webrtc/talk/media/base/rtpdump.cc
henrike@webrtc.org 28e2075280 Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk


git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 00:45:36 +00:00

425 lines
14 KiB
C++

/*
* 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/base/byteorder.h"
#include "talk/base/logging.h"
#include "talk/base/timeutils.h"
#include "talk/media/base/rtputils.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 start_ms, uint32 s, uint16 p)
: start_sec(start_ms / 1000),
start_usec(start_ms % 1000 * 1000),
source(s),
port(p),
padding(0) {
}
void RtpDumpFileHeader::WriteToByteBuffer(talk_base::ByteBuffer* buf) {
buf->WriteUInt32(start_sec);
buf->WriteUInt32(start_usec);
buf->WriteUInt32(source);
buf->WriteUInt16(port);
buf->WriteUInt16(padding);
}
static const uint32 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* ts) const {
return IsValidRtpPacket() &&
cricket::GetRtpTimestamp(&data[0], data.size(), ts);
}
bool RtpDumpPacket::GetRtpSsrc(uint32* 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 ssrc) {
ssrc_override_ = ssrc;
}
talk_base::StreamResult RtpDumpReader::ReadPacket(RtpDumpPacket* packet) {
if (!packet) return talk_base::SR_ERROR;
talk_base::StreamResult res = talk_base::SR_SUCCESS;
// Read the file header if it has not been read yet.
if (!file_header_read_) {
res = ReadFileHeader();
if (res != talk_base::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 != talk_base::SR_SUCCESS) {
return res;
}
talk_base::ByteBuffer buf(header, sizeof(header));
uint16 dump_packet_len;
uint16 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 == talk_base::SR_SUCCESS &&
packet->IsValidRtpPacket() &&
ssrc_override_ != 0) {
talk_base::SetBE32(&packet->data[kRtpSsrcOffset], ssrc_override_);
}
return res;
}
talk_base::StreamResult RtpDumpReader::ReadFileHeader() {
// Read the first line.
std::string first_line;
talk_base::StreamResult res = stream_->ReadLine(&first_line);
if (res != talk_base::SR_SUCCESS) {
return res;
}
if (!CheckFirstLine(first_line)) {
return talk_base::SR_ERROR;
}
// Read the 16 byte file header.
char header[RtpDumpFileHeader::kHeaderLength];
res = stream_->ReadAll(header, sizeof(header), NULL, NULL);
if (res == talk_base::SR_SUCCESS) {
talk_base::ByteBuffer buf(header, sizeof(header));
uint32 start_sec;
uint32 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(talk_base::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) {
}
talk_base::StreamResult RtpDumpLoopReader::ReadPacket(RtpDumpPacket* packet) {
if (!packet) return talk_base::SR_ERROR;
talk_base::StreamResult res = RtpDumpReader::ReadPacket(packet);
if (talk_base::SR_SUCCESS == res) {
if (0 == loop_count_) {
// During the first loop, we update the statistics of the input stream.
UpdateStreamStatistics(*packet);
}
} else if (talk_base::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 (talk_base::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 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 timestamp = 0;
packet->GetRtpTimestamp(&timestamp);
// 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.
talk_base::ByteBuffer buffer;
buffer.WriteUInt16(sequence);
buffer.WriteUInt32(timestamp);
memcpy(&packet->data[2], buffer.Data(), buffer.Length());
}
}
///////////////////////////////////////////////////////////////////////////
// Implementation of RtpDumpWriter.
///////////////////////////////////////////////////////////////////////////
RtpDumpWriter::RtpDumpWriter(talk_base::StreamInterface* stream)
: stream_(stream),
packet_filter_(PF_ALL),
file_header_written_(false),
start_time_ms_(talk_base::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 RtpDumpWriter::GetElapsedTime() const {
return talk_base::TimeSince(start_time_ms_);
}
talk_base::StreamResult RtpDumpWriter::WriteFileHeader() {
talk_base::StreamResult res = WriteToStream(
RtpDumpFileHeader::kFirstLine,
strlen(RtpDumpFileHeader::kFirstLine));
if (res != talk_base::SR_SUCCESS) {
return res;
}
talk_base::ByteBuffer buf;
RtpDumpFileHeader file_header(talk_base::Time(), 0, 0);
file_header.WriteToByteBuffer(&buf);
return WriteToStream(buf.Data(), buf.Length());
}
talk_base::StreamResult RtpDumpWriter::WritePacket(
const void* data, size_t data_len, uint32 elapsed, bool rtcp) {
if (!stream_ || !data || 0 == data_len) return talk_base::SR_ERROR;
talk_base::StreamResult res = talk_base::SR_SUCCESS;
// Write the file header if it has not been written yet.
if (!file_header_written_) {
res = WriteFileHeader();
if (res != talk_base::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 talk_base::SR_SUCCESS;
}
// Write the dump packet header.
talk_base::ByteBuffer buf;
buf.WriteUInt16(static_cast<uint16>(
RtpDumpPacket::kHeaderLength + write_len));
buf.WriteUInt16(static_cast<uint16>(rtcp ? 0 : data_len));
buf.WriteUInt32(elapsed);
res = WriteToStream(buf.Data(), buf.Length());
if (res != talk_base::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;
}
talk_base::StreamResult RtpDumpWriter::WriteToStream(
const void* data, size_t data_len) {
uint32 before = talk_base::Time();
talk_base::StreamResult result =
stream_->WriteAll(data, data_len, NULL, NULL);
uint32 delay = talk_base::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