blob: 8ea7800e9c74114db37971dcbe01dcb61947578d [file] [log] [blame]
* Copyright (c) 2010 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.
#include <string.h>
#include <string>
#include <vector>
#include "webrtc/base/basictypes.h"
#include "webrtc/base/bytebuffer.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/stream.h"
namespace cricket {
// We use the RTP dump file format compatible to the format used by rtptools
// ( and Wireshark
// ( In particular, the file starts with the
// first line "#!rtpplay1.0 address/port\n", followed by a 16 byte file header.
// For each packet, the file contains a 8 byte dump packet header, followed by
// the actual RTP or RTCP packet.
enum RtpDumpPacketFilter {
PF_NONE = 0x0,
PF_RTPPACKET = 0x3, // includes header
// PF_RTCPHEADER = 0x4, // TODO(juberti)
PF_RTCPPACKET = 0xC, // includes header
PF_ALL = 0xF
struct RtpDumpFileHeader {
RtpDumpFileHeader(int64_t start_ms, uint32_t s, uint16_t p);
void WriteToByteBuffer(rtc::ByteBufferWriter* buf);
static const char kFirstLine[];
static const size_t kHeaderLength = 16;
uint32_t start_sec; // start of recording, the seconds part.
uint32_t start_usec; // start of recording, the microseconds part.
uint32_t source; // network source (multicast address).
uint16_t port; // UDP port.
uint16_t padding; // 2 bytes padding.
struct RtpDumpPacket {
RtpDumpPacket() {}
RtpDumpPacket(const void* d, size_t s, uint32_t elapsed, bool rtcp)
: elapsed_time(elapsed), original_data_len((rtcp) ? 0 : s) {
memcpy(&data[0], d, s);
// In the rtpdump file format, RTCP packets have their data len set to zero,
// since RTCP has an internal length field.
bool is_rtcp() const { return original_data_len == 0; }
bool IsValidRtpPacket() const;
bool IsValidRtcpPacket() const;
// Get the payload type, sequence number, timestampe, and SSRC of the RTP
// packet. Return true and set the output parameter if successful.
bool GetRtpPayloadType(int* pt) const;
bool GetRtpSeqNum(int* seq_num) const;
bool GetRtpTimestamp(uint32_t* ts) const;
bool GetRtpSsrc(uint32_t* ssrc) const;
bool GetRtpHeaderLen(size_t* len) const;
// Get the type of the RTCP packet. Return true and set the output parameter
// if successful.
bool GetRtcpType(int* type) const;
static const size_t kHeaderLength = 8;
uint32_t elapsed_time; // Milliseconds since the start of recording.
std::vector<uint8_t> data; // The actual RTP or RTCP packet.
size_t original_data_len; // The original length of the packet; may be
// greater than data.size() if only part of the
// packet was recorded.
class RtpDumpReader {
explicit RtpDumpReader(rtc::StreamInterface* stream)
: stream_(stream),
ssrc_override_(0) {
virtual ~RtpDumpReader() {}
// Use the specified ssrc, rather than the ssrc from dump, for RTP packets.
void SetSsrc(uint32_t ssrc);
virtual rtc::StreamResult ReadPacket(RtpDumpPacket* packet);
rtc::StreamResult ReadFileHeader();
bool RewindToFirstDumpPacket() {
return stream_->SetPosition(first_line_and_file_header_len_);
// Check if its matches "#!rtpplay1.0 address/port\n".
bool CheckFirstLine(const std::string& first_line);
rtc::StreamInterface* stream_;
bool file_header_read_;
size_t first_line_and_file_header_len_;
int64_t start_time_ms_;
uint32_t ssrc_override_;
// RtpDumpLoopReader reads RTP dump packets from the input stream and rewinds
// the stream when it ends. RtpDumpLoopReader maintains the elapsed time, the
// RTP sequence number and the RTP timestamp properly. RtpDumpLoopReader can
// handle both RTP dump and RTCP dump. We assume that the dump does not mix
// RTP packets and RTCP packets.
class RtpDumpLoopReader : public RtpDumpReader {
explicit RtpDumpLoopReader(rtc::StreamInterface* stream);
virtual rtc::StreamResult ReadPacket(RtpDumpPacket* packet);
// During the first loop, update the statistics, including packet count, frame
// count, timestamps, and sequence number, of the input stream.
void UpdateStreamStatistics(const RtpDumpPacket& packet);
// At the end of first loop, calculate elapsed_time_increases_,
// rtp_seq_num_increase_, and rtp_timestamp_increase_.
void CalculateIncreases();
// During the second and later loops, update the elapsed time of the dump
// packet. If the dumped packet is a RTP packet, update its RTP sequence
// number and timestamp as well.
void UpdateDumpPacket(RtpDumpPacket* packet);
int loop_count_;
// How much to increase the elapsed time, RTP sequence number, RTP timestampe
// for each loop. They are calcualted with the variables below during the
// first loop.
uint32_t elapsed_time_increases_;
int rtp_seq_num_increase_;
uint32_t rtp_timestamp_increase_;
// How many RTP packets and how many payload frames in the input stream. RTP
// packets belong to the same frame have the same RTP timestamp, different
// dump timestamp, and different RTP sequence number.
uint32_t packet_count_;
uint32_t frame_count_;
// The elapsed time, RTP sequence number, and RTP timestamp of the first and
// the previous dump packets in the input stream.
uint32_t first_elapsed_time_;
int first_rtp_seq_num_;
int64_t first_rtp_timestamp_;
uint32_t prev_elapsed_time_;
int prev_rtp_seq_num_;
int64_t prev_rtp_timestamp_;
class RtpDumpWriter {
explicit RtpDumpWriter(rtc::StreamInterface* stream);
// Filter to control what packets we actually record.
void set_packet_filter(int filter);
// Write a RTP or RTCP packet. The parameters data points to the packet and
// data_len is its length.
rtc::StreamResult WriteRtpPacket(const void* data, size_t data_len) {
return WritePacket(data, data_len, GetElapsedTime(), false);
rtc::StreamResult WriteRtcpPacket(const void* data, size_t data_len) {
return WritePacket(data, data_len, GetElapsedTime(), true);
rtc::StreamResult WritePacket(const RtpDumpPacket& packet) {
return WritePacket(&[0],, packet.elapsed_time,
uint32_t GetElapsedTime() const;
bool GetDumpSize(size_t* size) {
// Note that we use GetPosition(), rather than GetSize(), to avoid flush the
// stream per write.
return stream_ && size && stream_->GetPosition(size);
rtc::StreamResult WriteFileHeader();
rtc::StreamResult WritePacket(const void* data,
size_t data_len,
uint32_t elapsed,
bool rtcp);
size_t FilterPacket(const void* data, size_t data_len, bool rtcp);
rtc::StreamResult WriteToStream(const void* data, size_t data_len);
rtc::StreamInterface* stream_;
int packet_filter_;
bool file_header_written_;
int64_t start_time_ms_; // Time when the record starts.
// If writing to the stream takes longer than this many ms, log a warning.
int64_t warn_slow_writes_delay_;
} // namespace cricket