
Currently, we only allow Opus DTX in combination with Opus kVoip mode. When one of them is toggled, the other might need to change as well. This CL is to introduce a flag to force a co-config. BUG= R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/40159004 Cr-Commit-Position: refs/heads/master@{#8698} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8698 4adac7df-926f-26a2-2b94-8c16560cd09d
283 lines
8.2 KiB
C++
283 lines
8.2 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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#include "webrtc/modules/audio_coding/main/test/TestVADDTX.h"
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#include <string>
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#include "webrtc/engine_configurations.h"
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#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
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#include "webrtc/modules/audio_coding/main/test/utility.h"
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#include "webrtc/test/testsupport/fileutils.h"
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namespace webrtc {
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#ifdef WEBRTC_CODEC_ISAC
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const CodecInst kIsacWb = {103, "ISAC", 16000, 480, 1, 32000};
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const CodecInst kIsacSwb = {104, "ISAC", 32000, 960, 1, 56000};
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#endif
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#ifdef WEBRTC_CODEC_ILBC
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const CodecInst kIlbc = {102, "ILBC", 8000, 240, 1, 13300};
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#endif
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#ifdef WEBRTC_CODEC_OPUS
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const CodecInst kOpus = {120, "opus", 48000, 960, 1, 64000};
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const CodecInst kOpusStereo = {120, "opus", 48000, 960, 2, 64000};
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#endif
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ActivityMonitor::ActivityMonitor() {
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ResetStatistics();
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}
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int32_t ActivityMonitor::InFrameType(FrameType frame_type) {
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counter_[frame_type]++;
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return 0;
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}
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void ActivityMonitor::PrintStatistics() {
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printf("\n");
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printf("kFrameEmpty %u\n", counter_[kFrameEmpty]);
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printf("kAudioFrameSpeech %u\n", counter_[kAudioFrameSpeech]);
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printf("kAudioFrameCN %u\n", counter_[kAudioFrameCN]);
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printf("kVideoFrameKey %u\n", counter_[kVideoFrameKey]);
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printf("kVideoFrameDelta %u\n", counter_[kVideoFrameDelta]);
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printf("\n\n");
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}
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void ActivityMonitor::ResetStatistics() {
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memset(counter_, 0, sizeof(counter_));
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}
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void ActivityMonitor::GetStatistics(uint32_t* counter) {
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memcpy(counter, counter_, sizeof(counter_));
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}
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TestVadDtx::TestVadDtx()
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: acm_send_(AudioCodingModule::Create(0)),
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acm_receive_(AudioCodingModule::Create(1)),
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channel_(new Channel),
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monitor_(new ActivityMonitor) {
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EXPECT_EQ(0, acm_send_->RegisterTransportCallback(channel_.get()));
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channel_->RegisterReceiverACM(acm_receive_.get());
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EXPECT_EQ(0, acm_send_->RegisterVADCallback(monitor_.get()));
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}
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void TestVadDtx::RegisterCodec(CodecInst codec_param) {
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// Set the codec for sending and receiving.
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EXPECT_EQ(0, acm_send_->RegisterSendCodec(codec_param));
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EXPECT_EQ(0, acm_receive_->RegisterReceiveCodec(codec_param));
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channel_->SetIsStereo(codec_param.channels > 1);
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}
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// Encoding a file and see if the numbers that various packets occur follow
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// the expectation.
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void TestVadDtx::Run(std::string in_filename, int frequency, int channels,
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std::string out_filename, bool append,
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const int* expects) {
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monitor_->ResetStatistics();
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PCMFile in_file;
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in_file.Open(in_filename, frequency, "rb");
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in_file.ReadStereo(channels > 1);
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PCMFile out_file;
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if (append) {
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out_file.Open(out_filename, kOutputFreqHz, "ab");
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} else {
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out_file.Open(out_filename, kOutputFreqHz, "wb");
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}
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uint16_t frame_size_samples = in_file.PayloadLength10Ms();
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uint32_t time_stamp = 0x12345678;
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AudioFrame audio_frame;
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while (!in_file.EndOfFile()) {
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in_file.Read10MsData(audio_frame);
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audio_frame.timestamp_ = time_stamp;
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time_stamp += frame_size_samples;
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EXPECT_GE(acm_send_->Add10MsData(audio_frame), 0);
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acm_receive_->PlayoutData10Ms(kOutputFreqHz, &audio_frame);
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out_file.Write10MsData(audio_frame);
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}
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in_file.Close();
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out_file.Close();
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#ifdef PRINT_STAT
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monitor_->PrintStatistics();
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#endif
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uint32_t stats[5];
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monitor_->GetStatistics(stats);
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monitor_->ResetStatistics();
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for (const auto& st : stats) {
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int i = &st - stats; // Calculate the current position in stats.
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switch (expects[i]) {
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case 0: {
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EXPECT_EQ(0u, st) << "stats[" << i << "] error.";
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break;
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}
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case 1: {
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EXPECT_GT(st, 0u) << "stats[" << i << "] error.";
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break;
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}
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}
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}
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}
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// Following is the implementation of TestWebRtcVadDtx.
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TestWebRtcVadDtx::TestWebRtcVadDtx()
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: vad_enabled_(false),
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dtx_enabled_(false),
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use_webrtc_dtx_(false),
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output_file_num_(0) {
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}
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void TestWebRtcVadDtx::Perform() {
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// Go through various test cases.
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#ifdef WEBRTC_CODEC_ISAC
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// Register iSAC WB as send codec
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RegisterCodec(kIsacWb);
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RunTestCases();
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// Register iSAC SWB as send codec
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RegisterCodec(kIsacSwb);
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RunTestCases();
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#endif
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#ifdef WEBRTC_CODEC_ILBC
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// Register iLBC as send codec
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RegisterCodec(kIlbc);
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RunTestCases();
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#endif
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#ifdef WEBRTC_CODEC_OPUS
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// Register Opus as send codec
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RegisterCodec(kOpus);
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RunTestCases();
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#endif
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}
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// Test various configurations on VAD/DTX.
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void TestWebRtcVadDtx::RunTestCases() {
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// #1 DTX = OFF, VAD = OFF, VADNormal
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SetVAD(false, false, VADNormal);
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Test(true);
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// #2 DTX = ON, VAD = ON, VADAggr
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SetVAD(true, true, VADAggr);
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Test(false);
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// #3 DTX = ON, VAD = ON, VADLowBitrate
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SetVAD(true, true, VADLowBitrate);
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Test(false);
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// #4 DTX = ON, VAD = ON, VADVeryAggr
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SetVAD(true, true, VADVeryAggr);
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Test(false);
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// #5 DTX = ON, VAD = ON, VADNormal
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SetVAD(true, true, VADNormal);
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Test(false);
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}
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// Set the expectation and run the test.
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void TestWebRtcVadDtx::Test(bool new_outfile) {
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int expects[] = {-1, 1, use_webrtc_dtx_, 0, 0};
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if (new_outfile) {
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output_file_num_++;
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}
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std::stringstream out_filename;
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out_filename << webrtc::test::OutputPath()
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<< "testWebRtcVadDtx_outFile_"
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<< output_file_num_
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<< ".pcm";
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Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
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32000, 1, out_filename.str(), !new_outfile, expects);
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}
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void TestWebRtcVadDtx::SetVAD(bool enable_dtx, bool enable_vad,
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ACMVADMode vad_mode) {
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ACMVADMode mode;
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EXPECT_EQ(0, acm_send_->SetVAD(enable_dtx, enable_vad, vad_mode));
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EXPECT_EQ(0, acm_send_->VAD(&dtx_enabled_, &vad_enabled_, &mode));
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CodecInst codec_param;
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acm_send_->SendCodec(&codec_param);
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if (STR_CASE_CMP(codec_param.plname, "opus") == 0) {
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// If send codec is Opus, WebRTC VAD/DTX cannot be used.
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enable_dtx = enable_vad = false;
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}
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EXPECT_EQ(dtx_enabled_ , enable_dtx); // DTX should be set as expected.
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bool replaced = false;
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acm_send_->IsInternalDTXReplacedWithWebRtc(&replaced);
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use_webrtc_dtx_ = dtx_enabled_ && replaced;
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if (use_webrtc_dtx_) {
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EXPECT_TRUE(vad_enabled_); // WebRTC DTX cannot run without WebRTC VAD.
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}
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if (!dtx_enabled_ || !use_webrtc_dtx_) {
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// Using no DTX or codec Internal DTX should not affect setting of VAD.
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EXPECT_EQ(enable_vad, vad_enabled_);
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}
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}
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// Following is the implementation of TestOpusDtx.
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void TestOpusDtx::Perform() {
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#ifdef WEBRTC_CODEC_ISAC
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// If we set other codec than Opus, DTX cannot be toggled.
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RegisterCodec(kIsacWb);
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EXPECT_EQ(-1, acm_send_->EnableOpusDtx(false));
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EXPECT_EQ(-1, acm_send_->DisableOpusDtx());
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#endif
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#ifdef WEBRTC_CODEC_OPUS
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int expects[] = {0, 1, 0, 0, 0};
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// Register Opus as send codec
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std::string out_filename = webrtc::test::OutputPath() +
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"testOpusDtx_outFile_mono.pcm";
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RegisterCodec(kOpus);
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EXPECT_EQ(0, acm_send_->DisableOpusDtx());
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Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
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32000, 1, out_filename, false, expects);
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EXPECT_EQ(0, acm_send_->EnableOpusDtx(false));
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expects[kFrameEmpty] = 1;
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Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
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32000, 1, out_filename, true, expects);
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// Register stereo Opus as send codec
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out_filename = webrtc::test::OutputPath() + "testOpusDtx_outFile_stereo.pcm";
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RegisterCodec(kOpusStereo);
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EXPECT_EQ(0, acm_send_->DisableOpusDtx());
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expects[kFrameEmpty] = 0;
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Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"),
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32000, 2, out_filename, false, expects);
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// Opus should be now in kAudio mode. Opus DTX should not be set without
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// forcing kVoip mode.
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EXPECT_EQ(-1, acm_send_->EnableOpusDtx(false));
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EXPECT_EQ(0, acm_send_->EnableOpusDtx(true));
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expects[kFrameEmpty] = 1;
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Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"),
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32000, 2, out_filename, true, expects);
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#endif
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}
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} // namespace webrtc
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