| /* |
| * libjingle |
| * Copyright 2010 Google Inc. |
| * |
| * Redistribution and use in source and binary forms, with or without |
| * modification, are permitted provided that the following conditions are met: |
| * |
| * 1. Redistributions of source code must retain the above copyright notice, |
| * this list of conditions and the following disclaimer. |
| * 2. Redistributions in binary form must reproduce the above copyright notice, |
| * this list of conditions and the following disclaimer in the documentation |
| * and/or other materials provided with the distribution. |
| * 3. The name of the author may not be used to endorse or promote products |
| * derived from this software without specific prior written permission. |
| * |
| * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED |
| * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF |
| * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO |
| * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, |
| * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, |
| * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; |
| * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, |
| * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR |
| * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF |
| * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| */ |
| |
| #include "talk/media/base/rtpdump.h" |
| |
| #include <ctype.h> |
| |
| #include <string> |
| |
| #include "talk/media/base/rtputils.h" |
| #include "webrtc/base/byteorder.h" |
| #include "webrtc/base/logging.h" |
| #include "webrtc/base/timeutils.h" |
| |
| namespace { |
| static const int kRtpSsrcOffset = 8; |
| const int kWarnSlowWritesDelayMs = 50; |
| } // namespace |
| |
| namespace cricket { |
| |
| const char RtpDumpFileHeader::kFirstLine[] = "#!rtpplay1.0 0.0.0.0/0\n"; |
| |
| RtpDumpFileHeader::RtpDumpFileHeader(uint32_t start_ms, uint32_t s, uint16_t p) |
| : start_sec(start_ms / 1000), |
| start_usec(start_ms % 1000 * 1000), |
| source(s), |
| port(p), |
| padding(0) { |
| } |
| |
| void RtpDumpFileHeader::WriteToByteBuffer(rtc::ByteBuffer* buf) { |
| buf->WriteUInt32(start_sec); |
| buf->WriteUInt32(start_usec); |
| buf->WriteUInt32(source); |
| buf->WriteUInt16(port); |
| buf->WriteUInt16(padding); |
| } |
| |
| static const uint32_t kDefaultTimeIncrease = 30; |
| |
| bool RtpDumpPacket::IsValidRtpPacket() const { |
| return original_data_len >= data.size() && |
| data.size() >= kMinRtpPacketLen; |
| } |
| |
| bool RtpDumpPacket::IsValidRtcpPacket() const { |
| return original_data_len == 0 && |
| data.size() >= kMinRtcpPacketLen; |
| } |
| |
| bool RtpDumpPacket::GetRtpPayloadType(int* pt) const { |
| return IsValidRtpPacket() && |
| cricket::GetRtpPayloadType(&data[0], data.size(), pt); |
| } |
| |
| bool RtpDumpPacket::GetRtpSeqNum(int* seq_num) const { |
| return IsValidRtpPacket() && |
| cricket::GetRtpSeqNum(&data[0], data.size(), seq_num); |
| } |
| |
| bool RtpDumpPacket::GetRtpTimestamp(uint32_t* ts) const { |
| return IsValidRtpPacket() && |
| cricket::GetRtpTimestamp(&data[0], data.size(), ts); |
| } |
| |
| bool RtpDumpPacket::GetRtpSsrc(uint32_t* ssrc) const { |
| return IsValidRtpPacket() && |
| cricket::GetRtpSsrc(&data[0], data.size(), ssrc); |
| } |
| |
| bool RtpDumpPacket::GetRtpHeaderLen(size_t* len) const { |
| return IsValidRtpPacket() && |
| cricket::GetRtpHeaderLen(&data[0], data.size(), len); |
| } |
| |
| bool RtpDumpPacket::GetRtcpType(int* type) const { |
| return IsValidRtcpPacket() && |
| cricket::GetRtcpType(&data[0], data.size(), type); |
| } |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // Implementation of RtpDumpReader. |
| /////////////////////////////////////////////////////////////////////////// |
| |
| void RtpDumpReader::SetSsrc(uint32_t ssrc) { |
| ssrc_override_ = ssrc; |
| } |
| |
| rtc::StreamResult RtpDumpReader::ReadPacket(RtpDumpPacket* packet) { |
| if (!packet) return rtc::SR_ERROR; |
| |
| rtc::StreamResult res = rtc::SR_SUCCESS; |
| // Read the file header if it has not been read yet. |
| if (!file_header_read_) { |
| res = ReadFileHeader(); |
| if (res != rtc::SR_SUCCESS) { |
| return res; |
| } |
| file_header_read_ = true; |
| } |
| |
| // Read the RTP dump packet header. |
| char header[RtpDumpPacket::kHeaderLength]; |
| res = stream_->ReadAll(header, sizeof(header), NULL, NULL); |
| if (res != rtc::SR_SUCCESS) { |
| return res; |
| } |
| rtc::ByteBuffer buf(header, sizeof(header)); |
| uint16_t dump_packet_len; |
| uint16_t data_len; |
| // Read the full length of the rtpdump packet, including the rtpdump header. |
| buf.ReadUInt16(&dump_packet_len); |
| packet->data.resize(dump_packet_len - sizeof(header)); |
| // Read the size of the original packet, which may be larger than the size in |
| // the rtpdump file, in the event that only part of the packet (perhaps just |
| // the header) was recorded. Note that this field is set to zero for RTCP |
| // packets, which have their own internal length field. |
| buf.ReadUInt16(&data_len); |
| packet->original_data_len = data_len; |
| // Read the elapsed time for this packet (different than RTP timestamp). |
| buf.ReadUInt32(&packet->elapsed_time); |
| |
| // Read the actual RTP or RTCP packet. |
| res = stream_->ReadAll(&packet->data[0], packet->data.size(), NULL, NULL); |
| |
| // If the packet is RTP and we have specified a ssrc, replace the RTP ssrc |
| // with the specified ssrc. |
| if (res == rtc::SR_SUCCESS && |
| packet->IsValidRtpPacket() && |
| ssrc_override_ != 0) { |
| rtc::SetBE32(&packet->data[kRtpSsrcOffset], ssrc_override_); |
| } |
| |
| return res; |
| } |
| |
| rtc::StreamResult RtpDumpReader::ReadFileHeader() { |
| // Read the first line. |
| std::string first_line; |
| rtc::StreamResult res = stream_->ReadLine(&first_line); |
| if (res != rtc::SR_SUCCESS) { |
| return res; |
| } |
| if (!CheckFirstLine(first_line)) { |
| return rtc::SR_ERROR; |
| } |
| |
| // Read the 16 byte file header. |
| char header[RtpDumpFileHeader::kHeaderLength]; |
| res = stream_->ReadAll(header, sizeof(header), NULL, NULL); |
| if (res == rtc::SR_SUCCESS) { |
| rtc::ByteBuffer buf(header, sizeof(header)); |
| uint32_t start_sec; |
| uint32_t start_usec; |
| buf.ReadUInt32(&start_sec); |
| buf.ReadUInt32(&start_usec); |
| start_time_ms_ = start_sec * 1000 + start_usec / 1000; |
| // Increase the length by 1 since first_line does not contain the ending \n. |
| first_line_and_file_header_len_ = first_line.size() + 1 + sizeof(header); |
| } |
| return res; |
| } |
| |
| bool RtpDumpReader::CheckFirstLine(const std::string& first_line) { |
| // The first line is like "#!rtpplay1.0 address/port" |
| bool matched = (0 == first_line.find("#!rtpplay1.0 ")); |
| |
| // The address could be IP or hostname. We do not check it here. Instead, we |
| // check the port at the end. |
| size_t pos = first_line.find('/'); |
| matched &= (pos != std::string::npos && pos < first_line.size() - 1); |
| for (++pos; pos < first_line.size() && matched; ++pos) { |
| matched &= (0 != isdigit(first_line[pos])); |
| } |
| |
| return matched; |
| } |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // Implementation of RtpDumpLoopReader. |
| /////////////////////////////////////////////////////////////////////////// |
| RtpDumpLoopReader::RtpDumpLoopReader(rtc::StreamInterface* stream) |
| : RtpDumpReader(stream), |
| loop_count_(0), |
| elapsed_time_increases_(0), |
| rtp_seq_num_increase_(0), |
| rtp_timestamp_increase_(0), |
| packet_count_(0), |
| frame_count_(0), |
| first_elapsed_time_(0), |
| first_rtp_seq_num_(0), |
| first_rtp_timestamp_(0), |
| prev_elapsed_time_(0), |
| prev_rtp_seq_num_(0), |
| prev_rtp_timestamp_(0) { |
| } |
| |
| rtc::StreamResult RtpDumpLoopReader::ReadPacket(RtpDumpPacket* packet) { |
| if (!packet) return rtc::SR_ERROR; |
| |
| rtc::StreamResult res = RtpDumpReader::ReadPacket(packet); |
| if (rtc::SR_SUCCESS == res) { |
| if (0 == loop_count_) { |
| // During the first loop, we update the statistics of the input stream. |
| UpdateStreamStatistics(*packet); |
| } |
| } else if (rtc::SR_EOS == res) { |
| if (0 == loop_count_) { |
| // At the end of the first loop, calculate elapsed_time_increases_, |
| // rtp_seq_num_increase_, and rtp_timestamp_increase_, which will be |
| // used during the second and later loops. |
| CalculateIncreases(); |
| } |
| |
| // Rewind the input stream to the first dump packet and read again. |
| ++loop_count_; |
| if (RewindToFirstDumpPacket()) { |
| res = RtpDumpReader::ReadPacket(packet); |
| } |
| } |
| |
| if (rtc::SR_SUCCESS == res && loop_count_ > 0) { |
| // During the second and later loops, we update the elapsed time of the dump |
| // packet. If the dumped packet is a RTP packet, we also update its RTP |
| // sequence number and timestamp. |
| UpdateDumpPacket(packet); |
| } |
| |
| return res; |
| } |
| |
| void RtpDumpLoopReader::UpdateStreamStatistics(const RtpDumpPacket& packet) { |
| // Get the RTP sequence number and timestamp of the dump packet. |
| int rtp_seq_num = 0; |
| packet.GetRtpSeqNum(&rtp_seq_num); |
| uint32_t rtp_timestamp = 0; |
| packet.GetRtpTimestamp(&rtp_timestamp); |
| |
| // Set the timestamps and sequence number for the first dump packet. |
| if (0 == packet_count_++) { |
| first_elapsed_time_ = packet.elapsed_time; |
| first_rtp_seq_num_ = rtp_seq_num; |
| first_rtp_timestamp_ = rtp_timestamp; |
| // The first packet belongs to a new payload frame. |
| ++frame_count_; |
| } else if (rtp_timestamp != prev_rtp_timestamp_) { |
| // The current and previous packets belong to different payload frames. |
| ++frame_count_; |
| } |
| |
| prev_elapsed_time_ = packet.elapsed_time; |
| prev_rtp_timestamp_ = rtp_timestamp; |
| prev_rtp_seq_num_ = rtp_seq_num; |
| } |
| |
| void RtpDumpLoopReader::CalculateIncreases() { |
| // At this time, prev_elapsed_time_, prev_rtp_seq_num_, and |
| // prev_rtp_timestamp_ are values of the last dump packet in the input stream. |
| rtp_seq_num_increase_ = prev_rtp_seq_num_ - first_rtp_seq_num_ + 1; |
| // If we have only one packet or frame, we use the default timestamp |
| // increase. Otherwise, we use the difference between the first and the last |
| // packets or frames. |
| elapsed_time_increases_ = packet_count_ <= 1 ? kDefaultTimeIncrease : |
| (prev_elapsed_time_ - first_elapsed_time_) * packet_count_ / |
| (packet_count_ - 1); |
| rtp_timestamp_increase_ = frame_count_ <= 1 ? kDefaultTimeIncrease : |
| (prev_rtp_timestamp_ - first_rtp_timestamp_) * frame_count_ / |
| (frame_count_ - 1); |
| } |
| |
| void RtpDumpLoopReader::UpdateDumpPacket(RtpDumpPacket* packet) { |
| // Increase the elapsed time of the dump packet. |
| packet->elapsed_time += loop_count_ * elapsed_time_increases_; |
| |
| if (packet->IsValidRtpPacket()) { |
| // Get the old RTP sequence number and timestamp. |
| int sequence = 0; |
| packet->GetRtpSeqNum(&sequence); |
| uint32_t timestamp = 0; |
| packet->GetRtpTimestamp(×tamp); |
| // Increase the RTP sequence number and timestamp. |
| sequence += loop_count_ * rtp_seq_num_increase_; |
| timestamp += loop_count_ * rtp_timestamp_increase_; |
| // Write the updated sequence number and timestamp back to the RTP packet. |
| rtc::ByteBuffer buffer; |
| buffer.WriteUInt16(sequence); |
| buffer.WriteUInt32(timestamp); |
| memcpy(&packet->data[2], buffer.Data(), buffer.Length()); |
| } |
| } |
| |
| /////////////////////////////////////////////////////////////////////////// |
| // Implementation of RtpDumpWriter. |
| /////////////////////////////////////////////////////////////////////////// |
| |
| RtpDumpWriter::RtpDumpWriter(rtc::StreamInterface* stream) |
| : stream_(stream), |
| packet_filter_(PF_ALL), |
| file_header_written_(false), |
| start_time_ms_(rtc::Time()), |
| warn_slow_writes_delay_(kWarnSlowWritesDelayMs) { |
| } |
| |
| void RtpDumpWriter::set_packet_filter(int filter) { |
| packet_filter_ = filter; |
| LOG(LS_INFO) << "RtpDumpWriter set_packet_filter to " << packet_filter_; |
| } |
| |
| uint32_t RtpDumpWriter::GetElapsedTime() const { |
| return rtc::TimeSince(start_time_ms_); |
| } |
| |
| rtc::StreamResult RtpDumpWriter::WriteFileHeader() { |
| rtc::StreamResult res = WriteToStream( |
| RtpDumpFileHeader::kFirstLine, |
| strlen(RtpDumpFileHeader::kFirstLine)); |
| if (res != rtc::SR_SUCCESS) { |
| return res; |
| } |
| |
| rtc::ByteBuffer buf; |
| RtpDumpFileHeader file_header(rtc::Time(), 0, 0); |
| file_header.WriteToByteBuffer(&buf); |
| return WriteToStream(buf.Data(), buf.Length()); |
| } |
| |
| rtc::StreamResult RtpDumpWriter::WritePacket(const void* data, |
| size_t data_len, |
| uint32_t elapsed, |
| bool rtcp) { |
| if (!stream_ || !data || 0 == data_len) return rtc::SR_ERROR; |
| |
| rtc::StreamResult res = rtc::SR_SUCCESS; |
| // Write the file header if it has not been written yet. |
| if (!file_header_written_) { |
| res = WriteFileHeader(); |
| if (res != rtc::SR_SUCCESS) { |
| return res; |
| } |
| file_header_written_ = true; |
| } |
| |
| // Figure out what to write. |
| size_t write_len = FilterPacket(data, data_len, rtcp); |
| if (write_len == 0) { |
| return rtc::SR_SUCCESS; |
| } |
| |
| // Write the dump packet header. |
| rtc::ByteBuffer buf; |
| buf.WriteUInt16( |
| static_cast<uint16_t>(RtpDumpPacket::kHeaderLength + write_len)); |
| buf.WriteUInt16(static_cast<uint16_t>(rtcp ? 0 : data_len)); |
| buf.WriteUInt32(elapsed); |
| res = WriteToStream(buf.Data(), buf.Length()); |
| if (res != rtc::SR_SUCCESS) { |
| return res; |
| } |
| |
| // Write the header or full packet as indicated by write_len. |
| return WriteToStream(data, write_len); |
| } |
| |
| size_t RtpDumpWriter::FilterPacket(const void* data, size_t data_len, |
| bool rtcp) { |
| size_t filtered_len = 0; |
| if (!rtcp) { |
| if ((packet_filter_ & PF_RTPPACKET) == PF_RTPPACKET) { |
| // RTP header + payload |
| filtered_len = data_len; |
| } else if ((packet_filter_ & PF_RTPHEADER) == PF_RTPHEADER) { |
| // RTP header only |
| size_t header_len; |
| if (GetRtpHeaderLen(data, data_len, &header_len)) { |
| filtered_len = header_len; |
| } |
| } |
| } else { |
| if ((packet_filter_ & PF_RTCPPACKET) == PF_RTCPPACKET) { |
| // RTCP header + payload |
| filtered_len = data_len; |
| } |
| } |
| |
| return filtered_len; |
| } |
| |
| rtc::StreamResult RtpDumpWriter::WriteToStream( |
| const void* data, size_t data_len) { |
| uint32_t before = rtc::Time(); |
| rtc::StreamResult result = |
| stream_->WriteAll(data, data_len, NULL, NULL); |
| uint32_t delay = rtc::TimeSince(before); |
| if (delay >= warn_slow_writes_delay_) { |
| LOG(LS_WARNING) << "Slow RtpDump: took " << delay << "ms to write " |
| << data_len << " bytes."; |
| } |
| return result; |
| } |
| |
| } // namespace cricket |