Remove use of VoECodec in video/call tests.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2447723002
Cr-Commit-Position: refs/heads/master@{#14797}
This commit is contained in:
solenberg
2016-10-27 00:23:06 -07:00
committed by Commit bot
parent 5e49c2f09e
commit 68e6bdd970
4 changed files with 8 additions and 31 deletions

View File

@ -40,9 +40,6 @@
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/perf_test.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_codec.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/include/voe_video_sync.h"
using webrtc::test::DriftingClock;
using webrtc::test::FakeAudioDevice;
@ -152,7 +149,6 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
metrics::Reset();
VoiceEngine* voice_engine = VoiceEngine::Create();
VoEBase* voe_base = VoEBase::GetInterface(voice_engine);
VoECodec* voe_codec = VoECodec::GetInterface(voice_engine);
const std::string audio_filename =
test::ResourcePath("voice_engine/audio_long16", "pcm");
ASSERT_STRNE("", audio_filename.c_str());
@ -226,12 +222,11 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
AudioSendStream::Config audio_send_config(&audio_send_transport);
audio_send_config.voe_channel_id = send_channel_id;
audio_send_config.rtp.ssrc = kAudioSendSsrc;
audio_send_config.send_codec_spec.codec_inst =
CodecInst{103, "ISAC", 16000, 480, 1, 32000};
AudioSendStream* audio_send_stream =
sender_call_->CreateAudioSendStream(audio_send_config);
CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
EXPECT_EQ(0, voe_codec->SetSendCodec(send_channel_id, isac));
video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
if (fec == FecMode::kOn) {
video_send_config_.rtp.ulpfec.red_payload_type = kRedPayloadType;
@ -297,7 +292,6 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
voe_base->DeleteChannel(send_channel_id);
voe_base->DeleteChannel(recv_channel_id);
voe_base->Release();
voe_codec->Release();
DestroyCalls();