blob: b997538d96b08aaad23894d8c4d7dd78287e9ae4 [file] [log] [blame]
* Copyright 2018 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 <stddef.h>
#include <stdint.h>
#include <memory>
#include <vector>
#include "api/video/video_frame.h"
#include "api/video/video_sink_interface.h"
#include "call/rtp_config.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "rtc_base/event.h"
#include "test/frame_generator_capturer.h"
#include "test/gtest.h"
#include "video/config/video_encoder_config.h"
#include "video/end_to_end_tests/multi_stream_tester.h"
namespace webrtc {
// Each renderer verifies that it receives the expected resolution, and as soon
// as every renderer has received a frame, the test finishes.
TEST(MultiStreamEndToEndTest, SendsAndReceivesMultipleStreams) {
class VideoOutputObserver : public rtc::VideoSinkInterface<VideoFrame> {
VideoOutputObserver(const MultiStreamTester::CodecSettings& settings,
uint32_t ssrc,
test::FrameGeneratorCapturer** frame_generator)
: settings_(settings), ssrc_(ssrc), frame_generator_(frame_generator) {}
void OnFrame(const VideoFrame& video_frame) override {
EXPECT_EQ(settings_.width, video_frame.width());
EXPECT_EQ(settings_.height, video_frame.height());
uint32_t Ssrc() { return ssrc_; }
bool Wait() { return done_.Wait(TimeDelta::Seconds(30)); }
const MultiStreamTester::CodecSettings& settings_;
const uint32_t ssrc_;
test::FrameGeneratorCapturer** const frame_generator_;
rtc::Event done_;
class Tester : public MultiStreamTester {
Tester() = default;
~Tester() override = default;
void Wait() override {
for (const auto& observer : observers_) {
<< "Time out waiting for from on ssrc " << observer->Ssrc();
void UpdateSendConfig(
size_t stream_index,
VideoSendStream::Config* send_config,
VideoEncoderConfig* encoder_config,
test::FrameGeneratorCapturer** frame_generator) override {
observers_[stream_index] = std::make_unique<VideoOutputObserver>(
codec_settings[stream_index], send_config->rtp.ssrcs.front(),
void UpdateReceiveConfig(
size_t stream_index,
VideoReceiveStreamInterface::Config* receive_config) override {
receive_config->renderer = observers_[stream_index].get();
std::unique_ptr<VideoOutputObserver> observers_[kNumStreams];
} tester;
} // namespace webrtc