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Remove use of VoECodec in video/call tests.

Browse Source 
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
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10
webrtc/call/call_perf_tests.cc
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();

15
webrtc/test/call_test.cc
View File

@ -13,7 +13,6 @@
#include "webrtc/test/call_test.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_codec.h"
namespace webrtc {
namespace test {
@ -201,6 +200,8 @@ void CallTest::CreateSendConfig(size_t num_video_streams,
audio_send_config_ = AudioSendStream::Config(send_transport);
audio_send_config_.voe_channel_id = voe_send_.channel_id;
audio_send_config_.rtp.ssrc = kAudioSendSsrc;
audio_send_config_.send_codec_spec.codec_inst =
CodecInst{kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000};
}
}
@ -227,9 +228,9 @@ void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
}
}
RTC_DCHECK(num_audio_streams_ <= 1);
RTC_DCHECK_GE(1u, num_audio_streams_);
if (num_audio_streams_ == 1) {
RTC_DCHECK(voe_send_.channel_id >= 0);
RTC_DCHECK_LE(0, voe_send_.channel_id);
AudioReceiveStream::Config audio_config;
audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
audio_config.rtcp_send_transport = rtcp_send_transport;
@ -291,8 +292,6 @@ void CallTest::CreateAudioStreams() {
audio_receive_streams_.push_back(
receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i]));
}
CodecInst isac = {kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000};
EXPECT_EQ(0, voe_send_.codec->SetSendCodec(voe_send_.channel_id, isac));
}
void CallTest::DestroyStreams() {
@ -316,7 +315,6 @@ void CallTest::CreateVoiceEngines() {
CreateFakeAudioDevices();
voe_send_.voice_engine = VoiceEngine::Create();
voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
voe_send_.codec = VoECodec::GetInterface(voe_send_.voice_engine);
EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr,
decoder_factory_));
VoEBase::ChannelConfig config;
@ -326,7 +324,6 @@ void CallTest::CreateVoiceEngines() {
voe_recv_.voice_engine = VoiceEngine::Create();
voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine);
voe_recv_.codec = VoECodec::GetInterface(voe_recv_.voice_engine);
EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), nullptr,
decoder_factory_));
voe_recv_.channel_id = voe_recv_.base->CreateChannel();
@ -338,15 +335,11 @@ void CallTest::DestroyVoiceEngines() {
voe_recv_.channel_id = -1;
voe_recv_.base->Release();
voe_recv_.base = nullptr;
voe_recv_.codec->Release();
voe_recv_.codec = nullptr;
voe_send_.base->DeleteChannel(voe_send_.channel_id);
voe_send_.channel_id = -1;
voe_send_.base->Release();
voe_send_.base = nullptr;
voe_send_.codec->Release();
voe_send_.codec = nullptr;
VoiceEngine::Delete(voe_send_.voice_engine);
voe_send_.voice_engine = nullptr;

3
webrtc/test/call_test.h
View File

@ -26,7 +26,6 @@
namespace webrtc {
class VoEBase;
class VoECodec;
namespace test {
@ -123,12 +122,10 @@ class CallTest : public ::testing::Test {
VoiceEngineState()
: voice_engine(nullptr),
base(nullptr),
codec(nullptr),
channel_id(-1) {}
VoiceEngine* voice_engine;
VoEBase* base;
VoECodec* codec;
int channel_id;
};

11
webrtc/video/video_quality_test.cc
View File

@ -37,7 +37,6 @@
#include "webrtc/test/vcm_capturer.h"
#include "webrtc/test/video_renderer.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_codec.h"
namespace {
@ -54,13 +53,11 @@ struct VoiceEngineState {
VoiceEngineState()
: voice_engine(nullptr),
base(nullptr),
codec(nullptr),
send_channel_id(-1),
receive_channel_id(-1) {}
webrtc::VoiceEngine* voice_engine;
webrtc::VoEBase* base;
webrtc::VoECodec* codec;
int send_channel_id;
int receive_channel_id;
};
@ -70,7 +67,6 @@ void CreateVoiceEngine(VoiceEngineState* voe,
decoder_factory) {
voe->voice_engine = webrtc::VoiceEngine::Create();
voe->base = webrtc::VoEBase::GetInterface(voe->voice_engine);
voe->codec = webrtc::VoECodec::GetInterface(voe->voice_engine);
EXPECT_EQ(0, voe->base->Init(nullptr, nullptr, decoder_factory));
webrtc::VoEBase::ChannelConfig config;
config.enable_voice_pacing = true;
@ -87,8 +83,6 @@ void DestroyVoiceEngine(VoiceEngineState* voe) {
voe->receive_channel_id = -1;
voe->base->Release();
voe->base = nullptr;
voe->codec->Release();
voe->codec = nullptr;
webrtc::VoiceEngine::Delete(voe->voice_engine);
voe->voice_engine = nullptr;
@ -1341,6 +1335,8 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
audio_send_config_.min_bitrate_kbps = kOpusMinBitrate / 1000;
audio_send_config_.max_bitrate_kbps = kOpusBitrateFb / 1000;
}
audio_send_config_.send_codec_spec.codec_inst =
CodecInst{120, "OPUS", 48000, 960, 2, 64000};
audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_);
@ -1356,9 +1352,6 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
audio_config.sync_group = kSyncGroup;
audio_receive_stream = call->CreateAudioReceiveStream(audio_config);
const CodecInst kOpusInst = {120, "OPUS", 48000, 960, 2, 64000};
EXPECT_EQ(0, voe.codec->SetSendCodec(voe.send_channel_id, kOpusInst));
}
StartEncodedFrameLogs(video_receive_stream);