47658f1269
RTPStream, and NetEq as such. Also mark all other virtual overrides in the same files. This will make further changes to these classes safer by ensuring that the compile breaks if the base class changes and not all overrides are fixed. This also deletes ACMTest.cc, which existed solely to define ~ACMTest(), which was marked pure virtual in the header. (Pure virtual destructors still need a definition.) Because there is another pure virtual method in this class, the class is already abstract, so there's no benefit to making the desturctor pure. Making it non-pure allows removing the separate source file. BUG=none TEST=none R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29389004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7144 4adac7df-926f-26a2-2b94-8c16560cd09d
128 lines
3.4 KiB
C++
128 lines
3.4 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
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#include <stdio.h>
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#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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#define MAX_NUM_PAYLOADS 50
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#define MAX_NUM_FRAMESIZES 6
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// TODO(turajs): Write constructor for this structure.
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struct ACMTestFrameSizeStats {
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uint16_t frameSizeSample;
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int16_t maxPayloadLen;
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uint32_t numPackets;
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uint64_t totalPayloadLenByte;
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uint64_t totalEncodedSamples;
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double rateBitPerSec;
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double usageLenSec;
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};
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// TODO(turajs): Write constructor for this structure.
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struct ACMTestPayloadStats {
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bool newPacket;
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int16_t payloadType;
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int16_t lastPayloadLenByte;
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uint32_t lastTimestamp;
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ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
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};
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class Channel : public AudioPacketizationCallback {
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public:
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Channel(int16_t chID = -1);
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~Channel();
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virtual int32_t SendData(
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const FrameType frameType, const uint8_t payloadType,
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const uint32_t timeStamp, const uint8_t* payloadData,
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const uint16_t payloadSize,
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const RTPFragmentationHeader* fragmentation) OVERRIDE;
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void RegisterReceiverACM(AudioCodingModule *acm);
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void ResetStats();
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int16_t Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats);
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void Stats(uint32_t* numPackets);
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void Stats(uint8_t* payloadLenByte, uint32_t* payloadType);
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void PrintStats(CodecInst& codecInst);
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void SetIsStereo(bool isStereo) {
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_isStereo = isStereo;
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}
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uint32_t LastInTimestamp();
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void SetFECTestWithPacketLoss(bool usePacketLoss) {
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_useFECTestWithPacketLoss = usePacketLoss;
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}
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double BitRate();
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void set_send_timestamp(uint32_t new_send_ts) {
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external_send_timestamp_ = new_send_ts;
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}
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void set_sequence_number(uint16_t new_sequence_number) {
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external_sequence_number_ = new_sequence_number;
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}
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void set_num_packets_to_drop(int new_num_packets_to_drop) {
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num_packets_to_drop_ = new_num_packets_to_drop;
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}
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private:
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void CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize);
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AudioCodingModule* _receiverACM;
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uint16_t _seqNo;
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// 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
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uint8_t _payloadData[60 * 32 * 2 * 2];
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CriticalSectionWrapper* _channelCritSect;
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FILE* _bitStreamFile;
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bool _saveBitStream;
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int16_t _lastPayloadType;
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ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
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bool _isStereo;
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WebRtcRTPHeader _rtpInfo;
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bool _leftChannel;
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uint32_t _lastInTimestamp;
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// FEC Test variables
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int16_t _packetLoss;
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bool _useFECTestWithPacketLoss;
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uint64_t _beginTime;
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uint64_t _totalBytes;
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// External timing info, defaulted to -1. Only used if they are
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// non-negative.
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int64_t external_send_timestamp_;
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int32_t external_sequence_number_;
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int num_packets_to_drop_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
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