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Unified Diff: webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc

Issue 2295283004: Remove unused audio_e2e_harness.cc and infrastructure. (Closed)
Patch Set: rebase Created 4 years, 3 months ago
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Index: webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc
diff --git a/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc b/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc
deleted file mode 100644
index fb07b569faf406032e961e82be931f7baed8030e..0000000000000000000000000000000000000000
--- a/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Sets up a simple VoiceEngine loopback call with the default audio devices
-// and runs forever. Some parameters can be configured through command-line
-// flags.
-
-#include <memory>
-
-#include "gflags/gflags.h"
-#include "testing/gtest/include/gtest/gtest.h"
-
-#include "webrtc/test/channel_transport/channel_transport.h"
-#include "webrtc/voice_engine/include/voe_audio_processing.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/voice_engine/include/voe_codec.h"
-#include "webrtc/voice_engine/include/voe_hardware.h"
-#include "webrtc/voice_engine/include/voe_network.h"
-
-DEFINE_string(render, "render", "render device name");
-DEFINE_string(codec, "ISAC", "codec name");
-DEFINE_int32(rate, 16000, "codec sample rate in Hz");
-
-namespace webrtc {
-namespace test {
-
-void RunHarness() {
- VoiceEngine* voe = VoiceEngine::Create();
- ASSERT_TRUE(voe != NULL);
- VoEAudioProcessing* audio = VoEAudioProcessing::GetInterface(voe);
- ASSERT_TRUE(audio != NULL);
- VoEBase* base = VoEBase::GetInterface(voe);
- ASSERT_TRUE(base != NULL);
- VoECodec* codec = VoECodec::GetInterface(voe);
- ASSERT_TRUE(codec != NULL);
- VoEHardware* hardware = VoEHardware::GetInterface(voe);
- ASSERT_TRUE(hardware != NULL);
- VoENetwork* network = VoENetwork::GetInterface(voe);
- ASSERT_TRUE(network != NULL);
-
- ASSERT_EQ(0, base->Init());
- int channel = base->CreateChannel();
- ASSERT_NE(-1, channel);
-
- std::unique_ptr<VoiceChannelTransport> voice_channel_transport(
- new VoiceChannelTransport(network, channel));
-
- ASSERT_EQ(0, voice_channel_transport->SetSendDestination("127.0.0.1", 1234));
- ASSERT_EQ(0, voice_channel_transport->SetLocalReceiver(1234));
-
- CodecInst codec_params = {0};
- bool codec_found = false;
- for (int i = 0; i < codec->NumOfCodecs(); i++) {
- ASSERT_EQ(0, codec->GetCodec(i, codec_params));
- if (FLAGS_codec.compare(codec_params.plname) == 0 &&
- FLAGS_rate == codec_params.plfreq) {
- codec_found = true;
- break;
- }
- }
- ASSERT_TRUE(codec_found);
- ASSERT_EQ(0, codec->SetSendCodec(channel, codec_params));
-
- int num_devices = 0;
- ASSERT_EQ(0, hardware->GetNumOfPlayoutDevices(num_devices));
- char device_name[128] = {0};
- char guid[128] = {0};
- bool device_found = false;
- int device_index;
- for (device_index = 0; device_index < num_devices; device_index++) {
- ASSERT_EQ(0, hardware->GetPlayoutDeviceName(device_index, device_name,
- guid));
- if (FLAGS_render.compare(device_name) == 0) {
- device_found = true;
- break;
- }
- }
- ASSERT_TRUE(device_found);
- ASSERT_EQ(0, hardware->SetPlayoutDevice(device_index));
-
- // Disable all audio processing.
- ASSERT_EQ(0, audio->SetAgcStatus(false));
- ASSERT_EQ(0, audio->SetEcStatus(false));
- ASSERT_EQ(0, audio->EnableHighPassFilter(false));
- ASSERT_EQ(0, audio->SetNsStatus(false));
-
- ASSERT_EQ(0, base->StartReceive(channel));
- ASSERT_EQ(0, base->StartPlayout(channel));
- ASSERT_EQ(0, base->StartSend(channel));
-
- // Run forever...
- while (1) {
- }
-}
-
-} // namespace test
-} // namespace webrtc
-
-int main(int argc, char** argv) {
- google::ParseCommandLineFlags(&argc, &argv, true);
- webrtc::test::RunHarness();
-}
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