28e2075280
trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
425 lines
14 KiB
C++
425 lines
14 KiB
C++
/*
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* libjingle
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* Copyright 2010 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#include "talk/media/base/rtpdump.h"
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#include <ctype.h>
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#include <string>
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#include "talk/base/byteorder.h"
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#include "talk/base/logging.h"
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#include "talk/base/timeutils.h"
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#include "talk/media/base/rtputils.h"
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namespace {
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static const int kRtpSsrcOffset = 8;
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const int kWarnSlowWritesDelayMs = 50;
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} // namespace
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namespace cricket {
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const char RtpDumpFileHeader::kFirstLine[] = "#!rtpplay1.0 0.0.0.0/0\n";
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RtpDumpFileHeader::RtpDumpFileHeader(uint32 start_ms, uint32 s, uint16 p)
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: start_sec(start_ms / 1000),
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start_usec(start_ms % 1000 * 1000),
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source(s),
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port(p),
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padding(0) {
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}
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void RtpDumpFileHeader::WriteToByteBuffer(talk_base::ByteBuffer* buf) {
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buf->WriteUInt32(start_sec);
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buf->WriteUInt32(start_usec);
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buf->WriteUInt32(source);
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buf->WriteUInt16(port);
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buf->WriteUInt16(padding);
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}
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static const uint32 kDefaultTimeIncrease = 30;
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bool RtpDumpPacket::IsValidRtpPacket() const {
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return original_data_len >= data.size() &&
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data.size() >= kMinRtpPacketLen;
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}
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bool RtpDumpPacket::IsValidRtcpPacket() const {
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return original_data_len == 0 &&
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data.size() >= kMinRtcpPacketLen;
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}
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bool RtpDumpPacket::GetRtpPayloadType(int* pt) const {
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return IsValidRtpPacket() &&
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cricket::GetRtpPayloadType(&data[0], data.size(), pt);
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}
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bool RtpDumpPacket::GetRtpSeqNum(int* seq_num) const {
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return IsValidRtpPacket() &&
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cricket::GetRtpSeqNum(&data[0], data.size(), seq_num);
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}
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bool RtpDumpPacket::GetRtpTimestamp(uint32* ts) const {
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return IsValidRtpPacket() &&
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cricket::GetRtpTimestamp(&data[0], data.size(), ts);
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}
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bool RtpDumpPacket::GetRtpSsrc(uint32* ssrc) const {
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return IsValidRtpPacket() &&
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cricket::GetRtpSsrc(&data[0], data.size(), ssrc);
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}
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bool RtpDumpPacket::GetRtpHeaderLen(size_t* len) const {
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return IsValidRtpPacket() &&
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cricket::GetRtpHeaderLen(&data[0], data.size(), len);
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}
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bool RtpDumpPacket::GetRtcpType(int* type) const {
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return IsValidRtcpPacket() &&
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cricket::GetRtcpType(&data[0], data.size(), type);
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}
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///////////////////////////////////////////////////////////////////////////
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// Implementation of RtpDumpReader.
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///////////////////////////////////////////////////////////////////////////
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void RtpDumpReader::SetSsrc(uint32 ssrc) {
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ssrc_override_ = ssrc;
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}
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talk_base::StreamResult RtpDumpReader::ReadPacket(RtpDumpPacket* packet) {
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if (!packet) return talk_base::SR_ERROR;
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talk_base::StreamResult res = talk_base::SR_SUCCESS;
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// Read the file header if it has not been read yet.
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if (!file_header_read_) {
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res = ReadFileHeader();
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if (res != talk_base::SR_SUCCESS) {
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return res;
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}
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file_header_read_ = true;
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}
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// Read the RTP dump packet header.
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char header[RtpDumpPacket::kHeaderLength];
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res = stream_->ReadAll(header, sizeof(header), NULL, NULL);
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if (res != talk_base::SR_SUCCESS) {
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return res;
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}
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talk_base::ByteBuffer buf(header, sizeof(header));
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uint16 dump_packet_len;
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uint16 data_len;
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// Read the full length of the rtpdump packet, including the rtpdump header.
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buf.ReadUInt16(&dump_packet_len);
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packet->data.resize(dump_packet_len - sizeof(header));
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// Read the size of the original packet, which may be larger than the size in
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// the rtpdump file, in the event that only part of the packet (perhaps just
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// the header) was recorded. Note that this field is set to zero for RTCP
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// packets, which have their own internal length field.
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buf.ReadUInt16(&data_len);
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packet->original_data_len = data_len;
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// Read the elapsed time for this packet (different than RTP timestamp).
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buf.ReadUInt32(&packet->elapsed_time);
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// Read the actual RTP or RTCP packet.
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res = stream_->ReadAll(&packet->data[0], packet->data.size(), NULL, NULL);
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// If the packet is RTP and we have specified a ssrc, replace the RTP ssrc
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// with the specified ssrc.
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if (res == talk_base::SR_SUCCESS &&
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packet->IsValidRtpPacket() &&
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ssrc_override_ != 0) {
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talk_base::SetBE32(&packet->data[kRtpSsrcOffset], ssrc_override_);
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}
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return res;
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}
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talk_base::StreamResult RtpDumpReader::ReadFileHeader() {
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// Read the first line.
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std::string first_line;
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talk_base::StreamResult res = stream_->ReadLine(&first_line);
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if (res != talk_base::SR_SUCCESS) {
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return res;
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}
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if (!CheckFirstLine(first_line)) {
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return talk_base::SR_ERROR;
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}
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// Read the 16 byte file header.
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char header[RtpDumpFileHeader::kHeaderLength];
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res = stream_->ReadAll(header, sizeof(header), NULL, NULL);
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if (res == talk_base::SR_SUCCESS) {
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talk_base::ByteBuffer buf(header, sizeof(header));
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uint32 start_sec;
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uint32 start_usec;
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buf.ReadUInt32(&start_sec);
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buf.ReadUInt32(&start_usec);
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start_time_ms_ = start_sec * 1000 + start_usec / 1000;
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// Increase the length by 1 since first_line does not contain the ending \n.
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first_line_and_file_header_len_ = first_line.size() + 1 + sizeof(header);
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}
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return res;
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}
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bool RtpDumpReader::CheckFirstLine(const std::string& first_line) {
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// The first line is like "#!rtpplay1.0 address/port"
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bool matched = (0 == first_line.find("#!rtpplay1.0 "));
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// The address could be IP or hostname. We do not check it here. Instead, we
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// check the port at the end.
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size_t pos = first_line.find('/');
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matched &= (pos != std::string::npos && pos < first_line.size() - 1);
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for (++pos; pos < first_line.size() && matched; ++pos) {
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matched &= (0 != isdigit(first_line[pos]));
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}
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return matched;
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}
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///////////////////////////////////////////////////////////////////////////
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// Implementation of RtpDumpLoopReader.
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///////////////////////////////////////////////////////////////////////////
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RtpDumpLoopReader::RtpDumpLoopReader(talk_base::StreamInterface* stream)
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: RtpDumpReader(stream),
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loop_count_(0),
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elapsed_time_increases_(0),
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rtp_seq_num_increase_(0),
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rtp_timestamp_increase_(0),
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packet_count_(0),
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frame_count_(0),
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first_elapsed_time_(0),
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first_rtp_seq_num_(0),
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first_rtp_timestamp_(0),
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prev_elapsed_time_(0),
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prev_rtp_seq_num_(0),
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prev_rtp_timestamp_(0) {
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}
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talk_base::StreamResult RtpDumpLoopReader::ReadPacket(RtpDumpPacket* packet) {
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if (!packet) return talk_base::SR_ERROR;
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talk_base::StreamResult res = RtpDumpReader::ReadPacket(packet);
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if (talk_base::SR_SUCCESS == res) {
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if (0 == loop_count_) {
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// During the first loop, we update the statistics of the input stream.
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UpdateStreamStatistics(*packet);
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}
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} else if (talk_base::SR_EOS == res) {
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if (0 == loop_count_) {
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// At the end of the first loop, calculate elapsed_time_increases_,
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// rtp_seq_num_increase_, and rtp_timestamp_increase_, which will be
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// used during the second and later loops.
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CalculateIncreases();
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}
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// Rewind the input stream to the first dump packet and read again.
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++loop_count_;
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if (RewindToFirstDumpPacket()) {
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res = RtpDumpReader::ReadPacket(packet);
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}
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}
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if (talk_base::SR_SUCCESS == res && loop_count_ > 0) {
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// During the second and later loops, we update the elapsed time of the dump
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// packet. If the dumped packet is a RTP packet, we also update its RTP
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// sequence number and timestamp.
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UpdateDumpPacket(packet);
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}
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return res;
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}
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void RtpDumpLoopReader::UpdateStreamStatistics(const RtpDumpPacket& packet) {
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// Get the RTP sequence number and timestamp of the dump packet.
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int rtp_seq_num = 0;
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packet.GetRtpSeqNum(&rtp_seq_num);
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uint32 rtp_timestamp = 0;
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packet.GetRtpTimestamp(&rtp_timestamp);
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// Set the timestamps and sequence number for the first dump packet.
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if (0 == packet_count_++) {
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first_elapsed_time_ = packet.elapsed_time;
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first_rtp_seq_num_ = rtp_seq_num;
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first_rtp_timestamp_ = rtp_timestamp;
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// The first packet belongs to a new payload frame.
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++frame_count_;
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} else if (rtp_timestamp != prev_rtp_timestamp_) {
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// The current and previous packets belong to different payload frames.
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++frame_count_;
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}
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prev_elapsed_time_ = packet.elapsed_time;
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prev_rtp_timestamp_ = rtp_timestamp;
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prev_rtp_seq_num_ = rtp_seq_num;
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}
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void RtpDumpLoopReader::CalculateIncreases() {
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// At this time, prev_elapsed_time_, prev_rtp_seq_num_, and
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// prev_rtp_timestamp_ are values of the last dump packet in the input stream.
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rtp_seq_num_increase_ = prev_rtp_seq_num_ - first_rtp_seq_num_ + 1;
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// If we have only one packet or frame, we use the default timestamp
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// increase. Otherwise, we use the difference between the first and the last
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// packets or frames.
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elapsed_time_increases_ = packet_count_ <= 1 ? kDefaultTimeIncrease :
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(prev_elapsed_time_ - first_elapsed_time_) * packet_count_ /
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(packet_count_ - 1);
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rtp_timestamp_increase_ = frame_count_ <= 1 ? kDefaultTimeIncrease :
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(prev_rtp_timestamp_ - first_rtp_timestamp_) * frame_count_ /
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(frame_count_ - 1);
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}
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void RtpDumpLoopReader::UpdateDumpPacket(RtpDumpPacket* packet) {
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// Increase the elapsed time of the dump packet.
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packet->elapsed_time += loop_count_ * elapsed_time_increases_;
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if (packet->IsValidRtpPacket()) {
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// Get the old RTP sequence number and timestamp.
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int sequence = 0;
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packet->GetRtpSeqNum(&sequence);
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uint32 timestamp = 0;
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packet->GetRtpTimestamp(×tamp);
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// Increase the RTP sequence number and timestamp.
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sequence += loop_count_ * rtp_seq_num_increase_;
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timestamp += loop_count_ * rtp_timestamp_increase_;
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// Write the updated sequence number and timestamp back to the RTP packet.
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talk_base::ByteBuffer buffer;
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buffer.WriteUInt16(sequence);
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buffer.WriteUInt32(timestamp);
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memcpy(&packet->data[2], buffer.Data(), buffer.Length());
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}
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}
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///////////////////////////////////////////////////////////////////////////
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// Implementation of RtpDumpWriter.
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///////////////////////////////////////////////////////////////////////////
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RtpDumpWriter::RtpDumpWriter(talk_base::StreamInterface* stream)
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: stream_(stream),
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packet_filter_(PF_ALL),
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file_header_written_(false),
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start_time_ms_(talk_base::Time()),
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warn_slow_writes_delay_(kWarnSlowWritesDelayMs) {
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}
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void RtpDumpWriter::set_packet_filter(int filter) {
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packet_filter_ = filter;
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LOG(LS_INFO) << "RtpDumpWriter set_packet_filter to " << packet_filter_;
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}
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uint32 RtpDumpWriter::GetElapsedTime() const {
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return talk_base::TimeSince(start_time_ms_);
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}
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talk_base::StreamResult RtpDumpWriter::WriteFileHeader() {
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talk_base::StreamResult res = WriteToStream(
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RtpDumpFileHeader::kFirstLine,
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strlen(RtpDumpFileHeader::kFirstLine));
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if (res != talk_base::SR_SUCCESS) {
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return res;
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}
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talk_base::ByteBuffer buf;
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RtpDumpFileHeader file_header(talk_base::Time(), 0, 0);
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file_header.WriteToByteBuffer(&buf);
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return WriteToStream(buf.Data(), buf.Length());
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}
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talk_base::StreamResult RtpDumpWriter::WritePacket(
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const void* data, size_t data_len, uint32 elapsed, bool rtcp) {
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if (!stream_ || !data || 0 == data_len) return talk_base::SR_ERROR;
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talk_base::StreamResult res = talk_base::SR_SUCCESS;
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// Write the file header if it has not been written yet.
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if (!file_header_written_) {
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res = WriteFileHeader();
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if (res != talk_base::SR_SUCCESS) {
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return res;
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}
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file_header_written_ = true;
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}
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// Figure out what to write.
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size_t write_len = FilterPacket(data, data_len, rtcp);
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if (write_len == 0) {
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return talk_base::SR_SUCCESS;
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}
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// Write the dump packet header.
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talk_base::ByteBuffer buf;
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buf.WriteUInt16(static_cast<uint16>(
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RtpDumpPacket::kHeaderLength + write_len));
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buf.WriteUInt16(static_cast<uint16>(rtcp ? 0 : data_len));
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buf.WriteUInt32(elapsed);
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res = WriteToStream(buf.Data(), buf.Length());
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if (res != talk_base::SR_SUCCESS) {
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return res;
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}
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// Write the header or full packet as indicated by write_len.
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return WriteToStream(data, write_len);
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}
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size_t RtpDumpWriter::FilterPacket(const void* data, size_t data_len,
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bool rtcp) {
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size_t filtered_len = 0;
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if (!rtcp) {
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if ((packet_filter_ & PF_RTPPACKET) == PF_RTPPACKET) {
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// RTP header + payload
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filtered_len = data_len;
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} else if ((packet_filter_ & PF_RTPHEADER) == PF_RTPHEADER) {
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// RTP header only
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size_t header_len;
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if (GetRtpHeaderLen(data, data_len, &header_len)) {
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filtered_len = header_len;
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}
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}
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} else {
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if ((packet_filter_ & PF_RTCPPACKET) == PF_RTCPPACKET) {
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// RTCP header + payload
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filtered_len = data_len;
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}
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}
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return filtered_len;
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}
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talk_base::StreamResult RtpDumpWriter::WriteToStream(
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const void* data, size_t data_len) {
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uint32 before = talk_base::Time();
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talk_base::StreamResult result =
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stream_->WriteAll(data, data_len, NULL, NULL);
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uint32 delay = talk_base::TimeSince(before);
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if (delay >= warn_slow_writes_delay_) {
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LOG(LS_WARNING) << "Slow RtpDump: took " << delay << "ms to write "
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<< data_len << " bytes.";
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}
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return result;
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}
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} // namespace cricket
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