mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 05:40:42 +01:00
Move video video receiver transformable frame to modules/rtc_rtcp/source
Step 1 of combining the sender and receiver types Also moved the RtpFrameObject to rtp_rtcp/source, as it's heavily used by the transformable receiver frame, I couldn't work out a better way of managing the dependencies, and everything else seemed to work fine. Bug: chromium:1412687 Change-Id: I55e816a0d7aa2962560ff9ebaf30ad63ab0b9810 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291710 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tony Herre <herre@google.com> Cr-Commit-Position: refs/heads/main@{#39255}
This commit is contained in:
parent
b311f6aba8
commit
be9b576188
28 changed files with 62 additions and 57 deletions
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@ -19,6 +19,7 @@
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#include "absl/container/inlined_vector.h"
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#include "absl/types/optional.h"
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#include "modules/rtp_rtcp/source/frame_object.h"
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#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
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#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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@ -28,7 +29,6 @@
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#include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h"
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#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h"
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#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h"
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#include "modules/video_coding/frame_object.h"
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#include "modules/video_coding/packet_buffer.h"
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#include "modules/video_coding/rtp_frame_reference_finder.h"
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#include "rtc_base/logging.h"
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@ -177,6 +177,8 @@ rtc_library("rtp_rtcp") {
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"source/forward_error_correction.h",
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"source/forward_error_correction_internal.cc",
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"source/forward_error_correction_internal.h",
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"source/frame_object.cc",
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"source/frame_object.h",
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"source/packet_loss_stats.cc",
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"source/packet_loss_stats.h",
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"source/packet_sequencer.cc",
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@ -224,6 +226,8 @@ rtc_library("rtp_rtcp") {
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"source/rtp_sender_video_frame_transformer_delegate.h",
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"source/rtp_sequence_number_map.cc",
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"source/rtp_sequence_number_map.h",
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"source/rtp_video_stream_receiver_frame_transformer_delegate.cc",
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"source/rtp_video_stream_receiver_frame_transformer_delegate.h",
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"source/source_tracker.cc",
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"source/source_tracker.h",
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"source/time_util.cc",
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@ -298,6 +302,7 @@ rtc_library("rtp_rtcp") {
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"../../api/video:video_rtp_headers",
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"../../api/video_codecs:video_codecs_api",
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"../../call:rtp_interfaces",
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"../../call:video_stream_api",
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"../../common_video",
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"../../logging:rtc_event_audio",
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"../../logging:rtc_event_rtp_rtcp",
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@ -322,6 +327,7 @@ rtc_library("rtp_rtcp") {
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"../../rtc_base:rtc_numerics",
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"../../rtc_base:safe_conversions",
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"../../rtc_base:safe_minmax",
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"../../rtc_base:threading",
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"../../rtc_base:timeutils",
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"../../rtc_base/containers:flat_map",
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"../../rtc_base/experiments:field_trial_parser",
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@ -603,6 +609,7 @@ if (rtc_include_tests) {
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"source/rtp_sequence_number_map_unittest.cc",
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"source/rtp_util_unittest.cc",
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"source/rtp_video_layers_allocation_extension_unittest.cc",
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"source/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc",
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"source/source_tracker_unittest.cc",
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"source/time_util_unittest.cc",
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"source/ulpfec_generator_unittest.cc",
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@ -655,6 +662,7 @@ if (rtc_include_tests) {
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"../../api/video:video_rtp_headers",
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"../../api/video_codecs:video_codecs_api",
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"../../call:rtp_receiver",
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"../../call:video_stream_api",
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"../../common_video",
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"../../common_video/generic_frame_descriptor",
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"../../common_video/test:utilities",
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@ -8,7 +8,7 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/video_coding/frame_object.h"
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#include "modules/rtp_rtcp/source/frame_object.h"
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#include <string.h>
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@ -93,8 +93,7 @@ RtpFrameObject::RtpFrameObject(
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is_last_spatial_layer = markerBit;
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}
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RtpFrameObject::~RtpFrameObject() {
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}
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RtpFrameObject::~RtpFrameObject() {}
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uint16_t RtpFrameObject::first_seq_num() const {
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return first_seq_num_;
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@ -8,8 +8,8 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_VIDEO_CODING_FRAME_OBJECT_H_
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#define MODULES_VIDEO_CODING_FRAME_OBJECT_H_
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#ifndef MODULES_RTP_RTCP_SOURCE_FRAME_OBJECT_H_
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#define MODULES_RTP_RTCP_SOURCE_FRAME_OBJECT_H_
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#include <vector>
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@ -70,4 +70,4 @@ class RtpFrameObject : public EncodedFrame {
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} // namespace webrtc
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#endif // MODULES_VIDEO_CODING_FRAME_OBJECT_H_
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#endif // MODULES_RTP_RTCP_SOURCE_FRAME_OBJECT_H_
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@ -8,7 +8,7 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "video/rtp_video_stream_receiver_frame_transformer_delegate.h"
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#include "modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.h"
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#include <utility>
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#include <vector>
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@ -8,14 +8,14 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef VIDEO_RTP_VIDEO_STREAM_RECEIVER_FRAME_TRANSFORMER_DELEGATE_H_
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#define VIDEO_RTP_VIDEO_STREAM_RECEIVER_FRAME_TRANSFORMER_DELEGATE_H_
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#ifndef MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_STREAM_RECEIVER_FRAME_TRANSFORMER_DELEGATE_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_STREAM_RECEIVER_FRAME_TRANSFORMER_DELEGATE_H_
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#include <memory>
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#include "api/frame_transformer_interface.h"
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#include "api/sequence_checker.h"
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#include "modules/video_coding/frame_object.h"
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#include "modules/rtp_rtcp/source/frame_object.h"
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#include "rtc_base/system/no_unique_address.h"
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#include "rtc_base/thread.h"
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} // namespace webrtc
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#endif // VIDEO_RTP_VIDEO_STREAM_RECEIVER_FRAME_TRANSFORMER_DELEGATE_H_
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_STREAM_RECEIVER_FRAME_TRANSFORMER_DELEGATE_H_
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@ -8,7 +8,7 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "video/rtp_video_stream_receiver_frame_transformer_delegate.h"
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#include "modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.h"
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#include <cstdio>
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#include <memory>
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// Check that the transformable frame passed to the frame transformer has the
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// correct metadata.
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EXPECT_CALL(*mock_frame_transformer, Transform)
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.WillOnce(
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[&](std::unique_ptr<TransformableFrameInterface>
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transformable_frame) {
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auto frame =
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absl::WrapUnique(static_cast<TransformableVideoFrameInterface*>(
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transformable_frame.release()));
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ASSERT_TRUE(frame);
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auto metadata = frame->GetMetadata();
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EXPECT_EQ(metadata.GetWidth(), 1280u);
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EXPECT_EQ(metadata.GetHeight(), 720u);
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EXPECT_EQ(metadata.GetFrameId(), 10);
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EXPECT_EQ(metadata.GetTemporalIndex(), 3);
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EXPECT_EQ(metadata.GetSpatialIndex(), 2);
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EXPECT_THAT(metadata.GetFrameDependencies(), ElementsAre(5));
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EXPECT_THAT(metadata.GetDecodeTargetIndications(),
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ElementsAre(DecodeTargetIndication::kSwitch));
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EXPECT_EQ(metadata.GetCsrcs(), csrcs);
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});
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.WillOnce([&](std::unique_ptr<TransformableFrameInterface>
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transformable_frame) {
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auto frame =
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absl::WrapUnique(static_cast<TransformableVideoFrameInterface*>(
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transformable_frame.release()));
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ASSERT_TRUE(frame);
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auto metadata = frame->GetMetadata();
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EXPECT_EQ(metadata.GetWidth(), 1280u);
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EXPECT_EQ(metadata.GetHeight(), 720u);
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EXPECT_EQ(metadata.GetFrameId(), 10);
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EXPECT_EQ(metadata.GetTemporalIndex(), 3);
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EXPECT_EQ(metadata.GetSpatialIndex(), 2);
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EXPECT_THAT(metadata.GetFrameDependencies(), ElementsAre(5));
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EXPECT_THAT(metadata.GetDecodeTargetIndications(),
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ElementsAre(DecodeTargetIndication::kSwitch));
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EXPECT_EQ(metadata.GetCsrcs(), csrcs);
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});
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// The delegate creates a transformable frame from the RtpFrameObject.
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delegate->TransformFrame(CreateRtpFrameObject(video_header, csrcs));
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}
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@ -207,8 +207,6 @@ rtc_library("video_coding") {
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"fec_controller_default.cc",
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"fec_controller_default.h",
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"fec_rate_table.h",
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"frame_object.cc",
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"frame_object.h",
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"generic_decoder.cc",
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"generic_decoder.h",
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"h264_sprop_parameter_sets.cc",
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"../../test:video_test_support",
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"../../test/time_controller:time_controller",
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"../../third_party/libyuv:libyuv",
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"../rtp_rtcp",
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"../rtp_rtcp:rtp_rtcp_format",
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"../rtp_rtcp:rtp_video_header",
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"codecs/av1:video_coding_codecs_av1_tests",
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#include "api/task_queue/task_queue_base.h"
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#include "api/units/time_delta.h"
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#include "api/units/timestamp.h"
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#include "modules/video_coding/frame_object.h"
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#include "modules/rtp_rtcp/source/frame_object.h"
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#include "modules/video_coding/timing/jitter_estimator.h"
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#include "modules/video_coding/timing/timing.h"
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#include "rtc_base/numerics/sequence_number_util.h"
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#include "api/array_view.h"
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#include "common_video/h264/h264_common.h"
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#include "modules/video_coding/frame_object.h"
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#include "modules/rtp_rtcp/source/frame_object.h"
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#include "rtc_base/numerics/sequence_number_unwrapper.h"
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#include "rtc_base/random.h"
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#include "test/field_trial.h"
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#include <memory>
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#include "absl/container/inlined_vector.h"
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#include "modules/video_coding/frame_object.h"
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#include "modules/rtp_rtcp/source/frame_object.h"
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#include "modules/video_coding/rtp_frame_reference_finder.h"
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#include "rtc_base/numerics/sequence_number_unwrapper.h"
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#include <utility>
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#include "absl/types/variant.h"
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#include "modules/video_coding/frame_object.h"
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#include "modules/rtp_rtcp/source/frame_object.h"
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#include "modules/video_coding/rtp_frame_id_only_ref_finder.h"
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#include "modules/video_coding/rtp_generic_ref_finder.h"
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#include "modules/video_coding/rtp_seq_num_only_ref_finder.h"
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#include <memory>
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#include "modules/video_coding/frame_object.h"
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#include "modules/rtp_rtcp/source/frame_object.h"
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namespace webrtc {
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namespace internal {
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/video_coding/rtp_frame_reference_finder.h"
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#include <cstring>
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#include <limits>
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#include <map>
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#include <utility>
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#include <vector>
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#include "modules/video_coding/frame_object.h"
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#include "modules/rtp_rtcp/source/frame_object.h"
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#include "modules/video_coding/packet_buffer.h"
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#include "modules/video_coding/rtp_frame_reference_finder.h"
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#include "rtc_base/random.h"
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#include "rtc_base/ref_count.h"
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#include "system_wrappers/include/clock.h"
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#include <memory>
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#include "modules/video_coding/frame_object.h"
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#include "modules/rtp_rtcp/source/frame_object.h"
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#include "modules/video_coding/rtp_frame_reference_finder.h"
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namespace webrtc {
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#include <utility>
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#include "absl/container/inlined_vector.h"
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#include "modules/video_coding/frame_object.h"
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#include "modules/rtp_rtcp/source/frame_object.h"
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#include "modules/video_coding/rtp_frame_reference_finder.h"
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#include "rtc_base/numerics/sequence_number_unwrapper.h"
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#include <set>
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#include "absl/container/inlined_vector.h"
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#include "modules/video_coding/frame_object.h"
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#include "modules/rtp_rtcp/source/frame_object.h"
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#include "modules/video_coding/rtp_frame_reference_finder.h"
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#include "rtc_base/numerics/sequence_number_unwrapper.h"
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#include <utility>
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#include <vector>
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#include "modules/video_coding/frame_object.h"
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#include "modules/rtp_rtcp/source/frame_object.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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#include <set>
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#include "absl/container/inlined_vector.h"
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#include "modules/video_coding/frame_object.h"
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#include "modules/rtp_rtcp/source/frame_object.h"
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#include "modules/video_coding/rtp_frame_reference_finder.h"
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#include "rtc_base/numerics/sequence_number_unwrapper.h"
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/video_coding/rtp_vp9_ref_finder.h"
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#include <utility>
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#include <vector>
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#include "modules/video_coding/frame_object.h"
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#include "modules/video_coding/rtp_vp9_ref_finder.h"
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#include "modules/rtp_rtcp/source/frame_object.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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@ -581,7 +581,8 @@ webrtc_fuzzer_test("rtp_frame_reference_finder_fuzzer") {
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deps = [
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"../../api:rtp_packet_info",
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"../../api:scoped_refptr",
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"../../modules/video_coding/",
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"../../modules/rtp_rtcp",
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"../../modules/video_coding:video_coding",
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"../../system_wrappers",
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]
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}
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#include <memory>
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#include "api/rtp_packet_infos.h"
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#include "modules/video_coding/frame_object.h"
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#include "modules/rtp_rtcp/source/frame_object.h"
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#include "modules/video_coding/rtp_frame_reference_finder.h"
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namespace webrtc {
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@ -53,8 +53,6 @@ rtc_library("video") {
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"rtp_streams_synchronizer2.h",
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"rtp_video_stream_receiver2.cc",
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"rtp_video_stream_receiver2.h",
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"rtp_video_stream_receiver_frame_transformer_delegate.cc",
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"rtp_video_stream_receiver_frame_transformer_delegate.h",
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"send_delay_stats.cc",
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"send_delay_stats.h",
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"send_statistics_proxy.cc",
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@ -791,7 +789,6 @@ if (rtc_include_tests) {
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"receive_statistics_proxy2_unittest.cc",
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"report_block_stats_unittest.cc",
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"rtp_video_stream_receiver2_unittest.cc",
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"rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc",
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"send_delay_stats_unittest.cc",
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"send_statistics_proxy_unittest.cc",
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"stats_counter_unittest.cc",
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#include <utility>
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#include <vector>
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#include "modules/rtp_rtcp/source/frame_object.h"
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#include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
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#include "modules/video_coding/frame_object.h"
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#include "rtc_base/logging.h"
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#include "system_wrappers/include/field_trial.h"
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#include "api/crypto/crypto_options.h"
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#include "api/crypto/frame_decryptor_interface.h"
|
||||
#include "api/field_trials_view.h"
|
||||
#include "modules/video_coding/frame_object.h"
|
||||
#include "modules/rtp_rtcp/source/frame_object.h"
|
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|
||||
namespace webrtc {
|
||||
|
||||
|
|
|
@ -26,6 +26,7 @@
|
|||
#include "modules/rtp_rtcp/include/receive_statistics.h"
|
||||
#include "modules/rtp_rtcp/include/rtp_cvo.h"
|
||||
#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
|
||||
#include "modules/rtp_rtcp/source/frame_object.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_format.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
|
||||
|
@ -36,7 +37,6 @@
|
|||
#include "modules/rtp_rtcp/source/ulpfec_receiver.h"
|
||||
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
|
||||
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h"
|
||||
#include "modules/video_coding/frame_object.h"
|
||||
#include "modules/video_coding/h264_sprop_parameter_sets.h"
|
||||
#include "modules/video_coding/h264_sps_pps_tracker.h"
|
||||
#include "modules/video_coding/nack_requester.h"
|
||||
|
|
|
@ -36,6 +36,7 @@
|
|||
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_video_header.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.h"
|
||||
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
|
||||
#include "modules/video_coding/h264_sps_pps_tracker.h"
|
||||
#include "modules/video_coding/loss_notification_controller.h"
|
||||
|
@ -47,7 +48,6 @@
|
|||
#include "rtc_base/system/no_unique_address.h"
|
||||
#include "rtc_base/thread_annotations.h"
|
||||
#include "video/buffered_frame_decryptor.h"
|
||||
#include "video/rtp_video_stream_receiver_frame_transformer_delegate.h"
|
||||
#include "video/unique_timestamp_counter.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
|
|
@ -19,6 +19,7 @@
|
|||
#include "call/test/mock_rtp_packet_sink_interface.h"
|
||||
#include "common_video/h264/h264_common.h"
|
||||
#include "media/base/media_constants.h"
|
||||
#include "modules/rtp_rtcp/source/frame_object.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_format.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_format_vp9.h"
|
||||
|
@ -27,7 +28,6 @@
|
|||
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
|
||||
#include "modules/video_coding/frame_object.h"
|
||||
#include "modules/video_coding/include/video_coding_defines.h"
|
||||
#include "modules/video_coding/rtp_frame_reference_finder.h"
|
||||
#include "rtc_base/byte_buffer.h"
|
||||
|
|
Loading…
Reference in a new issue