Index: webrtc/modules/rtp_rtcp/source/rtp_sender.h |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
index a7fab0f860b12e6c0f845ed4a5147cd247c4b140..f068ae35701f7f07b49612169d0459626afe5ef5 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
@@ -76,12 +76,12 @@ class RTPSenderInterface { |
virtual size_t MaxDataPayloadLength() const = 0; |
virtual uint16_t ActualSendBitrateKbit() const = 0; |
- virtual int32_t SendToNetwork(uint8_t* data_buffer, |
- size_t payload_length, |
- size_t rtp_header_length, |
- int64_t capture_time_ms, |
- StorageType storage, |
- RtpPacketSender::Priority priority) = 0; |
+ virtual bool SendToNetwork(uint8_t* data_buffer, |
+ size_t payload_length, |
+ size_t rtp_header_length, |
+ int64_t capture_time_ms, |
+ StorageType storage, |
+ RtpPacketSender::Priority priority) = 0; |
virtual bool UpdateVideoRotation(uint8_t* rtp_packet, |
size_t rtp_packet_length, |
@@ -154,14 +154,15 @@ class RTPSender : public RTPSenderInterface { |
void SetMaxPayloadLength(size_t max_payload_length); |
- int32_t SendOutgoingData(FrameType frame_type, |
- int8_t payload_type, |
- uint32_t timestamp, |
- int64_t capture_time_ms, |
- const uint8_t* payload_data, |
- size_t payload_size, |
- const RTPFragmentationHeader* fragmentation, |
- const RTPVideoHeader* rtp_header); |
+ bool SendOutgoingData(FrameType frame_type, |
+ int8_t payload_type, |
+ uint32_t timestamp, |
+ int64_t capture_time_ms, |
+ const uint8_t* payload_data, |
+ size_t payload_size, |
+ const RTPFragmentationHeader* fragmentation, |
+ const RTPVideoHeader* rtp_header, |
+ uint32_t* transport_frame_id_out); |
// RTP header extension |
int32_t SetTransmissionTimeOffset(int32_t transmission_time_offset); |
@@ -276,12 +277,12 @@ class RTPSender : public RTPSenderInterface { |
uint32_t Timestamp() const override; |
uint32_t SSRC() const override; |
- int32_t SendToNetwork(uint8_t* data_buffer, |
- size_t payload_length, |
- size_t rtp_header_length, |
- int64_t capture_time_ms, |
- StorageType storage, |
- RtpPacketSender::Priority priority) override; |
+ bool SendToNetwork(uint8_t* data_buffer, |
+ size_t payload_length, |
+ size_t rtp_header_length, |
+ int64_t capture_time_ms, |
+ StorageType storage, |
+ RtpPacketSender::Priority priority) override; |
// Audio. |