| 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.
|
|
|
|
|