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Diffstat (limited to 'third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.h')
-rw-r--r-- | third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.h | 61 |
1 files changed, 61 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.h new file mode 100644 index 0000000000..19abd3feb2 --- /dev/null +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.h @@ -0,0 +1,61 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ +#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ + +#include <stdint.h> + +#include <memory> +#include <vector> + +#include "absl/types/optional.h" +#include "api/array_view.h" +#include "modules/rtp_rtcp/source/rtp_video_header.h" + +namespace webrtc { + +class RtpPacketToSend; + +class RtpPacketizer { + public: + struct PayloadSizeLimits { + int max_payload_len = 1200; + int first_packet_reduction_len = 0; + int last_packet_reduction_len = 0; + // Reduction len for packet that is first & last at the same time. + int single_packet_reduction_len = 0; + }; + + // If type is not set, returns a raw packetizer. + static std::unique_ptr<RtpPacketizer> Create( + absl::optional<VideoCodecType> type, + rtc::ArrayView<const uint8_t> payload, + PayloadSizeLimits limits, + // Codec-specific details. + const RTPVideoHeader& rtp_video_header); + + virtual ~RtpPacketizer() = default; + + // Returns number of remaining packets to produce by the packetizer. + virtual size_t NumPackets() const = 0; + + // Get the next payload with payload header. + // Write payload and set marker bit of the `packet`. + // Returns true on success, false otherwise. + virtual bool NextPacket(RtpPacketToSend* packet) = 0; + + // Split payload_len into sum of integers with respect to `limits`. + // Returns empty vector on failure. + static std::vector<int> SplitAboutEqually(int payload_len, + const PayloadSizeLimits& limits); +}; +} // namespace webrtc +#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ |