(stream_bitrates.size(), rtp_modules_.size());
|
| - for (size_t i = 0; i < stream_bitrates.size(); ++i) {
|
| - rtp_modules_[i]->SetTargetSendBitrate(stream_bitrates[i]);
|
| + RTC_DCHECK_LE(streams_.size(), rtp_modules_.size());
|
| +
|
| + // TODO(sprang): Rebase https://codereview.webrtc.org/1913073002/ on top of
|
| + // this.
|
| + int bitrate_remainder = bitrate_bps;
|
| + for (size_t i = 0; i < streams_.size() && bitrate_remainder > 0; ++i) {
|
| + int stream_bitrate = 0;
|
| + if (streams_[i].max_bitrate_bps > bitrate_remainder) {
|
| + stream_bitrate = bitrate_remainder;
|
| + } else {
|
| + stream_bitrate = streams_[i].max_bitrate_bps;
|
| + }
|
| + bitrate_remainder -= stream_bitrate;
|
| + rtp_modules_[i]->SetTargetSendBitrate(stream_bitrate);
|
| }
|
| }
|
|
|
|
|