219 lines
7.8 KiB
C
219 lines
7.8 KiB
C
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/*
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* Copyright (c) 2010 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 WEBRTC_MEDIA_BASE_RTPDUMP_H_
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#define WEBRTC_MEDIA_BASE_RTPDUMP_H_
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#include <string.h>
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#include <string>
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#include <vector>
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#include "webrtc/base/basictypes.h"
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#include "webrtc/base/bytebuffer.h"
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/stream.h"
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namespace cricket {
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// We use the RTP dump file format compatible to the format used by rtptools
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// (http://www.cs.columbia.edu/irt/software/rtptools/) and Wireshark
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// (http://wiki.wireshark.org/rtpdump). In particular, the file starts with the
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// first line "#!rtpplay1.0 address/port\n", followed by a 16 byte file header.
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// For each packet, the file contains a 8 byte dump packet header, followed by
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// the actual RTP or RTCP packet.
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enum RtpDumpPacketFilter {
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PF_NONE = 0x0,
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PF_RTPHEADER = 0x1,
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PF_RTPPACKET = 0x3, // includes header
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// PF_RTCPHEADER = 0x4, // TODO(juberti)
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PF_RTCPPACKET = 0xC, // includes header
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PF_ALL = 0xF
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};
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struct RtpDumpFileHeader {
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RtpDumpFileHeader(int64_t start_ms, uint32_t s, uint16_t p);
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void WriteToByteBuffer(rtc::ByteBufferWriter* buf);
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static const char kFirstLine[];
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static const size_t kHeaderLength = 16;
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uint32_t start_sec; // start of recording, the seconds part.
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uint32_t start_usec; // start of recording, the microseconds part.
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uint32_t source; // network source (multicast address).
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uint16_t port; // UDP port.
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uint16_t padding; // 2 bytes padding.
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};
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struct RtpDumpPacket {
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RtpDumpPacket() {}
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RtpDumpPacket(const void* d, size_t s, uint32_t elapsed, bool rtcp)
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: elapsed_time(elapsed), original_data_len((rtcp) ? 0 : s) {
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data.resize(s);
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memcpy(&data[0], d, s);
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}
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// In the rtpdump file format, RTCP packets have their data len set to zero,
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// since RTCP has an internal length field.
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bool is_rtcp() const { return original_data_len == 0; }
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bool IsValidRtpPacket() const;
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bool IsValidRtcpPacket() const;
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// Get the payload type, sequence number, timestampe, and SSRC of the RTP
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// packet. Return true and set the output parameter if successful.
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bool GetRtpPayloadType(int* pt) const;
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bool GetRtpSeqNum(int* seq_num) const;
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bool GetRtpTimestamp(uint32_t* ts) const;
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bool GetRtpSsrc(uint32_t* ssrc) const;
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bool GetRtpHeaderLen(size_t* len) const;
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// Get the type of the RTCP packet. Return true and set the output parameter
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// if successful.
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bool GetRtcpType(int* type) const;
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static const size_t kHeaderLength = 8;
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uint32_t elapsed_time; // Milliseconds since the start of recording.
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std::vector<uint8_t> data; // The actual RTP or RTCP packet.
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size_t original_data_len; // The original length of the packet; may be
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// greater than data.size() if only part of the
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// packet was recorded.
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};
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class RtpDumpReader {
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public:
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explicit RtpDumpReader(rtc::StreamInterface* stream)
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: stream_(stream),
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file_header_read_(false),
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first_line_and_file_header_len_(0),
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start_time_ms_(0),
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ssrc_override_(0) {
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}
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virtual ~RtpDumpReader() {}
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// Use the specified ssrc, rather than the ssrc from dump, for RTP packets.
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void SetSsrc(uint32_t ssrc);
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virtual rtc::StreamResult ReadPacket(RtpDumpPacket* packet);
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protected:
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rtc::StreamResult ReadFileHeader();
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bool RewindToFirstDumpPacket() {
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return stream_->SetPosition(first_line_and_file_header_len_);
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}
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private:
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// Check if its matches "#!rtpplay1.0 address/port\n".
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bool CheckFirstLine(const std::string& first_line);
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rtc::StreamInterface* stream_;
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bool file_header_read_;
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size_t first_line_and_file_header_len_;
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int64_t start_time_ms_;
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uint32_t ssrc_override_;
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RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpReader);
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};
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// RtpDumpLoopReader reads RTP dump packets from the input stream and rewinds
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// the stream when it ends. RtpDumpLoopReader maintains the elapsed time, the
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// RTP sequence number and the RTP timestamp properly. RtpDumpLoopReader can
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// handle both RTP dump and RTCP dump. We assume that the dump does not mix
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// RTP packets and RTCP packets.
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class RtpDumpLoopReader : public RtpDumpReader {
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public:
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explicit RtpDumpLoopReader(rtc::StreamInterface* stream);
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virtual rtc::StreamResult ReadPacket(RtpDumpPacket* packet);
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private:
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// During the first loop, update the statistics, including packet count, frame
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// count, timestamps, and sequence number, of the input stream.
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void UpdateStreamStatistics(const RtpDumpPacket& packet);
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// At the end of first loop, calculate elapsed_time_increases_,
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// rtp_seq_num_increase_, and rtp_timestamp_increase_.
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void CalculateIncreases();
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// During the second and later loops, update the elapsed time of the dump
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// packet. If the dumped packet is a RTP packet, update its RTP sequence
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// number and timestamp as well.
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void UpdateDumpPacket(RtpDumpPacket* packet);
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int loop_count_;
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// How much to increase the elapsed time, RTP sequence number, RTP timestampe
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// for each loop. They are calcualted with the variables below during the
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// first loop.
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uint32_t elapsed_time_increases_;
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int rtp_seq_num_increase_;
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uint32_t rtp_timestamp_increase_;
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// How many RTP packets and how many payload frames in the input stream. RTP
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// packets belong to the same frame have the same RTP timestamp, different
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// dump timestamp, and different RTP sequence number.
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uint32_t packet_count_;
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uint32_t frame_count_;
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// The elapsed time, RTP sequence number, and RTP timestamp of the first and
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// the previous dump packets in the input stream.
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uint32_t first_elapsed_time_;
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int first_rtp_seq_num_;
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int64_t first_rtp_timestamp_;
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uint32_t prev_elapsed_time_;
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int prev_rtp_seq_num_;
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int64_t prev_rtp_timestamp_;
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RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpLoopReader);
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};
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class RtpDumpWriter {
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public:
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explicit RtpDumpWriter(rtc::StreamInterface* stream);
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// Filter to control what packets we actually record.
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void set_packet_filter(int filter);
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// Write a RTP or RTCP packet. The parameters data points to the packet and
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// data_len is its length.
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rtc::StreamResult WriteRtpPacket(const void* data, size_t data_len) {
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return WritePacket(data, data_len, GetElapsedTime(), false);
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}
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rtc::StreamResult WriteRtcpPacket(const void* data, size_t data_len) {
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return WritePacket(data, data_len, GetElapsedTime(), true);
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}
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rtc::StreamResult WritePacket(const RtpDumpPacket& packet) {
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return WritePacket(&packet.data[0], packet.data.size(), packet.elapsed_time,
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packet.is_rtcp());
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}
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uint32_t GetElapsedTime() const;
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bool GetDumpSize(size_t* size) {
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// Note that we use GetPosition(), rather than GetSize(), to avoid flush the
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// stream per write.
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return stream_ && size && stream_->GetPosition(size);
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}
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protected:
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rtc::StreamResult WriteFileHeader();
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private:
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rtc::StreamResult WritePacket(const void* data,
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size_t data_len,
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uint32_t elapsed,
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bool rtcp);
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size_t FilterPacket(const void* data, size_t data_len, bool rtcp);
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rtc::StreamResult WriteToStream(const void* data, size_t data_len);
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rtc::StreamInterface* stream_;
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int packet_filter_;
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bool file_header_written_;
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int64_t start_time_ms_; // Time when the record starts.
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// If writing to the stream takes longer than this many ms, log a warning.
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int64_t warn_slow_writes_delay_;
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RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpWriter);
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};
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} // namespace cricket
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#endif // WEBRTC_MEDIA_BASE_RTPDUMP_H_
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