Move talk/media to webrtc/media

talk/media/base/rtpdataengine.cc

Index: talk/media/base/rtpdataengine.cc
diff --git a/talk/media/base/rtpdataengine.cc b/talk/media/base/rtpdataengine.cc
deleted file mode 100644
index 9b26280560477c4bc07fbef8ccee43ed7eb9e717..0000000000000000000000000000000000000000
--- a/talk/media/base/rtpdataengine.cc
+++ /dev/null
@@ -1,370 +0,0 @@
-/*
- * libjingle
- * Copyright 2012 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/rtpdataengine.h"
-
-#include "talk/media/base/codec.h"
-#include "talk/media/base/constants.h"
-#include "talk/media/base/rtputils.h"
-#include "talk/media/base/streamparams.h"
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/helpers.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/ratelimiter.h"
-#include "webrtc/base/timing.h"
-
-namespace cricket {
-
-// We want to avoid IP fragmentation.
-static const size_t kDataMaxRtpPacketLen = 1200U;
-// We reserve space after the RTP header for future wiggle room.
-static const unsigned char kReservedSpace[] = {
-  0x00, 0x00, 0x00, 0x00
-};
-
-// Amount of overhead SRTP may take.  We need to leave room in the
-// buffer for it, otherwise SRTP will fail later.  If SRTP ever uses
-// more than this, we need to increase this number.
-static const size_t kMaxSrtpHmacOverhead = 16;
-
-RtpDataEngine::RtpDataEngine() {
-  data_codecs_.push_back(
-      DataCodec(kGoogleRtpDataCodecId,
-                kGoogleRtpDataCodecName, 0));
-  SetTiming(new rtc::Timing());
-}
-
-DataMediaChannel* RtpDataEngine::CreateChannel(
-    DataChannelType data_channel_type) {
-  if (data_channel_type != DCT_RTP) {
-    return NULL;
-  }
-  return new RtpDataMediaChannel(timing_.get());
-}
-
-bool FindCodecByName(const std::vector<DataCodec>& codecs,
-                     const std::string& name, DataCodec* codec_out) {
-  std::vector<DataCodec>::const_iterator iter;
-  for (iter = codecs.begin(); iter != codecs.end(); ++iter) {
-    if (iter->name == name) {
-      *codec_out = *iter;
-      return true;
-    }
-  }
-  return false;
-}
-
-RtpDataMediaChannel::RtpDataMediaChannel(rtc::Timing* timing) {
-  Construct(timing);
-}
-
-RtpDataMediaChannel::RtpDataMediaChannel() {
-  Construct(NULL);
-}
-
-void RtpDataMediaChannel::Construct(rtc::Timing* timing) {
-  sending_ = false;
-  receiving_ = false;
-  timing_ = timing;
-  send_limiter_.reset(new rtc::RateLimiter(kDataMaxBandwidth / 8, 1.0));
-}
-
-
-RtpDataMediaChannel::~RtpDataMediaChannel() {
-  std::map<uint32_t, RtpClock*>::const_iterator iter;
-  for (iter = rtp_clock_by_send_ssrc_.begin();
-       iter != rtp_clock_by_send_ssrc_.end();
-       ++iter) {
-    delete iter->second;
-  }
-}
-
-void RtpClock::Tick(double now, int* seq_num, uint32_t* timestamp) {
-  *seq_num = ++last_seq_num_;
-  *timestamp = timestamp_offset_ + static_cast<uint32_t>(now * clockrate_);
-}
-
-const DataCodec* FindUnknownCodec(const std::vector<DataCodec>& codecs) {
-  DataCodec data_codec(kGoogleRtpDataCodecId, kGoogleRtpDataCodecName, 0);
-  std::vector<DataCodec>::const_iterator iter;
-  for (iter = codecs.begin(); iter != codecs.end(); ++iter) {
-    if (!iter->Matches(data_codec)) {
-      return &(*iter);
-    }
-  }
-  return NULL;
-}
-
-const DataCodec* FindKnownCodec(const std::vector<DataCodec>& codecs) {
-  DataCodec data_codec(kGoogleRtpDataCodecId, kGoogleRtpDataCodecName, 0);
-  std::vector<DataCodec>::const_iterator iter;
-  for (iter = codecs.begin(); iter != codecs.end(); ++iter) {
-    if (iter->Matches(data_codec)) {
-      return &(*iter);
-    }
-  }
-  return NULL;
-}
-
-bool RtpDataMediaChannel::SetRecvCodecs(const std::vector<DataCodec>& codecs) {
-  const DataCodec* unknown_codec = FindUnknownCodec(codecs);
-  if (unknown_codec) {
-    LOG(LS_WARNING) << "Failed to SetRecvCodecs because of unknown codec: "
-                    << unknown_codec->ToString();
-    return false;
-  }
-
-  recv_codecs_ = codecs;
-  return true;
-}
-
-bool RtpDataMediaChannel::SetSendCodecs(const std::vector<DataCodec>& codecs) {
-  const DataCodec* known_codec = FindKnownCodec(codecs);
-  if (!known_codec) {
-    LOG(LS_WARNING) <<
-        "Failed to SetSendCodecs because there is no known codec.";
-    return false;
-  }
-
-  send_codecs_ = codecs;
-  return true;
-}
-
-bool RtpDataMediaChannel::SetSendParameters(const DataSendParameters& params) {
-  return (SetSendCodecs(params.codecs) &&
-          SetMaxSendBandwidth(params.max_bandwidth_bps));
-}
-
-bool RtpDataMediaChannel::SetRecvParameters(const DataRecvParameters& params) {
-  return SetRecvCodecs(params.codecs);
-}
-
-bool RtpDataMediaChannel::AddSendStream(const StreamParams& stream) {
-  if (!stream.has_ssrcs()) {
-    return false;
-  }
-
-  if (GetStreamBySsrc(send_streams_, stream.first_ssrc())) {
-    LOG(LS_WARNING) << "Not adding data send stream '" << stream.id
-                    << "' with ssrc=" << stream.first_ssrc()
-                    << " because stream already exists.";
-    return false;
-  }
-
-  send_streams_.push_back(stream);
-  // TODO(pthatcher): This should be per-stream, not per-ssrc.
-  // And we should probably allow more than one per stream.
-  rtp_clock_by_send_ssrc_[stream.first_ssrc()] = new RtpClock(
-      kDataCodecClockrate,
-      rtc::CreateRandomNonZeroId(), rtc::CreateRandomNonZeroId());
-
-  LOG(LS_INFO) << "Added data send stream '" << stream.id
-               << "' with ssrc=" << stream.first_ssrc();
-  return true;
-}
-
-bool RtpDataMediaChannel::RemoveSendStream(uint32_t ssrc) {
-  if (!GetStreamBySsrc(send_streams_, ssrc)) {
-    return false;
-  }
-
-  RemoveStreamBySsrc(&send_streams_, ssrc);
-  delete rtp_clock_by_send_ssrc_[ssrc];
-  rtp_clock_by_send_ssrc_.erase(ssrc);
-  return true;
-}
-
-bool RtpDataMediaChannel::AddRecvStream(const StreamParams& stream) {
-  if (!stream.has_ssrcs()) {
-    return false;
-  }
-
-  if (GetStreamBySsrc(recv_streams_, stream.first_ssrc())) {
-    LOG(LS_WARNING) << "Not adding data recv stream '" << stream.id
-                    << "' with ssrc=" << stream.first_ssrc()
-                    << " because stream already exists.";
-    return false;
-  }
-
-  recv_streams_.push_back(stream);
-  LOG(LS_INFO) << "Added data recv stream '" << stream.id
-               << "' with ssrc=" << stream.first_ssrc();
-  return true;
-}
-
-bool RtpDataMediaChannel::RemoveRecvStream(uint32_t ssrc) {
-  RemoveStreamBySsrc(&recv_streams_, ssrc);
-  return true;
-}
-
-void RtpDataMediaChannel::OnPacketReceived(
-    rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
-  RtpHeader header;
-  if (!GetRtpHeader(packet->data(), packet->size(), &header)) {
-    // Don't want to log for every corrupt packet.
-    // LOG(LS_WARNING) << "Could not read rtp header from packet of length "
-    //                 << packet->length() << ".";
-    return;
-  }
-
-  size_t header_length;
-  if (!GetRtpHeaderLen(packet->data(), packet->size(), &header_length)) {
-    // Don't want to log for every corrupt packet.
-    // LOG(LS_WARNING) << "Could not read rtp header"
-    //                 << length from packet of length "
-    //                 << packet->length() << ".";
-    return;
-  }
-  const char* data =
-      packet->data<char>() + header_length + sizeof(kReservedSpace);
-  size_t data_len = packet->size() - header_length - sizeof(kReservedSpace);
-
-  if (!receiving_) {
-    LOG(LS_WARNING) << "Not receiving packet "
-                    << header.ssrc << ":" << header.seq_num
-                    << " before SetReceive(true) called.";
-    return;
-  }
-
-  DataCodec codec;
-  if (!FindCodecById(recv_codecs_, header.payload_type, &codec)) {
-    // For bundling, this will be logged for every message.
-    // So disable this logging.
-    // LOG(LS_WARNING) << "Not receiving packet "
-    //                << header.ssrc << ":" << header.seq_num
-    //                << " (" << data_len << ")"
-    //                << " because unknown payload id: " << header.payload_type;
-    return;
-  }
-
-  if (!GetStreamBySsrc(recv_streams_, header.ssrc)) {
-    LOG(LS_WARNING) << "Received packet for unknown ssrc: " << header.ssrc;
-    return;
-  }
-
-  // Uncomment this for easy debugging.
-  // const auto* found_stream = GetStreamBySsrc(recv_streams_, header.ssrc);
-  // LOG(LS_INFO) << "Received packet"
-  //              << " groupid=" << found_stream.groupid
-  //              << ", ssrc=" << header.ssrc
-  //              << ", seqnum=" << header.seq_num
-  //              << ", timestamp=" << header.timestamp
-  //              << ", len=" << data_len;
-
-  ReceiveDataParams params;
-  params.ssrc = header.ssrc;
-  params.seq_num = header.seq_num;
-  params.timestamp = header.timestamp;
-  SignalDataReceived(params, data, data_len);
-}
-
-bool RtpDataMediaChannel::SetMaxSendBandwidth(int bps) {
-  if (bps <= 0) {
-    bps = kDataMaxBandwidth;
-  }
-  send_limiter_.reset(new rtc::RateLimiter(bps / 8, 1.0));
-  LOG(LS_INFO) << "RtpDataMediaChannel::SetSendBandwidth to " << bps << "bps.";
-  return true;
-}
-
-bool RtpDataMediaChannel::SendData(
-    const SendDataParams& params,
-    const rtc::Buffer& payload,
-    SendDataResult* result) {
-  if (result) {
-    // If we return true, we'll set this to SDR_SUCCESS.
-    *result = SDR_ERROR;
-  }
-  if (!sending_) {
-    LOG(LS_WARNING) << "Not sending packet with ssrc=" << params.ssrc
-                    << " len=" << payload.size() << " before SetSend(true).";
-    return false;
-  }
-
-  if (params.type != cricket::DMT_TEXT) {
-    LOG(LS_WARNING) << "Not sending data because binary type is unsupported.";
-    return false;
-  }
-
-  const StreamParams* found_stream =
-      GetStreamBySsrc(send_streams_, params.ssrc);
-  if (!found_stream) {
-    LOG(LS_WARNING) << "Not sending data because ssrc is unknown: "
-                    << params.ssrc;
-    return false;
-  }
-
-  DataCodec found_codec;
-  if (!FindCodecByName(send_codecs_, kGoogleRtpDataCodecName, &found_codec)) {
-    LOG(LS_WARNING) << "Not sending data because codec is unknown: "
-                    << kGoogleRtpDataCodecName;
-    return false;
-  }
-
-  size_t packet_len = (kMinRtpPacketLen + sizeof(kReservedSpace) +
-                       payload.size() + kMaxSrtpHmacOverhead);
-  if (packet_len > kDataMaxRtpPacketLen) {
-    return false;
-  }
-
-  double now = timing_->TimerNow();
-
-  if (!send_limiter_->CanUse(packet_len, now)) {
-    LOG(LS_VERBOSE) << "Dropped data packet of len=" << packet_len
-                    << "; already sent " << send_limiter_->used_in_period()
-                    << "/" << send_limiter_->max_per_period();
-    return false;
-  }
-
-  RtpHeader header;
-  header.payload_type = found_codec.id;
-  header.ssrc = params.ssrc;
-  rtp_clock_by_send_ssrc_[header.ssrc]->Tick(
-      now, &header.seq_num, &header.timestamp);
-
-  rtc::Buffer packet(kMinRtpPacketLen, packet_len);
-  if (!SetRtpHeader(packet.data(), packet.size(), header)) {
-    return false;
-  }
-  packet.AppendData(kReservedSpace);
-  packet.AppendData(payload);
-
-  LOG(LS_VERBOSE) << "Sent RTP data packet: "
-                  << " stream=" << found_stream->id << " ssrc=" << header.ssrc
-                  << ", seqnum=" << header.seq_num
-                  << ", timestamp=" << header.timestamp
-                  << ", len=" << payload.size();
-
-  MediaChannel::SendPacket(&packet, rtc::PacketOptions());
-  send_limiter_->Use(packet_len, now);
-  if (result) {
-    *result = SDR_SUCCESS;
-  }
-  return true;
-}
-
-}  // namespace cricket