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Diffstat (limited to 'webrtc/modules/audio_coding/main/test/Channel.h')
-rw-r--r-- | webrtc/modules/audio_coding/main/test/Channel.h | 130 |
1 files changed, 0 insertions, 130 deletions
diff --git a/webrtc/modules/audio_coding/main/test/Channel.h b/webrtc/modules/audio_coding/main/test/Channel.h deleted file mode 100644 index 39d4dabd98..0000000000 --- a/webrtc/modules/audio_coding/main/test/Channel.h +++ /dev/null @@ -1,130 +0,0 @@ -/* - * 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_MAIN_TEST_CHANNEL_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_ - -#include <stdio.h> - -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/interface/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_MAIN_TEST_CHANNEL_H_ |