1 files changed, 117 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/test/Channel.h b/third_party/libwebrtc/modules/audio_coding/test/Channel.h
new file mode 100644
index 0000000000..7a8829e1d2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/Channel.h
@@ -0,0 +1,117 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_TEST_CHANNEL_H_
+#define MODULES_AUDIO_CODING_TEST_CHANNEL_H_
+
+#include <stdio.h>
+
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+#define MAX_NUM_PAYLOADS 50
+#define MAX_NUM_FRAMESIZES 6
+
+// TODO(turajs): Write constructor for this structure.
+struct ACMTestFrameSizeStats {
+  uint16_t frameSizeSample;
+  size_t maxPayloadLen;
+  uint32_t numPackets;
+  uint64_t totalPayloadLenByte;
+  uint64_t totalEncodedSamples;
+  double rateBitPerSec;
+  double usageLenSec;
+};
+
+// TODO(turajs): Write constructor for this structure.
+struct ACMTestPayloadStats {
+  bool newPacket;
+  int16_t payloadType;
+  size_t lastPayloadLenByte;
+  uint32_t lastTimestamp;
+  ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
+};
+
+class Channel : public AudioPacketizationCallback {
+ public:
+  Channel(int16_t chID = -1);
+  ~Channel() override;
+
+  int32_t SendData(AudioFrameType frameType,
+                   uint8_t payloadType,
+                   uint32_t timeStamp,
+                   const uint8_t* payloadData,
+                   size_t payloadSize,
+                   int64_t absolute_capture_timestamp_ms) override;
+
+  void RegisterReceiverACM(AudioCodingModule* acm);
+
+  void ResetStats();
+
+  void SetIsStereo(bool isStereo) { _isStereo = isStereo; }
+
+  uint32_t LastInTimestamp();
+
+  void SetFECTestWithPacketLoss(bool usePacketLoss) {
+    _useFECTestWithPacketLoss = usePacketLoss;
+  }
+
+  double BitRate();
+
+  void set_send_timestamp(uint32_t new_send_ts) {
+    external_send_timestamp_ = new_send_ts;
+  }
+
+  void set_sequence_number(uint16_t new_sequence_number) {
+    external_sequence_number_ = new_sequence_number;
+  }
+
+  void set_num_packets_to_drop(int new_num_packets_to_drop) {
+    num_packets_to_drop_ = new_num_packets_to_drop;
+  }
+
+ private:
+  void CalcStatistics(const RTPHeader& rtp_header, size_t payloadSize);
+
+  AudioCodingModule* _receiverACM;
+  uint16_t _seqNo;
+  // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
+  uint8_t _payloadData[60 * 32 * 2 * 2];
+
+  Mutex _channelCritSect;
+  FILE* _bitStreamFile;
+  bool _saveBitStream;
+  int16_t _lastPayloadType;
+  ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
+  bool _isStereo;
+  RTPHeader _rtp_header;
+  bool _leftChannel;
+  uint32_t _lastInTimestamp;
+  bool _useLastFrameSize;
+  uint32_t _lastFrameSizeSample;
+  // FEC Test variables
+  int16_t _packetLoss;
+  bool _useFECTestWithPacketLoss;
+  uint64_t _beginTime;
+  uint64_t _totalBytes;
+
+  // External timing info, defaulted to -1. Only used if they are
+  // non-negative.
+  int64_t external_send_timestamp_;
+  int32_t external_sequence_number_;
+  int num_packets_to_drop_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_TEST_CHANNEL_H_