ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
131 lines
3.5 KiB
C++
131 lines
3.5 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/include/audio_coding_module.h"
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#include "webrtc/modules/include/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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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();
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int32_t SendData(FrameType 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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int16_t Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats);
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void Stats(uint32_t* numPackets);
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void Stats(uint8_t* payloadType, uint32_t* payloadLenByte);
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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, 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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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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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 // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
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