webrtc_m130/video/BUILD.gn
Sergey Silkin 86684960b3 Adding layering configurator and rate allocator for VP9 SVC.
The configurator decides number of spatial layers, their resolution
and bitrate thresholds based on given input resolution and maximum
number of spatial layers.

The allocator distributes available bitrate across spatial and
temporal layers. If there is not enough bitrate to provide acceptable
quality for all spatial layers allocator disables enhancement layers
one by one until the condition is met or number of layers is reduced
to one.

VP9 SVC related unit tests have been updated. Input resolution and
bitrate in these tests have been increased to the level enough to
provide desirable number of spatial layers.

Bug: webrtc:8518
Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b
Reviewed-on: https://webrtc-review.googlesource.com/60340
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22672}
2018-03-29 10:16:47 +00:00

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# 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",
"../:typedefs",
"../api:fec_controller_api",
"../api:libjingle_peerconnection_api",
"../api:optional",
"../api:transport_api",
"../api:video_frame_api",
"../api:video_frame_api_i420",
"../api/video_codecs:video_codecs_api",
"../call:bitrate_allocator",
"../call:call_interfaces",
"../call:rtp_interfaces",
"../call:video_stream_api",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/video_coding:codec_globals_headers",
"../modules/video_coding:video_codec_interface",
"../rtc_base:checks",
"../rtc_base:stringutils",
"../rtc_base/experiments:alr_experiment",
"../rtc_base/system:fallthrough",
"../system_wrappers:field_trial_api",
"../system_wrappers:metrics_api",
# For RtxReceiveStream.
"../call:rtp_receiver",
"../common_video",
"../logging:rtc_event_log_api",
"../modules:module_api",
"../modules/bitrate_controller",
"../modules/pacing",
"../modules/remote_bitrate_estimator",
"../modules/rtp_rtcp",
"../modules/utility",
"../modules/video_coding",
"../modules/video_coding:video_coding_utility",
"../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",
"../rtc_base/time:timestamp_extrapolator",
"../system_wrappers",
]
if (!build_with_mozilla) {
deps += [ "../media:rtc_media_base" ]
}
}
rtc_source_set("video_stream_decoder_impl") {
visibility = [ "*" ]
sources = [
"video_stream_decoder_impl.cc",
"video_stream_decoder_impl.h",
]
deps = [
"../api:encoded_frame_api",
"../api:optional",
"../api:video_frame_api",
"../api:video_stream_decoder",
"../api/video_codecs:video_codecs_api",
"../modules/video_coding:video_coding",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue_api",
"../system_wrappers:system_wrappers",
]
}
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 = [
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_output",
"../media:rtc_audio_video",
"../media:rtc_internal_video_codecs",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/rtp_rtcp",
"../modules/video_coding:webrtc_h264",
"../modules/video_coding:webrtc_multiplex",
"../modules/video_coding:webrtc_vp8",
"../modules/video_coding:webrtc_vp9",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../system_wrappers",
"../test:fileutils",
"../test:perf_test",
"../test:rtp_test_utils",
"../test:test_common",
"../test:test_renderer",
"../test:test_support",
"../test:test_support_test_artifacts",
"../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" ]
}
}
rtc_source_set("video_full_stack_tests") {
testonly = true
sources = [
"full_stack_tests.cc",
]
deps = [
":video_quality_test",
"../modules/pacing:pacing",
"../rtc_base/experiments:alr_experiment",
"../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:field_trial_default",
"../system_wrappers:metrics_default",
"../system_wrappers:runtime_enabled_features_default",
"../test:field_trial",
"../test:run_test",
"../test:run_test_interface",
"../test:test_common",
"../test:test_renderer",
"../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" ]
}
}
rtc_executable("screenshare_loopback") {
testonly = true
sources = [
"screenshare_loopback.cc",
]
deps = [
":video_quality_test",
"../rtc_base:rtc_base_approved",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
"../system_wrappers:runtime_enabled_features_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("sv_loopback") {
testonly = true
sources = [
"sv_loopback.cc",
]
deps = [
":video_quality_test",
"../rtc_base:rtc_base_approved",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
"../system_wrappers:runtime_enabled_features_default",
"../test:field_trial",
"../test:run_test",
"../test:run_test_interface",
"../test:test_common",
"../test:test_renderer",
"../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" ]
}
}
rtc_executable("video_replay") {
testonly = true
sources = [
"replay.cc",
]
deps = [
"..:webrtc_common",
"../:typedefs",
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",
"../common_video",
"../logging:rtc_event_log_api",
"../modules/rtp_rtcp",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
"../system_wrappers:metrics_default",
"../system_wrappers:runtime_enabled_features_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
defines = []
sources = [
"call_stats_unittest.cc",
"encoder_rtcp_feedback_unittest.cc",
"end_to_end_tests/bandwidth_tests.cc",
"end_to_end_tests/call_operation_tests.cc",
"end_to_end_tests/codec_tests.cc",
"end_to_end_tests/config_tests.cc",
"end_to_end_tests/extended_reports_tests.cc",
"end_to_end_tests/fec_tests.cc",
"end_to_end_tests/histogram_tests.cc",
"end_to_end_tests/log_tests.cc",
"end_to_end_tests/multi_stream_tester.cc",
"end_to_end_tests/multi_stream_tester.h",
"end_to_end_tests/multi_stream_tests.cc",
"end_to_end_tests/network_state_tests.cc",
"end_to_end_tests/probing_tests.cc",
"end_to_end_tests/retransmission_tests.cc",
"end_to_end_tests/rtp_rtcp_tests.cc",
"end_to_end_tests/ssrc_tests.cc",
"end_to_end_tests/stats_tests.cc",
"end_to_end_tests/transport_feedback_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_frame_api_i420",
"../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_audio_video",
"../media:rtc_internal_video_codecs",
"../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/rtp_rtcp:rtp_rtcp_format",
"../modules/utility",
"../modules/video_coding",
"../modules/video_coding:codec_globals_headers",
"../modules/video_coding:video_codec_interface",
"../modules/video_coding:video_coding_utility",
"../modules/video_coding:webrtc_h264",
"../modules/video_coding:webrtc_multiplex",
"../modules/video_coding:webrtc_vp8_helpers",
"../modules/video_coding:webrtc_vp9",
"../rtc_base:checks",
"../rtc_base:rate_limiter",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:rtc_numerics",
"../rtc_base:rtc_task_queue",
"../rtc_base/experiments:alr_experiment",
"../system_wrappers",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_api",
"../system_wrappers:metrics_default",
"../test:direct_transport",
"../test:field_trial",
"../test:fileutils",
"../test:perf_test",
"../test:rtp_test_utils",
"../test:test_common",
"../test:test_support",
"../test:video_test_common",
"//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" ]
}
if (!build_with_mozilla) {
deps += [ "../media:rtc_media_base" ]
}
}
}