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Diffstat (limited to 'webrtc/modules/audio_coding/test/Channel.h')
-rw-r--r-- | webrtc/modules/audio_coding/test/Channel.h | 130 |
1 files changed, 130 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/test/Channel.h b/webrtc/modules/audio_coding/test/Channel.h new file mode 100644 index 0000000000..b047aa9909 --- /dev/null +++ b/webrtc/modules/audio_coding/test/Channel.h @@ -0,0 +1,130 @@ +/* + * 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 WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_ + +#include <stdio.h> + +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +class CriticalSectionWrapper; + +#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(); + + 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; + } + + private: + 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]; + + CriticalSectionWrapper* _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 + +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_ |