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62238097c9
This is a safe cleanup change since top-level const applied to parameters in function declarations (that are not also definitions) are ignored by the compiler. Hence, such changes do not change the type of the declared functions and are simply no-ops. Bug: webrtc:13610 Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35802}
110 lines
2.9 KiB
C++
110 lines
2.9 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/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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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(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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int64_t absolute_capture_timestamp_ms) override;
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private:
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static void MakeRTPheader(uint8_t* rtpHeader,
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uint8_t payloadType,
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int16_t seqNo,
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uint32_t timeStamp,
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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,
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RTPStream* rtpStream,
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std::string in_file_name,
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int in_sample_rate,
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int payload_type,
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SdpAudioFormat format);
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void Teardown();
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void Run();
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bool Add10MsData();
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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,
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RTPStream* rtpStream,
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std::string out_file_name,
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size_t channels,
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int file_num);
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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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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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RTPHeader _rtpHeader;
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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 {
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public:
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EncodeDecodeTest();
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void Perform();
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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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