Chromium Code Reviews| Index: webrtc/voice_engine/test/auto_test/standard/dtmf_test.cc |
| diff --git a/webrtc/voice_engine/test/auto_test/standard/dtmf_test.cc b/webrtc/voice_engine/test/auto_test/standard/dtmf_test.cc |
| deleted file mode 100644 |
| index 2e6c73311dca0701e32244734b20b3933617bec9..0000000000000000000000000000000000000000 |
| --- a/webrtc/voice_engine/test/auto_test/standard/dtmf_test.cc |
| +++ /dev/null |
| @@ -1,66 +0,0 @@ |
| -/* |
| - * Copyright (c) 2011 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. |
| - */ |
| - |
| -#include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h" |
| -#include "webrtc/voice_engine/voice_engine_defines.h" |
| - |
| -class DtmfTest : public AfterStreamingFixture { |
| - protected: |
| - void RunSixteenDtmfEvents() { |
| - TEST_LOG("Sending telephone events:\n"); |
| - for (int i = 0; i < 16; i++) { |
| - TEST_LOG("%d ", i); |
| - TEST_LOG_FLUSH; |
| - EXPECT_TRUE(channel_proxy_->SendTelephoneEventOutband(i, 160)); |
| - Sleep(500); |
| - } |
| - TEST_LOG("\n"); |
| - } |
| -}; |
| - |
| -TEST_F(DtmfTest, ManualSuccessfullySendsOutOfBandTelephoneEvents) { |
| - RunSixteenDtmfEvents(); |
| -} |
| - |
| -TEST_F(DtmfTest, TestTwoNonDtmfEvents) { |
|
ossu
2017/09/14 11:51:57
Is this no longer supported? Should it be tested f
the sun
2017/09/14 12:36:39
Anything >16 counts as "non-DTMF", so in practice
ossu
2017/09/14 12:51:32
Ah, alright.
|
| - EXPECT_TRUE(channel_proxy_->SendTelephoneEventOutband(32, 160)); |
| - EXPECT_TRUE(channel_proxy_->SendTelephoneEventOutband(110, 160)); |
| -} |
| - |
| -// This test modifies the DTMF payload type from the default 106 to 88 |
| -// and then runs through 16 DTMF out.of-band events. |
| -TEST_F(DtmfTest, ManualCanChangeDtmfPayloadType) { |
| - webrtc::CodecInst codec_instance = webrtc::CodecInst(); |
| - |
| - TEST_LOG("Changing DTMF payload type.\n"); |
| - |
| - // Start by modifying the receiving side. |
| - for (int i = 0; i < voe_codec_->NumOfCodecs(); i++) { |
| - EXPECT_EQ(0, voe_codec_->GetCodec(i, codec_instance)); |
| - if (!STR_CASE_CMP("telephone-event", codec_instance.plname)) { |
| - codec_instance.pltype = 88; // Use 88 instead of default 106. |
| - EXPECT_EQ(0, voe_base_->StopSend(channel_)); |
| - EXPECT_EQ(0, voe_base_->StopPlayout(channel_)); |
| - EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance)); |
| - EXPECT_EQ(0, voe_base_->StartPlayout(channel_)); |
| - EXPECT_EQ(0, voe_base_->StartSend(channel_)); |
| - break; |
| - } |
| - } |
| - |
| - Sleep(500); |
| - |
| - // Next, we must modify the sending side as well. |
| - EXPECT_TRUE( |
| - channel_proxy_->SetSendTelephoneEventPayloadType(codec_instance.pltype, |
| - codec_instance.plfreq)); |
| - |
| - RunSixteenDtmfEvents(); |
| -} |