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87e2d785a0
This cl deprecates the FrameType enum, and adds aliases AudioFrameType and VideoFrameType. After downstream usage is updated, the enums will be separated and be moved out of common_types.h. Bug: webrtc:6883 Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27011}
117 lines
3.2 KiB
C++
117 lines
3.2 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 MODULES_AUDIO_CODING_TEST_CHANNEL_H_
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#define MODULES_AUDIO_CODING_TEST_CHANNEL_H_
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#include <stdio.h>
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#include "modules/audio_coding/include/audio_coding_module.h"
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#include "modules/include/module_common_types.h"
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#include "rtc_base/critical_section.h"
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namespace webrtc {
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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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size_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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size_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() override;
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int32_t SendData(AudioFrameType frameType,
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uint8_t payloadType,
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uint32_t timeStamp,
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const uint8_t* payloadData,
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size_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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void SetIsStereo(bool isStereo) { _isStereo = isStereo; }
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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(const RTPHeader& rtp_header, size_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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rtc::CriticalSection _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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RTPHeader _rtp_header;
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bool _leftChannel;
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uint32_t _lastInTimestamp;
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bool _useLastFrameSize;
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uint32_t _lastFrameSizeSample;
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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 // MODULES_AUDIO_CODING_TEST_CHANNEL_H_
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