Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 274 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/test/Channel.cc b/third_party/libwebrtc/modules/audio_coding/test/Channel.cc
new file mode 100644
index 0000000000..35aa6cb6b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/Channel.cc
@@ -0,0 +1,274 @@
+/*
+ *  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 "modules/audio_coding/test/Channel.h"
+
+#include <iostream>
+
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+int32_t Channel::SendData(AudioFrameType frameType,
+                          uint8_t payloadType,
+                          uint32_t timeStamp,
+                          const uint8_t* payloadData,
+                          size_t payloadSize,
+                          int64_t absolute_capture_timestamp_ms) {
+  RTPHeader rtp_header;
+  int32_t status;
+  size_t payloadDataSize = payloadSize;
+
+  rtp_header.markerBit = false;
+  rtp_header.ssrc = 0;
+  rtp_header.sequenceNumber =
+      (external_sequence_number_ < 0)
+          ? _seqNo++
+          : static_cast<uint16_t>(external_sequence_number_);
+  rtp_header.payloadType = payloadType;
+  rtp_header.timestamp = (external_send_timestamp_ < 0)
+                             ? timeStamp
+                             : static_cast<uint32_t>(external_send_timestamp_);
+
+  if (frameType == AudioFrameType::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;
+  }
+
+  memcpy(_payloadData, payloadData, payloadDataSize);
+  if (_isStereo) {
+    if (_leftChannel) {
+      _rtp_header = rtp_header;
+      _leftChannel = false;
+    } else {
+      rtp_header = _rtp_header;
+      _leftChannel = true;
+    }
+  }
+
+  _channelCritSect.Lock();
+  if (_saveBitStream) {
+    // fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile);
+  }
+
+  if (!_isStereo) {
+    CalcStatistics(rtp_header, payloadSize);
+  }
+  _useLastFrameSize = false;
+  _lastInTimestamp = timeStamp;
+  _totalBytes += payloadDataSize;
+  _channelCritSect.Unlock();
+
+  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, rtp_header);
+
+  return status;
+}
+
+// TODO(turajs): rewite this method.
+void Channel::CalcStatistics(const RTPHeader& rtp_header, size_t payloadSize) {
+  int n;
+  if ((rtp_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 = rtp_header.payloadType;
+
+  bool newPayload = true;
+  ACMTestPayloadStats* currentPayloadStr = NULL;
+  for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
+    if (rtp_header.payloadType == _payloadStats[n].payloadType) {
+      newPayload = false;
+      currentPayloadStr = &_payloadStats[n];
+      break;
+    }
+  }
+
+  if (!newPayload) {
+    if (!currentPayloadStr->newPacket) {
+      if (!_useLastFrameSize) {
+        _lastFrameSizeSample =
+            (uint32_t)((uint32_t)rtp_header.timestamp -
+                       (uint32_t)currentPayloadStr->lastTimestamp);
+      }
+      RTC_DCHECK_GT(_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 = rtp_header.timestamp;
+      currentPayloadStr->lastPayloadLenByte = payloadSize;
+    } else {
+      currentPayloadStr->newPacket = false;
+      currentPayloadStr->lastPayloadLenByte = payloadSize;
+      currentPayloadStr->lastTimestamp = rtp_header.timestamp;
+      currentPayloadStr->payloadType = rtp_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 = rtp_header.timestamp;
+    _payloadStats[n].payloadType = rtp_header.payloadType;
+    memset(_payloadStats[n].frameSizeStats, 0,
+           MAX_NUM_FRAMESIZES * sizeof(ACMTestFrameSizeStats));
+  }
+}
+
+Channel::Channel(int16_t chID)
+    : _receiverACM(NULL),
+      _seqNo(0),
+      _bitStreamFile(NULL),
+      _saveBitStream(false),
+      _lastPayloadType(-1),
+      _isStereo(false),
+      _leftChannel(true),
+      _lastInTimestamp(0),
+      _useLastFrameSize(false),
+      _lastFrameSizeSample(0),
+      _packetLoss(0),
+      _useFECTestWithPacketLoss(false),
+      _beginTime(rtc::TimeMillis()),
+      _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;
+    rtc::StringBuilder ss;
+    ss.AppendFormat("bitStream_%d.dat", chID);
+    _bitStreamFile = fopen(ss.str().c_str(), "wb");
+  } else {
+    _saveBitStream = false;
+  }
+}
+
+Channel::~Channel() {}
+
+void Channel::RegisterReceiverACM(AudioCodingModule* acm) {
+  _receiverACM = acm;
+  return;
+}
+
+void Channel::ResetStats() {
+  int n;
+  int k;
+  _channelCritSect.Lock();
+  _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 = rtc::TimeMillis();
+  _totalBytes = 0;
+  _channelCritSect.Unlock();
+}
+
+uint32_t Channel::LastInTimestamp() {
+  uint32_t timestamp;
+  _channelCritSect.Lock();
+  timestamp = _lastInTimestamp;
+  _channelCritSect.Unlock();
+  return timestamp;
+}
+
+double Channel::BitRate() {
+  double rate;
+  uint64_t currTime = rtc::TimeMillis();
+  _channelCritSect.Lock();
+  rate = ((double)_totalBytes * 8.0) / (double)(currTime - _beginTime);
+  _channelCritSect.Unlock();
+  return rate;
+}
+
+}  // namespace webrtc