blob: aeb535a574e82a8a0e74de10d9024e72ae34977f [file] [log] [blame]
* 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 <stdio.h>
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/typedefs.h"
namespace webrtc {
// 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 {
Channel(int16_t chID = -1);
~Channel() override;
int32_t SendData(FrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
size_t payloadSize,
const RTPFragmentationHeader* fragmentation) override;
void RegisterReceiverACM(AudioCodingModule *acm);
void ResetStats();
int16_t Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats);
void Stats(uint32_t* numPackets);
void Stats(uint8_t* payloadType, uint32_t* payloadLenByte);
void PrintStats(CodecInst& codecInst);
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;
void CalcStatistics(WebRtcRTPHeader& rtpInfo, 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];
rtc::CriticalSection _channelCritSect;
FILE* _bitStreamFile;
bool _saveBitStream;
int16_t _lastPayloadType;
ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
bool _isStereo;
WebRtcRTPHeader _rtpInfo;
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