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Unified Diff: webrtc/video/payload_router.h

Issue 1903193002: Revert of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 8 months ago
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Index: webrtc/video/payload_router.h
diff --git a/webrtc/video/payload_router.h b/webrtc/video/payload_router.h
index c2f4b0442cb226bb77667330614e1a1251492fd1..81ec0dd4b1cb1342f44c1e6bc4588efea8dee056 100644
--- a/webrtc/video/payload_router.h
+++ b/webrtc/video/payload_router.h
@@ -17,7 +17,6 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
-#include "webrtc/video_encoder.h"
#include "webrtc/system_wrappers/include/atomic32.h"
namespace webrtc {
@@ -28,11 +27,10 @@
// PayloadRouter routes outgoing data to the correct sending RTP module, based
// on the simulcast layer in RTPVideoHeader.
-class PayloadRouter : public EncodedImageCallback {
+class PayloadRouter {
public:
// Rtp modules are assumed to be sorted in simulcast index order.
- explicit PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
- int payload_type);
+ explicit PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules);
~PayloadRouter();
static size_t DefaultMaxPayloadLength();
@@ -43,11 +41,16 @@
void set_active(bool active);
bool active();
- // Implements EncodedImageCallback.
- // Returns 0 if the packet was routed / sent, -1 otherwise.
- int32_t Encoded(const EncodedImage& encoded_image,
- const CodecSpecificInfo* codec_specific_info,
- const RTPFragmentationHeader* fragmentation) override;
+ // Input parameters according to the signature of RtpRtcp::SendOutgoingData.
+ // Returns true if the packet was routed / sent, false otherwise.
+ bool RoutePayload(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_hdr);
// Configures current target bitrate per module. 'stream_bitrates' is assumed
// to be in the same order as 'SetSendingRtpModules'.
@@ -66,7 +69,6 @@
// Rtp modules are assumed to be sorted in simulcast index order. Not owned.
const std::vector<RtpRtcp*> rtp_modules_;
- const int payload_type_;
RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
};
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