Chromium Code Reviews| Index: webrtc/video/payload_router.h | 
| diff --git a/webrtc/video/payload_router.h b/webrtc/video/payload_router.h | 
| index c2f4b0442cb226bb77667330614e1a1251492fd1..c3f2386051684b4ea9a9fbb03e01e8dbef248d2b 100644 | 
| --- a/webrtc/video/payload_router.h | 
| +++ b/webrtc/video/payload_router.h | 
| @@ -36,7 +36,7 @@ class PayloadRouter : public EncodedImageCallback { | 
| ~PayloadRouter(); | 
| static size_t DefaultMaxPayloadLength(); | 
| - void SetSendingRtpModules(size_t num_sending_modules); | 
| + void SetSendCodec(const VideoCodec& codec); | 
| 
 
pbos-webrtc
2016/04/22 15:24:20
Can this take a std::vector<VideoStream> instead?
 
perkj_webrtc
2016/04/27 08:00:57
ok- but I had to add the streams to the EncoderCon
 
 | 
| // PayloadRouter will only route packets if being active, all packets will be | 
| // dropped otherwise. | 
| @@ -49,9 +49,8 @@ class PayloadRouter : public EncodedImageCallback { | 
| const CodecSpecificInfo* codec_specific_info, | 
| const RTPFragmentationHeader* fragmentation) override; | 
| - // Configures current target bitrate per module. 'stream_bitrates' is assumed | 
| - // to be in the same order as 'SetSendingRtpModules'. | 
| - void SetTargetSendBitrates(const std::vector<uint32_t>& stream_bitrates); | 
| + // Configures current target bitrate. | 
| + void SetTargetSendBitrate(uint32_t bitrate_bps); | 
| // Returns the maximum allowed data payload length, given the configured MTU | 
| // and RTP headers. | 
| @@ -62,6 +61,7 @@ class PayloadRouter : public EncodedImageCallback { | 
| rtc::CriticalSection crit_; | 
| bool active_ GUARDED_BY(crit_); | 
| + VideoCodec codec_ GUARDED_BY(crit_); | 
| size_t num_sending_modules_ GUARDED_BY(crit_); | 
| // Rtp modules are assumed to be sorted in simulcast index order. Not owned. |