Chromium Code Reviews| Index: webrtc/modules/rtp_rtcp/source/rtp_sender_video.h |
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h |
| index 9fb4648b82e2138e2bfdb0e75e2d87a816710b37..a38e787e63855cb5c8113a7284ccd701f2f1cbaa 100644 |
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h |
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h |
| @@ -11,9 +11,8 @@ |
| #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_ |
| #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_ |
| -#include <list> |
| +#include <map> |
| #include <memory> |
| -#include <vector> |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/rtp_rtcp/include/flexfec_sender.h" |
| @@ -32,6 +31,7 @@ |
| #include "webrtc/typedefs.h" |
| namespace webrtc { |
| +class RtpPacketizer; |
| class RtpPacketToSend; |
| class RTPSenderVideo { |
| @@ -55,7 +55,8 @@ class RTPSenderVideo { |
| const uint8_t* payload_data, |
| size_t payload_size, |
| const RTPFragmentationHeader* fragmentation, |
| - const RTPVideoHeader* video_header); |
| + const RTPVideoHeader* video_header, |
| + int64_t expected_retransmission_time_ms); |
| void SetVideoCodecType(RtpVideoCodecTypes type); |
| @@ -76,7 +77,21 @@ class RTPSenderVideo { |
| int SelectiveRetransmissions() const; |
| void SetSelectiveRetransmissions(uint8_t settings); |
| + protected: |
| + StorageType GetStorageType(const RTPVideoHeader& header, |
| + int32_t retransmission_settings, |
| + RtpPacketizer* packetizer, |
| + int64_t expected_retransmission_time); |
| + |
| private: |
| + struct TemporalLayerStats { |
| + explicit TemporalLayerStats(int64_t window_size_ms) |
| + : frame_rate_fp1000s_(window_size_ms, 1000 * 1000), |
| + last_frame_time_(0) {} |
| + RateStatistics frame_rate_fp1000s_; |
| + int64_t last_frame_time_; |
|
danilchap
2017/08/29 17:31:20
struct members shouldn't have '_' in the end.
sprang_webrtc
2017/08/31 15:54:29
Done.
|
| + }; |
| + |
| size_t CalculateFecPacketOverhead() const EXCLUSIVE_LOCKS_REQUIRED(crit_); |
| void SendVideoPacket(std::unique_ptr<RtpPacketToSend> packet, |
| @@ -103,6 +118,9 @@ class RTPSenderVideo { |
| bool flexfec_enabled() const { return flexfec_sender_ != nullptr; } |
| + TemporalLayerStats* GetTemporalLayerStats(int temporal_id) |
| + EXCLUSIVE_LOCKS_REQUIRED(stats_crit_); |
| + |
| RTPSender* const rtp_sender_; |
| Clock* const clock_; |
| @@ -131,6 +149,10 @@ class RTPSenderVideo { |
| RateStatistics fec_bitrate_ GUARDED_BY(stats_crit_); |
| // Bitrate used for video payload and RTP headers. |
| RateStatistics video_bitrate_ GUARDED_BY(stats_crit_); |
| + |
| + std::map<int, TemporalLayerStats> frame_stats_by_temporal_layer_ |
| + GUARDED_BY(stats_crit_); |
| + |
| OneTimeEvent first_frame_sent_; |
| }; |