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10542f21c8
Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
69 lines
2.1 KiB
C++
69 lines
2.1 KiB
C++
/*
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* Copyright (c) 2014 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_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
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#include <stdio.h>
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#include <memory>
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#include <string>
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#include "absl/types/optional.h"
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#include "modules/audio_coding/neteq/tools/packet_source.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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class RtpHeaderParser;
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namespace test {
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class RtpFileReader;
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class RtpFileSource : public PacketSource {
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public:
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// Creates an RtpFileSource reading from |file_name|. If the file cannot be
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// opened, or has the wrong format, NULL will be returned.
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static RtpFileSource* Create(
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const std::string& file_name,
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absl::optional<uint32_t> ssrc_filter = absl::nullopt);
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// Checks whether a files is a valid RTP dump or PCAP (Wireshark) file.
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static bool ValidRtpDump(const std::string& file_name);
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static bool ValidPcap(const std::string& file_name);
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~RtpFileSource() override;
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// Registers an RTP header extension and binds it to |id|.
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virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
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std::unique_ptr<Packet> NextPacket() override;
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private:
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static const int kFirstLineLength = 40;
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static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2;
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static const size_t kPacketHeaderSize = 8;
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explicit RtpFileSource(absl::optional<uint32_t> ssrc_filter);
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bool OpenFile(const std::string& file_name);
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std::unique_ptr<RtpFileReader> rtp_reader_;
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std::unique_ptr<RtpHeaderParser> parser_;
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const absl::optional<uint32_t> ssrc_filter_;
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RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
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