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88aee288f8
This test is so old, it used to be interactive with an automated mode bolted on to the side. That automatic mode is the only one that's used nowadays. Bug: webrtc:8396 Change-Id: I3b473f53ff6afa363b9691e8471a5754f46d3d3f Reviewed-on: https://webrtc-review.googlesource.com/83583 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23620}
116 lines
3 KiB
C++
116 lines
3 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_ENCODEDECODETEST_H_
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#define MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
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#include <stdio.h>
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#include <string.h>
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#include "modules/audio_coding/include/audio_coding_module.h"
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#include "modules/audio_coding/test/ACMTest.h"
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#include "modules/audio_coding/test/PCMFile.h"
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#include "modules/audio_coding/test/RTPFile.h"
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#include "modules/include/module_common_types.h"
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#include "typedefs.h" // NOLINT(build/include)
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namespace webrtc {
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#define MAX_INCOMING_PAYLOAD 8096
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// TestPacketization callback which writes the encoded payloads to file
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class TestPacketization : public AudioPacketizationCallback {
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public:
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TestPacketization(RTPStream *rtpStream, uint16_t frequency);
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~TestPacketization();
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int32_t SendData(const FrameType frameType,
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const uint8_t payloadType,
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const uint32_t timeStamp,
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const uint8_t* payloadData,
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const size_t payloadSize,
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const RTPFragmentationHeader* fragmentation) override;
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private:
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static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
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int16_t seqNo, uint32_t timeStamp, uint32_t ssrc);
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RTPStream* _rtpStream;
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int32_t _frequency;
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int16_t _seqNo;
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};
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class Sender {
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public:
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Sender();
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void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
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std::string in_file_name, int sample_rate, size_t channels);
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void Teardown();
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void Run();
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bool Add10MsData();
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uint8_t codeId;
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protected:
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AudioCodingModule* _acm;
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private:
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PCMFile _pcmFile;
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AudioFrame _audioFrame;
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TestPacketization* _packetization;
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};
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class Receiver {
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public:
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Receiver();
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virtual ~Receiver() {};
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void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
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std::string out_file_name, size_t channels);
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void Teardown();
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void Run();
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virtual bool IncomingPacket();
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bool PlayoutData();
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//for auto_test and logging
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uint8_t codeId;
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private:
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PCMFile _pcmFile;
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int16_t* _playoutBuffer;
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uint16_t _playoutLengthSmpls;
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int32_t _frequency;
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bool _firstTime;
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protected:
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AudioCodingModule* _acm;
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uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
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RTPStream* _rtpStream;
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WebRtcRTPHeader _rtpInfo;
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size_t _realPayloadSizeBytes;
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size_t _payloadSizeBytes;
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uint32_t _nextTime;
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};
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class EncodeDecodeTest : public ACMTest {
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public:
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explicit EncodeDecodeTest(int test_mode);
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void Perform() override;
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uint16_t _playoutFreq;
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private:
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std::string EncodeToFile(int fileType, int codeId, int* codePars);
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protected:
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Sender _sender;
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Receiver _receiver;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
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