Channel.h - mozsearch

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/*

 *  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/acm2/acm_receiver.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(acm2::AcmReceiver* acm_receiver);

  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);

  acm2::AcmReceiver* _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_