This reverts commit 2666cf7eba4bdd697d59d0451a8f74a05d4d207e.
Reason for revert: On Lollipop Nexus 4, the 240p tests fail too.
Original change's description:
> Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld).
>
> * Add audio_ prefix to CallTest::{en,de}coder_factory_.
> * Let VideoQualityTest only instantiate encoders using encoder factories.
> * Add HW encoder factories to VideoQualityTest.
> * Add full stack tests:
> - sqcif7 at 30 kbps: libvpx.
> - 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx.
>
> BUG=webrtc:8219
> TBR=asapersson@webrtc.org,kjellander@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
>
> Change-Id: I464409ac0d5276defa78c1bf66034c6cca717d74
> Reviewed-on: https://webrtc-review.googlesource.com/4740
> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20041}
TBR=kjellander@webrtc.org,brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,stefan@webrtc.org
Change-Id: If558b7fb86740658e50a6897d1eeeb72103a54ec
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8219
Reviewed-on: https://webrtc-review.googlesource.com/4900
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20044}
320 lines
9.7 KiB
Plaintext
320 lines
9.7 KiB
Plaintext
# Copyright (c) 2014 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.
|
|
|
|
import("../webrtc.gni")
|
|
|
|
rtc_static_library("video") {
|
|
sources = [
|
|
"call_stats.cc",
|
|
"call_stats.h",
|
|
"encoder_rtcp_feedback.cc",
|
|
"encoder_rtcp_feedback.h",
|
|
"overuse_frame_detector.cc",
|
|
"overuse_frame_detector.h",
|
|
"payload_router.cc",
|
|
"payload_router.h",
|
|
"quality_threshold.cc",
|
|
"quality_threshold.h",
|
|
"receive_statistics_proxy.cc",
|
|
"receive_statistics_proxy.h",
|
|
"report_block_stats.cc",
|
|
"report_block_stats.h",
|
|
"rtp_streams_synchronizer.cc",
|
|
"rtp_streams_synchronizer.h",
|
|
"rtp_video_stream_receiver.cc",
|
|
"rtp_video_stream_receiver.h",
|
|
"send_delay_stats.cc",
|
|
"send_delay_stats.h",
|
|
"send_statistics_proxy.cc",
|
|
"send_statistics_proxy.h",
|
|
"stats_counter.cc",
|
|
"stats_counter.h",
|
|
"stream_synchronization.cc",
|
|
"stream_synchronization.h",
|
|
"transport_adapter.cc",
|
|
"transport_adapter.h",
|
|
"video_receive_stream.cc",
|
|
"video_receive_stream.h",
|
|
"video_send_stream.cc",
|
|
"video_send_stream.h",
|
|
"video_stream_decoder.cc",
|
|
"video_stream_decoder.h",
|
|
"video_stream_encoder.cc",
|
|
"video_stream_encoder.h",
|
|
]
|
|
|
|
if (!build_with_chromium && is_clang) {
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
}
|
|
|
|
deps = [
|
|
"..:webrtc_common",
|
|
"../api:optional",
|
|
"../api:transport_api",
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../call:call_interfaces",
|
|
"../call:rtp_interfaces",
|
|
"../call:video_stream_api",
|
|
|
|
# For RtxReceiveStream.
|
|
"../call:rtp_receiver",
|
|
"../common_video",
|
|
"../logging:rtc_event_log_api",
|
|
"../media:rtc_media_base",
|
|
"../modules:module_api",
|
|
"../modules/bitrate_controller",
|
|
"../modules/congestion_controller",
|
|
"../modules/pacing",
|
|
"../modules/remote_bitrate_estimator",
|
|
"../modules/rtp_rtcp",
|
|
"../modules/utility",
|
|
"../modules/video_coding",
|
|
"../modules/video_coding:video_coding_utility",
|
|
"../modules/video_coding:webrtc_vp8",
|
|
"../modules/video_processing",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_numerics",
|
|
"../rtc_base:rtc_task_queue",
|
|
"../rtc_base:sequenced_task_checker",
|
|
"../rtc_base:weak_ptr",
|
|
"../system_wrappers",
|
|
"../voice_engine",
|
|
]
|
|
}
|
|
|
|
if (rtc_include_tests) {
|
|
rtc_source_set("video_quality_test") {
|
|
testonly = true
|
|
visibility = [ ":*" ] # Only targets in this file can depend on this.
|
|
sources = [
|
|
"video_quality_test.cc",
|
|
"video_quality_test.h",
|
|
]
|
|
deps = [
|
|
"../api:optional",
|
|
"../call:call_interfaces",
|
|
"../common_video",
|
|
"../logging:rtc_event_log_api",
|
|
"../media:rtc_media",
|
|
"../media:rtc_media_base",
|
|
"../modules/audio_mixer:audio_mixer_impl",
|
|
"../modules/rtp_rtcp",
|
|
"../modules/video_coding:webrtc_h264",
|
|
"../modules/video_coding:webrtc_vp8",
|
|
"../modules/video_coding:webrtc_vp9",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_base_tests_utils",
|
|
"../rtc_base:rtc_task_queue",
|
|
"../system_wrappers",
|
|
"../test:rtp_test_utils",
|
|
"../test:test_common",
|
|
"../test:test_renderer",
|
|
"../test:test_renderer",
|
|
"../test:test_support",
|
|
"../test:test_support_test_artifacts",
|
|
"../test:video_test_common",
|
|
"../test:video_test_common",
|
|
"../test:video_test_support",
|
|
"../voice_engine",
|
|
"//testing/gtest",
|
|
]
|
|
if (!build_with_chromium && is_clang) {
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
}
|
|
}
|
|
|
|
rtc_source_set("video_full_stack_tests") {
|
|
testonly = true
|
|
|
|
# Skip restricting visibility on mobile platforms since the tests on those
|
|
# gets additional generated targets which would require many lines here to
|
|
# cover (which would be confusing to read and hard to maintain).
|
|
if (!is_android && !is_ios) {
|
|
visibility = [ "..:webrtc_perf_tests" ]
|
|
}
|
|
sources = [
|
|
"full_stack_tests.cc",
|
|
]
|
|
deps = [
|
|
":video_quality_test",
|
|
"../modules/pacing:pacing",
|
|
"../test:field_trial",
|
|
"../test:test_common",
|
|
"../test:test_support",
|
|
"//testing/gtest",
|
|
]
|
|
if (!build_with_chromium && is_clang) {
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
}
|
|
if (rtc_use_h264) {
|
|
defines = [ "WEBRTC_USE_H264" ]
|
|
}
|
|
}
|
|
|
|
rtc_executable("video_loopback") {
|
|
testonly = true
|
|
sources = [
|
|
"video_loopback.cc",
|
|
]
|
|
deps = [
|
|
":video_quality_test",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../system_wrappers:metrics_default",
|
|
"../test:field_trial",
|
|
"../test:run_test",
|
|
"../test:run_test_interface",
|
|
"../test:test_common",
|
|
"../test:test_renderer",
|
|
"../test:test_support",
|
|
"//testing/gmock",
|
|
"//testing/gtest",
|
|
]
|
|
if (!build_with_chromium && is_clang) {
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
}
|
|
}
|
|
|
|
rtc_executable("screenshare_loopback") {
|
|
testonly = true
|
|
sources = [
|
|
"screenshare_loopback.cc",
|
|
]
|
|
|
|
deps = [
|
|
":video_quality_test",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../system_wrappers:metrics_default",
|
|
"../test:field_trial",
|
|
"../test:run_test",
|
|
"../test:run_test_interface",
|
|
"../test:test_common",
|
|
"../test:test_renderer",
|
|
"../test:test_support",
|
|
]
|
|
if (!build_with_chromium && is_clang) {
|
|
# Suppress warnings from Chrome's Clang plugins.
|
|
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
}
|
|
}
|
|
|
|
rtc_executable("video_replay") {
|
|
testonly = true
|
|
sources = [
|
|
"replay.cc",
|
|
]
|
|
deps = [
|
|
"..:webrtc_common",
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../call:call_interfaces",
|
|
"../common_video",
|
|
"../logging:rtc_event_log_api",
|
|
"../modules/rtp_rtcp",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../system_wrappers",
|
|
"../system_wrappers:metrics_default",
|
|
"../test:field_trial",
|
|
"../test:rtp_test_utils",
|
|
"../test:run_test",
|
|
"../test:run_test_interface",
|
|
"../test:test_common",
|
|
"../test:test_renderer",
|
|
"../test:test_support",
|
|
"../test:video_test_common",
|
|
"../test:video_test_support",
|
|
]
|
|
if (!build_with_chromium && is_clang) {
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
}
|
|
}
|
|
|
|
# TODO(pbos): Rename test suite.
|
|
rtc_source_set("video_tests") {
|
|
testonly = true
|
|
|
|
# Skip restricting visibility on mobile platforms since the tests on those
|
|
# gets additional generated targets which would require many lines here to
|
|
# cover (which would be confusing to read and hard to maintain).
|
|
if (!is_android && !is_ios) {
|
|
visibility = [ "..:video_engine_tests" ]
|
|
}
|
|
defines = []
|
|
sources = [
|
|
"call_stats_unittest.cc",
|
|
"encoder_rtcp_feedback_unittest.cc",
|
|
"end_to_end_tests.cc",
|
|
"overuse_frame_detector_unittest.cc",
|
|
"payload_router_unittest.cc",
|
|
"picture_id_tests.cc",
|
|
"quality_threshold_unittest.cc",
|
|
"receive_statistics_proxy_unittest.cc",
|
|
"report_block_stats_unittest.cc",
|
|
"rtp_video_stream_receiver_unittest.cc",
|
|
"send_delay_stats_unittest.cc",
|
|
"send_statistics_proxy_unittest.cc",
|
|
"stats_counter_unittest.cc",
|
|
"stream_synchronization_unittest.cc",
|
|
"video_receive_stream_unittest.cc",
|
|
"video_send_stream_tests.cc",
|
|
"video_stream_encoder_unittest.cc",
|
|
]
|
|
deps = [
|
|
":video",
|
|
"../api:optional",
|
|
"../api:video_frame_api",
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../call:call_interfaces",
|
|
"../call:mock_rtp_interfaces",
|
|
"../call:rtp_receiver",
|
|
"../call:rtp_sender",
|
|
"../call:video_stream_api",
|
|
"../common_video",
|
|
"../logging:rtc_event_log_api",
|
|
"../media:rtc_media",
|
|
"../media:rtc_media_base",
|
|
"../media:rtc_media_tests_utils",
|
|
"../modules:module_api",
|
|
"../modules/pacing",
|
|
"../modules/rtp_rtcp",
|
|
"../modules/rtp_rtcp:mock_rtp_rtcp",
|
|
"../modules/utility",
|
|
"../modules/video_coding",
|
|
"../modules/video_coding:video_coding_utility",
|
|
"../modules/video_coding:webrtc_h264",
|
|
"../modules/video_coding:webrtc_vp8",
|
|
"../modules/video_coding:webrtc_vp9",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_base_tests_utils",
|
|
"../system_wrappers",
|
|
"../system_wrappers:field_trial_default",
|
|
"../system_wrappers:metrics_api",
|
|
"../system_wrappers:metrics_default",
|
|
"../test:direct_transport",
|
|
"../test:field_trial",
|
|
"../test:rtp_test_utils",
|
|
"../test:test_common",
|
|
"../test:test_support",
|
|
"../test:video_test_common",
|
|
"//testing/gmock",
|
|
"//testing/gtest",
|
|
]
|
|
if (!build_with_chromium && is_clang) {
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
}
|
|
if (rtc_use_h264) {
|
|
defines += [ "WEBRTC_USE_H264" ]
|
|
}
|
|
}
|
|
}
|