7f3c724e12
In addition to our naming guidelines, this will cause these files to get parsed by Sonar, and to make searching/grepping the source using file extensions easier in the future. BUG= TEST=Compiling on Linux. Review URL: http://webrtc-codereview.appspot.com/348005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1405 4adac7df-926f-26a2-2b94-8c16560cd09d
411 lines
11 KiB
C++
411 lines
11 KiB
C++
/*
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* Copyright (c) 2011 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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#include "EncodeDecodeTest.h"
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include "audio_coding_module.h"
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#include "common_types.h"
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#include "gtest/gtest.h"
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#include "trace.h"
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#include "testsupport/fileutils.h"
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#include "utility.h"
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namespace webrtc {
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TestPacketization::TestPacketization(RTPStream *rtpStream,
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WebRtc_UWord16 frequency)
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: _rtpStream(rtpStream),
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_frequency(frequency),
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_seqNo(0) {
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}
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TestPacketization::~TestPacketization() { }
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WebRtc_Word32 TestPacketization::SendData(
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const FrameType /* frameType */,
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const WebRtc_UWord8 payloadType,
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const WebRtc_UWord32 timeStamp,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadSize,
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const RTPFragmentationHeader* /* fragmentation */) {
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_rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
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_frequency);
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return 1;
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}
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Sender::Sender()
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: _acm(NULL),
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_pcmFile(),
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_audioFrame(),
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_payloadSize(0),
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_timeStamp(0),
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_packetization(NULL) {
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}
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void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
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acm->InitializeSender();
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struct CodecInst sendCodec;
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int noOfCodecs = acm->NumberOfCodecs();
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int codecNo;
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if (testMode == 1) {
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// Set the codec, input file, and parameters for the current test.
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codecNo = codeId;
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// Use same input file for now.
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char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
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_pcmFile.Open(fileName, 32000, "rb");
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} else if (testMode == 0) {
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// Set the codec, input file, and parameters for the current test.
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codecNo = codeId;
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acm->Codec(codecNo, sendCodec);
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// Use same input file for now.
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char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
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_pcmFile.Open(fileName, 32000, "rb");
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} else {
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printf("List of supported codec.\n");
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for (int n = 0; n < noOfCodecs; n++) {
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acm->Codec(n, sendCodec);
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printf("%d %s\n", n, sendCodec.plname);
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}
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printf("Choose your codec:");
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ASSERT_GT(scanf("%d", &codecNo), 0);
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char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
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_pcmFile.Open(fileName, 32000, "rb");
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}
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acm->Codec(codecNo, sendCodec);
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acm->RegisterSendCodec(sendCodec);
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_packetization = new TestPacketization(rtpStream, sendCodec.plfreq);
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if (acm->RegisterTransportCallback(_packetization) < 0) {
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printf("Registering Transport Callback failed, for run: codecId: %d: --\n",
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codeId);
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}
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_acm = acm;
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}
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void Sender::Teardown() {
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_pcmFile.Close();
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delete _packetization;
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}
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bool Sender::Add10MsData() {
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if (!_pcmFile.EndOfFile()) {
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_pcmFile.Read10MsData(_audioFrame);
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WebRtc_Word32 ok = _acm->Add10MsData(_audioFrame);
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if (ok != 0) {
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printf("Error calling Add10MsData: for run: codecId: %d\n", codeId);
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exit(1);
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}
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return true;
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}
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return false;
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}
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bool Sender::Process() {
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WebRtc_Word32 ok = _acm->Process();
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if (ok < 0) {
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printf("Error calling Add10MsData: for run: codecId: %d\n", codeId);
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exit(1);
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}
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return true;
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}
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void Sender::Run() {
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while (true) {
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if (!Add10MsData()) {
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break;
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}
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if (!Process()) { // This could be done in a processing thread
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break;
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}
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}
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}
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Receiver::Receiver()
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: _playoutLengthSmpls(WEBRTC_10MS_PCM_AUDIO),
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_payloadSizeBytes(MAX_INCOMING_PAYLOAD) {
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}
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void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
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struct CodecInst recvCodec;
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int noOfCodecs;
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acm->InitializeReceiver();
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noOfCodecs = acm->NumberOfCodecs();
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for (int i = 0; i < noOfCodecs; i++) {
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acm->Codec((WebRtc_UWord8) i, recvCodec);
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if (acm->RegisterReceiveCodec(recvCodec) != 0) {
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printf("Unable to register codec: for run: codecId: %d\n", codeId);
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exit(1);
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}
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}
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char filename[128];
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_rtpStream = rtpStream;
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int playSampFreq;
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if (testMode == 1) {
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playSampFreq=recvCodec.plfreq;
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//output file for current run
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sprintf(filename,"%s/out%dFile.pcm", webrtc::test::OutputPath().c_str(),
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codeId);
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_pcmFile.Open(filename, recvCodec.plfreq, "wb+");
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} else if (testMode == 0) {
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playSampFreq=32000;
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//output file for current run
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sprintf(filename, "%s/encodeDecode_out%d.pcm",
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webrtc::test::OutputPath().c_str(), codeId);
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_pcmFile.Open(filename, 32000/*recvCodec.plfreq*/, "wb+");
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} else {
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printf("\nValid output frequencies:\n");
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printf("8000\n16000\n32000\n-1,");
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printf("which means output freq equal to received signal freq");
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printf("\n\nChoose output sampling frequency: ");
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ASSERT_GT(scanf("%d", &playSampFreq), 0);
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sprintf(filename, "%s/outFile.pcm", webrtc::test::OutputPath().c_str());
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_pcmFile.Open(filename, 32000, "wb+");
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}
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_realPayloadSizeBytes = 0;
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_playoutBuffer = new WebRtc_Word16[WEBRTC_10MS_PCM_AUDIO];
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_frequency = playSampFreq;
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_acm = acm;
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_firstTime = true;
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}
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void Receiver::Teardown() {
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delete [] _playoutBuffer;
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_pcmFile.Close();
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if (testMode > 1)
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Trace::ReturnTrace();
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}
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bool Receiver::IncomingPacket() {
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if (!_rtpStream->EndOfFile()) {
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if (_firstTime) {
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_firstTime = false;
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_realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
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_payloadSizeBytes, &_nextTime);
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if (_realPayloadSizeBytes < 0) {
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printf("Error in reading incoming payload.\n");
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return false;
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}
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if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
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_firstTime = true;
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return true;
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}
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}
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WebRtc_Word32 ok = _acm->IncomingPacket(_incomingPayload,
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_realPayloadSizeBytes, _rtpInfo);
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if (ok != 0) {
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printf("Error when inserting packet to ACM, for run: codecId: %d\n",
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codeId);
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exit(1);
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}
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_realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
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_payloadSizeBytes, &_nextTime);
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if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
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_firstTime = true;
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}
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}
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return true;
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}
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bool Receiver::PlayoutData() {
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AudioFrame audioFrame;
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if (_acm->PlayoutData10Ms(_frequency, audioFrame) != 0) {
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printf("Error when calling PlayoutData10Ms, for run: codecId: %d\n",
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codeId);
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exit(1);
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}
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if (_playoutLengthSmpls == 0) {
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return false;
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}
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_pcmFile.Write10MsData(audioFrame._payloadData,
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audioFrame._payloadDataLengthInSamples);
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return true;
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}
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void Receiver::Run() {
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WebRtc_UWord8 counter500Ms = 50;
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WebRtc_UWord32 clock = 0;
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while (counter500Ms > 0) {
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if (clock == 0 || clock >= _nextTime) {
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IncomingPacket();
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if (clock == 0) {
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clock = _nextTime;
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}
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}
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if ((clock % 10) == 0) {
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if (!PlayoutData()) {
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clock++;
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continue;
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}
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}
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if (_rtpStream->EndOfFile()) {
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counter500Ms--;
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}
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clock++;
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}
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}
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EncodeDecodeTest::EncodeDecodeTest() {
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_testMode = 2;
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Trace::CreateTrace();
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Trace::SetTraceFile("acm_encdec_test.txt");
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}
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EncodeDecodeTest::EncodeDecodeTest(int testMode) {
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//testMode == 0 for autotest
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//testMode == 1 for testing all codecs/parameters
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//testMode > 1 for specific user-input test (as it was used before)
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_testMode = testMode;
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if(_testMode != 0) {
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Trace::CreateTrace();
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Trace::SetTraceFile("acm_encdec_test.txt");
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}
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}
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void EncodeDecodeTest::Perform() {
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if (_testMode == 0) {
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printf("Running Encode/Decode Test");
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WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1,
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"---------- EncodeDecodeTest ----------");
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}
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int numCodecs = 1;
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int codePars[3]; //freq, pacsize, rate
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int playoutFreq[3]; //8, 16, 32k
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int numPars[52]; //number of codec parameters sets (rate,freq,pacsize)to test,
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//for a given codec
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codePars[0] = 0;
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codePars[1] = 0;
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codePars[2] = 0;
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if (_testMode == 1) {
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AudioCodingModule *acmTmp = AudioCodingModule::Create(0);
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struct CodecInst sendCodecTmp;
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numCodecs = acmTmp->NumberOfCodecs();
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printf("List of supported codec.\n");
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for(int n = 0; n < numCodecs; n++) {
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acmTmp->Codec(n, sendCodecTmp);
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if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) {
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numPars[n] = 0;
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} else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) {
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numPars[n] = 0;
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} else if (STR_CASE_CMP(sendCodecTmp.plname, "red") == 0) {
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numPars[n] = 0;
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} else {
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numPars[n] = 1;
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printf("%d %s\n", n, sendCodecTmp.plname);
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}
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}
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AudioCodingModule::Destroy(acmTmp);
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playoutFreq[1] = 16000;
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} else if (_testMode == 0) {
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AudioCodingModule *acmTmp = AudioCodingModule::Create(0);
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numCodecs = acmTmp->NumberOfCodecs();
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AudioCodingModule::Destroy(acmTmp);
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struct CodecInst dummyCodec;
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//chose range of testing for codecs/parameters
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for(int i = 0 ; i < numCodecs ; i++) {
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numPars[i] = 1;
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acmTmp->Codec(i, dummyCodec);
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if (STR_CASE_CMP(dummyCodec.plname, "telephone-event") == 0) {
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numPars[i] = 0;
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} else if (STR_CASE_CMP(dummyCodec.plname, "cn") == 0) {
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numPars[i] = 0;
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} else if (STR_CASE_CMP(dummyCodec.plname, "red") == 0) {
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numPars[i] = 0;
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}
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}
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playoutFreq[1] = 16000;
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} else {
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numCodecs = 1;
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numPars[0] = 1;
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playoutFreq[1]=16000;
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}
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_receiver.testMode = _testMode;
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//loop over all codecs:
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for (int codeId = 0; codeId < numCodecs; codeId++) {
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//only encode using real encoders, not telephone-event anc cn
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for (int loopPars = 1; loopPars <= numPars[codeId]; loopPars++) {
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if (_testMode == 1) {
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printf("\n");
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printf("***FOR RUN: codeId: %d\n", codeId);
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printf("\n");
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} else if (_testMode == 0) {
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printf(".");
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}
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EncodeToFile(1, codeId, codePars, _testMode);
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AudioCodingModule *acm = AudioCodingModule::Create(10);
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RTPFile rtpFile;
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std::string fileName = webrtc::test::OutputPath() + "outFile.rtp";
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rtpFile.Open(fileName.c_str(), "rb");
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_receiver.codeId = codeId;
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rtpFile.ReadHeader();
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_receiver.Setup(acm, &rtpFile);
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_receiver.Run();
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_receiver.Teardown();
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rtpFile.Close();
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AudioCodingModule::Destroy(acm);
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if (_testMode == 1) {
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printf("***COMPLETED RUN FOR: codecID: %d ***\n", codeId);
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}
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}
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}
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if (_testMode == 0) {
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printf("Done!\n");
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}
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if (_testMode == 1)
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Trace::ReturnTrace();
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}
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void EncodeDecodeTest::EncodeToFile(int fileType, int codeId, int* codePars,
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int testMode) {
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AudioCodingModule *acm = AudioCodingModule::Create(0);
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RTPFile rtpFile;
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std::string fileName = webrtc::test::OutputPath() + "outFile.rtp";
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rtpFile.Open(fileName.c_str(), "wb+");
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rtpFile.WriteHeader();
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//for auto_test and logging
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_sender.testMode = testMode;
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_sender.codeId = codeId;
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_sender.Setup(acm, &rtpFile);
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struct CodecInst sendCodecInst;
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if (acm->SendCodec(sendCodecInst) >= 0) {
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_sender.Run();
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}
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_sender.Teardown();
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rtpFile.Close();
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AudioCodingModule::Destroy(acm);
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}
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} // namespace webrtc
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