Reland of Adding audio to video_quality_test.
The original commit was https://codereview.webrtc.org/2136573002/.
BUG=
Review-Url: https://codereview.webrtc.org/2259783002
Cr-Commit-Position: refs/heads/master@{#13818}
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index a401d6d..51160c9 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -34,15 +34,68 @@
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/video_renderer.h"
#include "webrtc/video/video_quality_test.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
+
+namespace {
+
+constexpr int kSendStatsPollingIntervalMs = 1000;
+constexpr int kPayloadTypeH264 = 122;
+constexpr int kPayloadTypeVP8 = 123;
+constexpr int kPayloadTypeVP9 = 124;
+constexpr size_t kMaxComparisons = 10;
+constexpr char kSyncGroup[] = "av_sync";
+constexpr int kOpusMinBitrate = 6000;
+constexpr int kOpusBitrateFb = 32000;
+
+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;
+};
+
+void CreateVoiceEngine(VoiceEngineState* voe,
+ rtc::scoped_refptr<webrtc::AudioDecoderFactory>
+ 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::Config voe_config;
+ voe_config.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
+ voe->send_channel_id = voe->base->CreateChannel(voe_config);
+ EXPECT_GE(voe->send_channel_id, 0);
+ voe->receive_channel_id = voe->base->CreateChannel();
+ EXPECT_GE(voe->receive_channel_id, 0);
+}
+
+void DestroyVoiceEngine(VoiceEngineState* voe) {
+ voe->base->DeleteChannel(voe->send_channel_id);
+ voe->send_channel_id = -1;
+ voe->base->DeleteChannel(voe->receive_channel_id);
+ 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;
+}
+
+} // namespace
namespace webrtc {
-static const int kSendStatsPollingIntervalMs = 1000;
-static const int kPayloadTypeH264 = 122;
-static const int kPayloadTypeVP8 = 123;
-static const int kPayloadTypeVP9 = 124;
-static const size_t kMaxComparisons = 10;
-
class VideoAnalyzer : public PacketReceiver,
public Transport,
public rtc::VideoSinkInterface<VideoFrame>,
@@ -1002,6 +1055,7 @@
void VideoQualityTest::RunWithAnalyzer(const Params& params) {
params_ = params;
+ RTC_CHECK(!params_.audio);
// TODO(ivica): Merge with RunWithRenderer and use a flag / argument to
// differentiate between the analyzer and the renderer case.
CheckParams();
@@ -1099,7 +1153,7 @@
fclose(graph_data_output_file);
}
-void VideoQualityTest::RunWithVideoRenderer(const Params& params) {
+void VideoQualityTest::RunWithRenderers(const Params& params) {
params_ = params;
CheckParams();
@@ -1123,6 +1177,15 @@
// match the full stack tests.
Call::Config call_config;
call_config.bitrate_config = params_.common.call_bitrate_config;
+
+ ::VoiceEngineState voe;
+ if (params_.audio) {
+ CreateVoiceEngine(&voe, decoder_factory_);
+ AudioState::Config audio_state_config;
+ audio_state_config.voice_engine = voe.voice_engine;
+ call_config.audio_state = AudioState::Create(audio_state_config);
+ }
+
std::unique_ptr<Call> call(Call::Create(call_config));
test::LayerFilteringTransport transport(
@@ -1137,6 +1200,8 @@
video_send_config_.local_renderer = local_preview.get();
video_receive_configs_[stream_id].renderer = loopback_video.get();
+ if (params_.audio && params_.audio_video_sync)
+ video_receive_configs_[stream_id].sync_group = kSyncGroup;
video_send_config_.suspend_below_min_bitrate =
params_.common.suspend_below_min_bitrate;
@@ -1155,24 +1220,91 @@
video_send_stream_ =
call->CreateVideoSendStream(video_send_config_, video_encoder_config_);
- VideoReceiveStream* receive_stream =
+ VideoReceiveStream* video_receive_stream =
call->CreateVideoReceiveStream(video_receive_configs_[stream_id].Copy());
CreateCapturer(video_send_stream_->Input());
- receive_stream->Start();
+ AudioReceiveStream* audio_receive_stream = nullptr;
+ if (params_.audio) {
+ audio_send_config_ = AudioSendStream::Config(&transport);
+ audio_send_config_.voe_channel_id = voe.send_channel_id;
+ audio_send_config_.rtp.ssrc = kAudioSendSsrc;
+
+ // Add extension to enable audio send side BWE, and allow audio bit rate
+ // adaptation.
+ audio_send_config_.rtp.extensions.clear();
+ if (params_.common.send_side_bwe) {
+ audio_send_config_.rtp.extensions.push_back(webrtc::RtpExtension(
+ webrtc::RtpExtension::kTransportSequenceNumberUri,
+ test::kTransportSequenceNumberExtensionId));
+ audio_send_config_.min_bitrate_kbps = kOpusMinBitrate / 1000;
+ audio_send_config_.max_bitrate_kbps = kOpusBitrateFb / 1000;
+ }
+
+ audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_);
+
+ AudioReceiveStream::Config audio_config;
+ audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
+ audio_config.rtcp_send_transport = &transport;
+ audio_config.voe_channel_id = voe.receive_channel_id;
+ audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
+ audio_config.rtp.transport_cc = params_.common.send_side_bwe;
+ audio_config.rtp.extensions = audio_send_config_.rtp.extensions;
+ audio_config.decoder_factory = decoder_factory_;
+ if (params_.audio_video_sync)
+ 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));
+ }
+
+ // Start sending and receiving video.
+ video_receive_stream->Start();
video_send_stream_->Start();
capturer_->Start();
+ if (params_.audio) {
+ // Start receiving audio.
+ audio_receive_stream->Start();
+ EXPECT_EQ(0, voe.base->StartPlayout(voe.receive_channel_id));
+ EXPECT_EQ(0, voe.base->StartReceive(voe.receive_channel_id));
+
+ // Start sending audio.
+ audio_send_stream_->Start();
+ EXPECT_EQ(0, voe.base->StartSend(voe.send_channel_id));
+ }
+
test::PressEnterToContinue();
+ if (params_.audio) {
+ // Stop sending audio.
+ EXPECT_EQ(0, voe.base->StopSend(voe.send_channel_id));
+ audio_send_stream_->Stop();
+
+ // Stop receiving audio.
+ EXPECT_EQ(0, voe.base->StopReceive(voe.receive_channel_id));
+ EXPECT_EQ(0, voe.base->StopPlayout(voe.receive_channel_id));
+ audio_receive_stream->Stop();
+ }
+
+ // Stop receiving and sending video.
capturer_->Stop();
video_send_stream_->Stop();
- receive_stream->Stop();
+ video_receive_stream->Stop();
- call->DestroyVideoReceiveStream(receive_stream);
+ call->DestroyVideoReceiveStream(video_receive_stream);
call->DestroyVideoSendStream(video_send_stream_);
+ if (params_.audio) {
+ call->DestroyAudioSendStream(audio_send_stream_);
+ call->DestroyAudioReceiveStream(audio_receive_stream);
+ }
+
transport.StopSending();
+ if (params_.audio)
+ DestroyVoiceEngine(&voe);
}
} // namespace webrtc
