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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}
98 lines
3.6 KiB
C++
98 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/field_trial.h"
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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_ANDROID)
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namespace {
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const int kForemanNumFrames = 300;
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const std::nullptr_t kNoVisualizationParams = nullptr;
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} // namespace
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class VideoProcessorIntegrationTestMediaCodec
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: public VideoProcessorIntegrationTest {
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protected:
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VideoProcessorIntegrationTestMediaCodec() {
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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_mediacodec");
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config_.verbose = false;
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config_.hw_encoder = true;
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config_.hw_decoder = true;
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}
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};
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TEST_F(VideoProcessorIntegrationTestMediaCodec, ForemanCif500kbpsVp8) {
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SetCodecSettings(&config_, kVideoCodecVP8, 1, false, false, false, false,
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false, 352, 288);
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RateProfile rate_profile;
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SetRateProfile(&rate_profile, 0, 500, 30, 0); // Start below |low_kbps|.
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rate_profile.frame_index_rate_update[1] = kForemanNumFrames + 1;
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rate_profile.num_frames = kForemanNumFrames;
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// The thresholds below may have to be tweaked to let even poor MediaCodec
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// implementations pass. If this test fails on the bots, disable it and
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// ping brandtr@.
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std::vector<RateControlThresholds> rc_thresholds;
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AddRateControlThresholds(20, 95, 22, 11, 10, 0, 1, &rc_thresholds);
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QualityThresholds quality_thresholds(30.0, 14.0, 0.86, 0.39);
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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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TEST_F(VideoProcessorIntegrationTestMediaCodec,
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Foreman240p100kbpsVp8WithForcedSwFallback) {
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ScopedFieldTrials override_field_trials(
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"WebRTC-VP8-Forced-Fallback-Encoder/Enabled-150,175,10000,1/");
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config_.filename = "foreman_320x240";
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config_.input_filename = ResourcePath(config_.filename, "yuv");
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config_.sw_fallback_encoder = true;
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SetCodecSettings(&config_, kVideoCodecVP8, 1, false, false, false, false,
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false, 320, 240);
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RateProfile rate_profile;
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SetRateProfile(&rate_profile, 0, 100, 10, 0); // Start below |low_kbps|.
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SetRateProfile(&rate_profile, 1, 100, 10, 80); // Fallback in this bucket.
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SetRateProfile(&rate_profile, 2, 200, 10, 200); // Switch back here.
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rate_profile.frame_index_rate_update[3] = kForemanNumFrames + 1;
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rate_profile.num_frames = kForemanNumFrames;
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// The thresholds below may have to be tweaked to let even poor MediaCodec
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// implementations pass. If this test fails on the bots, disable it and
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// ping brandtr@.
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std::vector<RateControlThresholds> rc_thresholds;
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AddRateControlThresholds(0, 50, 75, 70, 10, 0, 1, &rc_thresholds);
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AddRateControlThresholds(0, 50, 25, 12, 60, 0, 1, &rc_thresholds);
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AddRateControlThresholds(0, 65, 15, 5, 5, 0, 1, &rc_thresholds);
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QualityThresholds quality_thresholds(33.0, 30.0, 0.90, 0.85);
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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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#endif // defined(WEBRTC_ANDROID)
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} // namespace test
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} // namespace webrtc
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