VAD/DTX tests: Don't let the ACM create audio encoders
It will soon lose the ability to do so. Bug: webrtc:8396 Change-Id: I06dce417bba855b57130bd1a052988b2f235dcbd Reviewed-on: https://webrtc-review.googlesource.com/102882 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24921}
This commit is contained in:
@ -12,8 +12,15 @@
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#include <string>
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "modules/audio_coding/codecs/audio_format_conversion.h"
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#include "api/audio_codecs/audio_decoder_factory_template.h"
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#include "api/audio_codecs/audio_encoder_factory_template.h"
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#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"
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#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
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#include "api/audio_codecs/isac/audio_decoder_isac_float.h"
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#include "api/audio_codecs/isac/audio_encoder_isac_float.h"
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#include "api/audio_codecs/opus/audio_decoder_opus.h"
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#include "api/audio_codecs/opus/audio_encoder_opus.h"
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#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
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#include "modules/audio_coding/test/PCMFile.h"
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#include "modules/audio_coding/test/utility.h"
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#include "rtc_base/strings/string_builder.h"
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@ -21,20 +28,6 @@
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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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@ -63,10 +56,16 @@ void ActivityMonitor::GetStatistics(uint32_t* counter) {
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}
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TestVadDtx::TestVadDtx()
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: acm_send_(AudioCodingModule::Create(
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AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
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: encoder_factory_(CreateAudioEncoderFactory<AudioEncoderIlbc,
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AudioEncoderIsacFloat,
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AudioEncoderOpus>()),
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decoder_factory_(CreateAudioDecoderFactory<AudioDecoderIlbc,
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AudioDecoderIsacFloat,
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AudioDecoderOpus>()),
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acm_send_(AudioCodingModule::Create(
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AudioCodingModule::Config(decoder_factory_))),
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acm_receive_(AudioCodingModule::Create(
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AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
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AudioCodingModule::Config(decoder_factory_))),
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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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@ -74,12 +73,29 @@ TestVadDtx::TestVadDtx()
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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(true, acm_receive_->RegisterReceiveCodec(
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codec_param.pltype, CodecInstToSdp(codec_param)));
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channel_->SetIsStereo(codec_param.channels > 1);
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bool TestVadDtx::RegisterCodec(const SdpAudioFormat& codec_format,
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absl::optional<Vad::Aggressiveness> vad_mode) {
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constexpr int payload_type = 17, cn_payload_type = 117;
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bool added_comfort_noise = false;
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auto encoder = encoder_factory_->MakeAudioEncoder(payload_type, codec_format,
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absl::nullopt);
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if (vad_mode.has_value() &&
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STR_CASE_CMP(codec_format.name.c_str(), "opus") != 0) {
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AudioEncoderCng::Config config;
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config.speech_encoder = std::move(encoder);
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config.num_channels = 1;
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config.payload_type = cn_payload_type;
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config.vad_mode = vad_mode.value();
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encoder = absl::make_unique<AudioEncoderCng>(std::move(config));
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added_comfort_noise = true;
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}
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channel_->SetIsStereo(encoder->NumChannels() > 1);
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acm_send_->SetEncoder(std::move(encoder));
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EXPECT_EQ(true,
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acm_receive_->RegisterReceiveCodec(payload_type, codec_format));
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return added_comfort_noise;
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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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@ -148,60 +164,36 @@ void TestVadDtx::Run(std::string in_filename,
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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), dtx_enabled_(false), output_file_num_(0) {}
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TestWebRtcVadDtx::TestWebRtcVadDtx() : output_file_num_(0) {}
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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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RunTestCases({"ISAC", 16000, 1});
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RunTestCases({"ISAC", 32000, 1});
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RunTestCases({"ILBC", 8000, 1});
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RunTestCases({"opus", 48000, 2});
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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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void TestWebRtcVadDtx::RunTestCases(const SdpAudioFormat& codec_format) {
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Test(/*new_outfile=*/true,
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/*expect_dtx_enabled=*/RegisterCodec(codec_format, absl::nullopt));
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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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Test(/*new_outfile=*/false,
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/*expect_dtx_enabled=*/RegisterCodec(codec_format, Vad::kVadAggressive));
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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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Test(/*new_outfile=*/false,
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/*expect_dtx_enabled=*/RegisterCodec(codec_format, Vad::kVadLowBitrate));
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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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Test(/*new_outfile=*/false, /*expect_dtx_enabled=*/RegisterCodec(
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codec_format, Vad::kVadVeryAggressive));
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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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Test(/*new_outfile=*/false,
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/*expect_dtx_enabled=*/RegisterCodec(codec_format, Vad::kVadNormal));
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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, dtx_enabled_, 0, 0};
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void TestWebRtcVadDtx::Test(bool new_outfile, bool expect_dtx_enabled) {
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int expects[] = {-1, 1, expect_dtx_enabled, 0, 0};
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if (new_outfile) {
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output_file_num_++;
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}
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@ -212,46 +204,19 @@ void TestWebRtcVadDtx::Test(bool new_outfile) {
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out_filename.str(), !new_outfile, expects);
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}
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void TestWebRtcVadDtx::SetVAD(bool enable_dtx,
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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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auto codec_param = acm_send_->SendCodec();
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ASSERT_TRUE(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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if (dtx_enabled_) {
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EXPECT_TRUE(vad_enabled_); // WebRTC DTX cannot run without WebRTC VAD.
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} else {
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// Using no 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 switched on.
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RegisterCodec(kIsacWb);
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RegisterCodec({"ISAC", 16000, 1}, absl::nullopt);
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EXPECT_EQ(-1, acm_send_->EnableOpusDtx());
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EXPECT_EQ(0, 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 =
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webrtc::test::OutputPath() + "testOpusDtx_outFile_mono.pcm";
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RegisterCodec(kOpus);
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RegisterCodec({"opus", 48000, 2}, absl::nullopt);
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EXPECT_EQ(0, acm_send_->DisableOpusDtx());
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Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1,
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@ -265,7 +230,7 @@ void TestOpusDtx::Perform() {
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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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RegisterCodec({"opus", 48000, 2, {{"stereo", "1"}}}, absl::nullopt);
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EXPECT_EQ(0, acm_send_->DisableOpusDtx());
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expects[kEmptyFrame] = 0;
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expects[kAudioFrameCN] = 0;
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@ -278,7 +243,6 @@ void TestOpusDtx::Perform() {
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expects[kAudioFrameCN] = 1;
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Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"), 32000,
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2, out_filename, true, expects);
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#endif
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}
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} // namespace webrtc
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