Index: webrtc/modules/audio_coding/main/test/Channel.h |
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 ff6937ec08c3c92fd296ecb48f81bc7abb418904..0000000000000000000000000000000000000000 |
--- 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/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_MAIN_TEST_CHANNEL_H_ |