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Add EncodedImageCallback::OnEncodedImage().

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OnEncodedImage() is going to replace Encoded(), which is deprecated now.
The new OnEncodedImage() returns Result struct that contains frame_id,
which tells the encoder RTP timestamp for the frame.

BUG=chromium:621691
R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2089773002 .

Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
Cr-Original-Commit-Position: refs/heads/master@{#13613}
Cr-Commit-Position: refs/heads/master@{#13615}
This commit is contained in:
Sergey Ulanov
2016-08-02 15:14:39 -07:00
parent ac4dc2cefe
commit 4c7f4cd2ef
43 changed files with 475 additions and 380 deletions
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18
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
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@ -112,15 +112,15 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
// Used by the codec module to deliver a video or audio frame for
// packetization.
int32_t SendOutgoingData(
FrameType frame_type,
int8_t payload_type,
uint32_t time_stamp,
int64_t capture_time_ms,
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation = NULL,
const RTPVideoHeader* rtp_video_header = NULL) override;
bool SendOutgoingData(FrameType frame_type,
int8_t payload_type,
uint32_t time_stamp,
int64_t capture_time_ms,
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation,
const RTPVideoHeader* rtp_video_header,
uint32_t* transport_frame_id_out) override;
bool TimeToSendPacket(uint32_t ssrc,
uint16_t sequence_number,