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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender.h

Issue 2089773002: Add EncodedImageCallback::OnEncodedImage(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: . Created 4 years, 4 months ago
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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.
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