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..ed5f1aa7e18db6b28708f97606bd2e993e60e5af 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" |
@@ -32,10 +31,13 @@ |
#include "webrtc/typedefs.h" |
namespace webrtc { |
+class RtpPacketizer; |
class RtpPacketToSend; |
class RTPSenderVideo { |
public: |
+ static constexpr int64_t kTLRateWindowSizeMs = 2500; |
+ |
RTPSenderVideo(Clock* clock, |
RTPSender* rtpSender, |
FlexfecSender* flexfec_sender); |
@@ -55,7 +57,8 @@ 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 +79,22 @@ class RTPSenderVideo { |
int SelectiveRetransmissions() const; |
void SetSelectiveRetransmissions(uint8_t settings); |
+ protected: |
+ uint8_t GetTemporalId(const RTPVideoHeader& header) const; |
danilchap
2017/09/04 09:03:05
look like this one can be static or free function
sprang_webrtc
2017/09/04 11:09:28
Made it static. Call from unit test.
|
+ StorageType GetStorageType(uint8_t temporal_id, |
+ int32_t retransmission_settings, |
+ RtpPacketizer* packetizer, |
+ int64_t expected_retransmission_time); |
+ |
private: |
+ struct TemporalLayerStats { |
+ TemporalLayerStats() |
+ : frame_rate_fp1000s(kTLRateWindowSizeMs, 1000 * 1000), |
brandtr
2017/09/04 10:29:22
Nice unit :)
sprang_webrtc
2017/09/04 11:09:28
Hehe, thanks :)
Maybe I'll add a comment explicitl
|
+ last_frame_time(0) {} |
+ RateStatistics frame_rate_fp1000s; |
+ int64_t last_frame_time; |
danilchap
2017/09/04 09:03:05
add _ms
sprang_webrtc
2017/09/04 11:09:28
Done.
|
+ }; |
+ |
size_t CalculateFecPacketOverhead() const EXCLUSIVE_LOCKS_REQUIRED(crit_); |
void SendVideoPacket(std::unique_ptr<RtpPacketToSend> packet, |
@@ -103,6 +121,10 @@ class RTPSenderVideo { |
bool flexfec_enabled() const { return flexfec_sender_ != nullptr; } |
+ bool UpdateConditionalRetransmit(uint8_t temporal_id, |
+ int64_t expected_retransmission_time_ms) |
+ EXCLUSIVE_LOCKS_REQUIRED(stats_crit_); |
+ |
RTPSender* const rtp_sender_; |
Clock* const clock_; |
@@ -131,6 +153,10 @@ 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_; |
}; |