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This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
95 lines
3.3 KiB
C++
95 lines
3.3 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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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_RTP_RTCP_SOURCE_RTP_GENERIC_FRAME_DESCRIPTOR_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_GENERIC_FRAME_DESCRIPTOR_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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namespace webrtc {
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class RtpGenericFrameDescriptorExtension;
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// Data to put on the wire for FrameDescriptor rtp header extension.
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class RtpGenericFrameDescriptor {
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public:
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static constexpr int kMaxNumFrameDependencies = 8;
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static constexpr int kMaxTemporalLayers = 8;
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static constexpr int kMaxSpatialLayers = 8;
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RtpGenericFrameDescriptor();
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RtpGenericFrameDescriptor(const RtpGenericFrameDescriptor&);
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~RtpGenericFrameDescriptor();
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bool FirstPacketInSubFrame() const { return beginning_of_subframe_; }
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void SetFirstPacketInSubFrame(bool first) { beginning_of_subframe_ = first; }
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bool LastPacketInSubFrame() const { return end_of_subframe_; }
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void SetLastPacketInSubFrame(bool last) { end_of_subframe_ = last; }
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// Denotes whether the frame is discardable. That is, whether skipping it
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// would have no effect on the decodability of subsequent frames.
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// An absl::optional is used because version 0 of the extension did not
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// support this flag. (The optional aspect is relevant only when parsing.)
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// TODO(bugs.webrtc.org/10243): Make this into a plain bool when v00 of
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// the extension is deprecated.
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absl::optional<bool> Discardable() const { return discardable_; }
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void SetDiscardable(bool discardable) { discardable_ = discardable; }
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// Properties below undefined if !FirstPacketInSubFrame()
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// Valid range for temporal layer: [0, 7]
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int TemporalLayer() const;
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void SetTemporalLayer(int temporal_layer);
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// Frame might by used, possible indirectly, for spatial layer sid iff
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// (bitmask & (1 << sid)) != 0
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int SpatialLayer() const;
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uint8_t SpatialLayersBitmask() const;
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void SetSpatialLayersBitmask(uint8_t spatial_layers);
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int Width() const { return width_; }
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int Height() const { return height_; }
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void SetResolution(int width, int height);
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uint16_t FrameId() const;
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void SetFrameId(uint16_t frame_id);
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rtc::ArrayView<const uint16_t> FrameDependenciesDiffs() const;
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void ClearFrameDependencies() { num_frame_deps_ = 0; }
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// Returns false on failure, i.e. number of dependencies is too large.
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bool AddFrameDependencyDiff(uint16_t fdiff);
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void SetByteRepresentation(rtc::ArrayView<const uint8_t> representation);
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rtc::ArrayView<const uint8_t> GetByteRepresentation();
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private:
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bool beginning_of_subframe_ = false;
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bool end_of_subframe_ = false;
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absl::optional<bool> discardable_;
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uint16_t frame_id_ = 0;
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uint8_t spatial_layers_ = 1;
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uint8_t temporal_layer_ = 0;
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size_t num_frame_deps_ = 0;
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uint16_t frame_deps_id_diffs_[kMaxNumFrameDependencies];
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int width_ = 0;
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int height_ = 0;
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std::vector<uint8_t> byte_representation_;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_GENERIC_FRAME_DESCRIPTOR_H_
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