audio_coding: remove "main" directory

webrtc/modules/audio_coding/main/test/Channel.cc

Index: webrtc/modules/audio_coding/main/test/Channel.cc
diff --git a/webrtc/modules/audio_coding/main/test/Channel.cc b/webrtc/modules/audio_coding/main/test/Channel.cc
deleted file mode 100644
index 02bd783a38e884cbb2557a971f3c2fdbeb065563..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/test/Channel.cc
+++ /dev/null
@@ -1,424 +0,0 @@
-/*
- *  Copyright (c) 2012 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 "webrtc/modules/audio_coding/main/test/Channel.h"
-
-#include <assert.h>
-#include <iostream>
-
-#include "webrtc/base/format_macros.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-
-namespace webrtc {
-
-int32_t Channel::SendData(FrameType frameType,
-                          uint8_t payloadType,
-                          uint32_t timeStamp,
-                          const uint8_t* payloadData,
-                          size_t payloadSize,
-                          const RTPFragmentationHeader* fragmentation) {
-  WebRtcRTPHeader rtpInfo;
-  int32_t status;
-  size_t payloadDataSize = payloadSize;
-
-  rtpInfo.header.markerBit = false;
-  rtpInfo.header.ssrc = 0;
-  rtpInfo.header.sequenceNumber = (external_sequence_number_ < 0) ?
-      _seqNo++ : static_cast<uint16_t>(external_sequence_number_);
-  rtpInfo.header.payloadType = payloadType;
-  rtpInfo.header.timestamp = (external_send_timestamp_ < 0) ? timeStamp :
-      static_cast<uint32_t>(external_send_timestamp_);
-
-  if (frameType == kAudioFrameCN) {
-    rtpInfo.type.Audio.isCNG = true;
-  } else {
-    rtpInfo.type.Audio.isCNG = false;
-  }
-  if (frameType == kEmptyFrame) {
-    // When frame is empty, we should not transmit it. The frame size of the
-    // next non-empty frame will be based on the previous frame size.
-    _useLastFrameSize = _lastFrameSizeSample > 0;
-    return 0;
-  }
-
-  rtpInfo.type.Audio.channel = 1;
-  // Treat fragmentation separately
-  if (fragmentation != NULL) {
-    // If silence for too long, send only new data.
-    if ((fragmentation->fragmentationVectorSize == 2) &&
-        (fragmentation->fragmentationTimeDiff[1] <= 0x3fff)) {
-      // only 0x80 if we have multiple blocks
-      _payloadData[0] = 0x80 + fragmentation->fragmentationPlType[1];
-      size_t REDheader = (fragmentation->fragmentationTimeDiff[1] << 10) +
-          fragmentation->fragmentationLength[1];
-      _payloadData[1] = uint8_t((REDheader >> 16) & 0x000000FF);
-      _payloadData[2] = uint8_t((REDheader >> 8) & 0x000000FF);
-      _payloadData[3] = uint8_t(REDheader & 0x000000FF);
-
-      _payloadData[4] = fragmentation->fragmentationPlType[0];
-      // copy the RED data
-      memcpy(_payloadData + 5,
-             payloadData + fragmentation->fragmentationOffset[1],
-             fragmentation->fragmentationLength[1]);
-      // copy the normal data
-      memcpy(_payloadData + 5 + fragmentation->fragmentationLength[1],
-             payloadData + fragmentation->fragmentationOffset[0],
-             fragmentation->fragmentationLength[0]);
-      payloadDataSize += 5;
-    } else {
-      // single block (newest one)
-      memcpy(_payloadData, payloadData + fragmentation->fragmentationOffset[0],
-             fragmentation->fragmentationLength[0]);
-      payloadDataSize = fragmentation->fragmentationLength[0];
-      rtpInfo.header.payloadType = fragmentation->fragmentationPlType[0];
-    }
-  } else {
-    memcpy(_payloadData, payloadData, payloadDataSize);
-    if (_isStereo) {
-      if (_leftChannel) {
-        memcpy(&_rtpInfo, &rtpInfo, sizeof(WebRtcRTPHeader));
-        _leftChannel = false;
-        rtpInfo.type.Audio.channel = 1;
-      } else {
-        memcpy(&rtpInfo, &_rtpInfo, sizeof(WebRtcRTPHeader));
-        _leftChannel = true;
-        rtpInfo.type.Audio.channel = 2;
-      }
-    }
-  }
-
-  _channelCritSect->Enter();
-  if (_saveBitStream) {
-    //fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile);
-  }
-
-  if (!_isStereo) {
-    CalcStatistics(rtpInfo, payloadSize);
-  }
-  _useLastFrameSize = false;
-  _lastInTimestamp = timeStamp;
-  _totalBytes += payloadDataSize;
-  _channelCritSect->Leave();
-
-  if (_useFECTestWithPacketLoss) {
-    _packetLoss += 1;
-    if (_packetLoss == 3) {
-      _packetLoss = 0;
-      return 0;
-    }
-  }
-
-  if (num_packets_to_drop_ > 0) {
-    num_packets_to_drop_--;
-    return 0;
-  }
-
-  status = _receiverACM->IncomingPacket(_payloadData, payloadDataSize, rtpInfo);
-
-  return status;
-}
-
-// TODO(turajs): rewite this method.
-void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
-  int n;
-  if ((rtpInfo.header.payloadType != _lastPayloadType)
-      && (_lastPayloadType != -1)) {
-    // payload-type is changed.
-    // we have to terminate the calculations on the previous payload type
-    // we ignore the last packet in that payload type just to make things
-    // easier.
-    for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
-      if (_lastPayloadType == _payloadStats[n].payloadType) {
-        _payloadStats[n].newPacket = true;
-        break;
-      }
-    }
-  }
-  _lastPayloadType = rtpInfo.header.payloadType;
-
-  bool newPayload = true;
-  ACMTestPayloadStats* currentPayloadStr = NULL;
-  for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
-    if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) {
-      newPayload = false;
-      currentPayloadStr = &_payloadStats[n];
-      break;
-    }
-  }
-
-  if (!newPayload) {
-    if (!currentPayloadStr->newPacket) {
-      if (!_useLastFrameSize) {
-        _lastFrameSizeSample = (uint32_t) ((uint32_t) rtpInfo.header.timestamp -
-            (uint32_t) currentPayloadStr->lastTimestamp);
-      }
-      assert(_lastFrameSizeSample > 0);
-      int k = 0;
-      for (; k < MAX_NUM_FRAMESIZES; ++k) {
-        if ((currentPayloadStr->frameSizeStats[k].frameSizeSample ==
-            _lastFrameSizeSample) ||
-            (currentPayloadStr->frameSizeStats[k].frameSizeSample == 0)) {
-          break;
-        }
-      }
-      if (k == MAX_NUM_FRAMESIZES) {
-        // New frame size found but no space to count statistics on it. Skip it.
-        printf("No memory to store statistics for payload %d : frame size %d\n",
-               _lastPayloadType, _lastFrameSizeSample);
-        return;
-      }
-      ACMTestFrameSizeStats* currentFrameSizeStats = &(currentPayloadStr
-          ->frameSizeStats[k]);
-      currentFrameSizeStats->frameSizeSample = (int16_t) _lastFrameSizeSample;
-
-      // increment the number of encoded samples.
-      currentFrameSizeStats->totalEncodedSamples += _lastFrameSizeSample;
-      // increment the number of recveived packets
-      currentFrameSizeStats->numPackets++;
-      // increment the total number of bytes (this is based on
-      // the previous payload we don't know the frame-size of
-      // the current payload.
-      currentFrameSizeStats->totalPayloadLenByte += currentPayloadStr
-          ->lastPayloadLenByte;
-      // store the maximum payload-size (this is based on
-      // the previous payload we don't know the frame-size of
-      // the current payload.
-      if (currentFrameSizeStats->maxPayloadLen
-          < currentPayloadStr->lastPayloadLenByte) {
-        currentFrameSizeStats->maxPayloadLen = currentPayloadStr
-            ->lastPayloadLenByte;
-      }
-      // store the current values for the next time
-      currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
-      currentPayloadStr->lastPayloadLenByte = payloadSize;
-    } else {
-      currentPayloadStr->newPacket = false;
-      currentPayloadStr->lastPayloadLenByte = payloadSize;
-      currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
-      currentPayloadStr->payloadType = rtpInfo.header.payloadType;
-      memset(currentPayloadStr->frameSizeStats, 0, MAX_NUM_FRAMESIZES *
-             sizeof(ACMTestFrameSizeStats));
-    }
-  } else {
-    n = 0;
-    while (_payloadStats[n].payloadType != -1) {
-      n++;
-    }
-    // first packet
-    _payloadStats[n].newPacket = false;
-    _payloadStats[n].lastPayloadLenByte = payloadSize;
-    _payloadStats[n].lastTimestamp = rtpInfo.header.timestamp;
-    _payloadStats[n].payloadType = rtpInfo.header.payloadType;
-    memset(_payloadStats[n].frameSizeStats, 0, MAX_NUM_FRAMESIZES *
-           sizeof(ACMTestFrameSizeStats));
-  }
-}
-
-Channel::Channel(int16_t chID)
-    : _receiverACM(NULL),
-      _seqNo(0),
-      _channelCritSect(CriticalSectionWrapper::CreateCriticalSection()),
-      _bitStreamFile(NULL),
-      _saveBitStream(false),
-      _lastPayloadType(-1),
-      _isStereo(false),
-      _leftChannel(true),
-      _lastInTimestamp(0),
-      _useLastFrameSize(false),
-      _lastFrameSizeSample(0),
-      _packetLoss(0),
-      _useFECTestWithPacketLoss(false),
-      _beginTime(TickTime::MillisecondTimestamp()),
-      _totalBytes(0),
-      external_send_timestamp_(-1),
-      external_sequence_number_(-1),
-      num_packets_to_drop_(0) {
-  int n;
-  int k;
-  for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
-    _payloadStats[n].payloadType = -1;
-    _payloadStats[n].newPacket = true;
-    for (k = 0; k < MAX_NUM_FRAMESIZES; k++) {
-      _payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
-      _payloadStats[n].frameSizeStats[k].maxPayloadLen = 0;
-      _payloadStats[n].frameSizeStats[k].numPackets = 0;
-      _payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
-      _payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
-    }
-  }
-  if (chID >= 0) {
-    _saveBitStream = true;
-    char bitStreamFileName[500];
-    sprintf(bitStreamFileName, "bitStream_%d.dat", chID);
-    _bitStreamFile = fopen(bitStreamFileName, "wb");
-  } else {
-    _saveBitStream = false;
-  }
-}
-
-Channel::~Channel() {
-  delete _channelCritSect;
-}
-
-void Channel::RegisterReceiverACM(AudioCodingModule* acm) {
-  _receiverACM = acm;
-  return;
-}
-
-void Channel::ResetStats() {
-  int n;
-  int k;
-  _channelCritSect->Enter();
-  _lastPayloadType = -1;
-  for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
-    _payloadStats[n].payloadType = -1;
-    _payloadStats[n].newPacket = true;
-    for (k = 0; k < MAX_NUM_FRAMESIZES; k++) {
-      _payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
-      _payloadStats[n].frameSizeStats[k].maxPayloadLen = 0;
-      _payloadStats[n].frameSizeStats[k].numPackets = 0;
-      _payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
-      _payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
-    }
-  }
-  _beginTime = TickTime::MillisecondTimestamp();
-  _totalBytes = 0;
-  _channelCritSect->Leave();
-}
-
-int16_t Channel::Stats(CodecInst& codecInst,
-                       ACMTestPayloadStats& payloadStats) {
-  _channelCritSect->Enter();
-  int n;
-  payloadStats.payloadType = -1;
-  for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
-    if (_payloadStats[n].payloadType == codecInst.pltype) {
-      memcpy(&payloadStats, &_payloadStats[n], sizeof(ACMTestPayloadStats));
-      break;
-    }
-  }
-  if (payloadStats.payloadType == -1) {
-    _channelCritSect->Leave();
-    return -1;
-  }
-  for (n = 0; n < MAX_NUM_FRAMESIZES; n++) {
-    if (payloadStats.frameSizeStats[n].frameSizeSample == 0) {
-      _channelCritSect->Leave();
-      return 0;
-    }
-    payloadStats.frameSizeStats[n].usageLenSec = (double) payloadStats
-        .frameSizeStats[n].totalEncodedSamples / (double) codecInst.plfreq;
-
-    payloadStats.frameSizeStats[n].rateBitPerSec =
-        payloadStats.frameSizeStats[n].totalPayloadLenByte * 8
-            / payloadStats.frameSizeStats[n].usageLenSec;
-
-  }
-  _channelCritSect->Leave();
-  return 0;
-}
-
-void Channel::Stats(uint32_t* numPackets) {
-  _channelCritSect->Enter();
-  int k;
-  int n;
-  memset(numPackets, 0, MAX_NUM_PAYLOADS * sizeof(uint32_t));
-  for (k = 0; k < MAX_NUM_PAYLOADS; k++) {
-    if (_payloadStats[k].payloadType == -1) {
-      break;
-    }
-    numPackets[k] = 0;
-    for (n = 0; n < MAX_NUM_FRAMESIZES; n++) {
-      if (_payloadStats[k].frameSizeStats[n].frameSizeSample == 0) {
-        break;
-      }
-      numPackets[k] += _payloadStats[k].frameSizeStats[n].numPackets;
-    }
-  }
-  _channelCritSect->Leave();
-}
-
-void Channel::Stats(uint8_t* payloadType, uint32_t* payloadLenByte) {
-  _channelCritSect->Enter();
-
-  int k;
-  int n;
-  memset(payloadLenByte, 0, MAX_NUM_PAYLOADS * sizeof(uint32_t));
-  for (k = 0; k < MAX_NUM_PAYLOADS; k++) {
-    if (_payloadStats[k].payloadType == -1) {
-      break;
-    }
-    payloadType[k] = (uint8_t) _payloadStats[k].payloadType;
-    payloadLenByte[k] = 0;
-    for (n = 0; n < MAX_NUM_FRAMESIZES; n++) {
-      if (_payloadStats[k].frameSizeStats[n].frameSizeSample == 0) {
-        break;
-      }
-      payloadLenByte[k] += (uint16_t) _payloadStats[k].frameSizeStats[n]
-          .totalPayloadLenByte;
-    }
-  }
-
-  _channelCritSect->Leave();
-}
-
-void Channel::PrintStats(CodecInst& codecInst) {
-  ACMTestPayloadStats payloadStats;
-  Stats(codecInst, payloadStats);
-  printf("%s %d kHz\n", codecInst.plname, codecInst.plfreq / 1000);
-  printf("=====================================================\n");
-  if (payloadStats.payloadType == -1) {
-    printf("No Packets are sent with payload-type %d (%s)\n\n",
-           codecInst.pltype, codecInst.plname);
-    return;
-  }
-  for (int k = 0; k < MAX_NUM_FRAMESIZES; k++) {
-    if (payloadStats.frameSizeStats[k].frameSizeSample == 0) {
-      break;
-    }
-    printf("Frame-size.................... %d samples\n",
-           payloadStats.frameSizeStats[k].frameSizeSample);
-    printf("Average Rate.................. %.0f bits/sec\n",
-           payloadStats.frameSizeStats[k].rateBitPerSec);
-    printf("Maximum Payload-Size.......... %" PRIuS " Bytes\n",
-           payloadStats.frameSizeStats[k].maxPayloadLen);
-    printf(
-        "Maximum Instantaneous Rate.... %.0f bits/sec\n",
-        ((double) payloadStats.frameSizeStats[k].maxPayloadLen * 8.0
-            * (double) codecInst.plfreq)
-            / (double) payloadStats.frameSizeStats[k].frameSizeSample);
-    printf("Number of Packets............. %u\n",
-           (unsigned int) payloadStats.frameSizeStats[k].numPackets);
-    printf("Duration...................... %0.3f sec\n\n",
-           payloadStats.frameSizeStats[k].usageLenSec);
-
-  }
-
-}
-
-uint32_t Channel::LastInTimestamp() {
-  uint32_t timestamp;
-  _channelCritSect->Enter();
-  timestamp = _lastInTimestamp;
-  _channelCritSect->Leave();
-  return timestamp;
-}
-
-double Channel::BitRate() {
-  double rate;
-  uint64_t currTime = TickTime::MillisecondTimestamp();
-  _channelCritSect->Enter();
-  rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime);
-  _channelCritSect->Leave();
-  return rate;
-}
-
-}  // namespace webrtc