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612f858ba0
Test enables single-nalu mode, sets limit for nalu lenght and verifies that encoder follows that limit. I found that QP jumps significantly when the mode is enabled. In result encoder might produce 4kbyte and 0.4kbyte frames back-to-back. But it seems that happens only to couple of frames in the beginning. This caused test to fail with default RC thresholds. To bypass this I increased frame size mismatch threshold from 20 to 30%. This should be Ok considering single-nalu mode is rare. BUG=webrtc:8070 Review-Url: https://codereview.webrtc.org/3014623002 Cr-Commit-Position: refs/heads/master@{#20023}
102 lines
3.6 KiB
C++
102 lines
3.6 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/video_coding/codecs/test/videoprocessor_integrationtest.h"
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#include <vector>
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#include "test/testsupport/fileutils.h"
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namespace webrtc {
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namespace test {
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#if defined(WEBRTC_USE_H264)
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namespace {
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// Codec settings.
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const bool kResilienceOn = true;
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const int kCifWidth = 352;
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const int kCifHeight = 288;
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const int kNumFrames = 100;
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const std::nullptr_t kNoVisualizationParams = nullptr;
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} // namespace
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class VideoProcessorIntegrationTestOpenH264
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: public VideoProcessorIntegrationTest {
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protected:
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VideoProcessorIntegrationTestOpenH264() {
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config_.filename = "foreman_cif";
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config_.input_filename = ResourcePath(config_.filename, "yuv");
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config_.output_filename =
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TempFilename(OutputPath(), "videoprocessor_integrationtest_libvpx");
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config_.networking_config.packet_loss_probability = 0.0;
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// Only allow encoder/decoder to use single core, for predictability.
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config_.use_single_core = true;
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config_.verbose = false;
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config_.hw_encoder = false;
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config_.hw_decoder = false;
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}
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};
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// H264: Run with no packet loss and fixed bitrate. Quality should be very high.
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// Note(hbos): The PacketManipulatorImpl code used to simulate packet loss in
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// these unittests appears to drop "packets" in a way that is not compatible
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// with H264. Therefore ProcessXPercentPacketLossH264, X != 0, unittests have
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// not been added.
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TEST_F(VideoProcessorIntegrationTestOpenH264, Process0PercentPacketLoss) {
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SetCodecSettings(&config_, kVideoCodecH264, 1, false, false, true, false,
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kResilienceOn, kCifWidth, kCifHeight);
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RateProfile rate_profile;
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SetRateProfile(&rate_profile, 0, 500, 30, 0);
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rate_profile.frame_index_rate_update[1] = kNumFrames + 1;
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rate_profile.num_frames = kNumFrames;
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std::vector<RateControlThresholds> rc_thresholds;
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AddRateControlThresholds(2, 60, 20, 10, 20, 0, 1, &rc_thresholds);
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QualityThresholds quality_thresholds(35.0, 25.0, 0.93, 0.70);
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ProcessFramesAndMaybeVerify(rate_profile, &rc_thresholds, &quality_thresholds,
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nullptr, kNoVisualizationParams);
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}
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// H264: Enable SingleNalUnit packetization mode. Encoder should split
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// large frames into multiple slices and limit length of NAL units.
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TEST_F(VideoProcessorIntegrationTestOpenH264, ProcessNoLossSingleNalUnit) {
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config_.packetization_mode = H264PacketizationMode::SingleNalUnit;
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config_.networking_config.max_payload_size_in_bytes = 500;
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SetCodecSettings(&config_, kVideoCodecH264, 1, false, false, true, false,
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kResilienceOn, kCifWidth, kCifHeight);
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RateProfile rate_profile;
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SetRateProfile(&rate_profile, 0, 500, 30, 0);
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rate_profile.frame_index_rate_update[1] = kNumFrames + 1;
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rate_profile.num_frames = kNumFrames;
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std::vector<RateControlThresholds> rc_thresholds;
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AddRateControlThresholds(2, 60, 30, 10, 20, 0, 1, &rc_thresholds);
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QualityThresholds quality_thresholds(35.0, 25.0, 0.93, 0.70);
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BitstreamThresholds bs_thresholds(
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config_.networking_config.max_payload_size_in_bytes);
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ProcessFramesAndMaybeVerify(rate_profile, &rc_thresholds, &quality_thresholds,
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&bs_thresholds, kNoVisualizationParams);
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}
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#endif // defined(WEBRTC_USE_H264)
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} // namespace test
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} // namespace webrtc
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