mirror of
https://github.com/mollyim/webrtc.git
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Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ )
Reason for revert:
Suspect of breaking Chrome FYI bots.
See
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder
Example logs:
../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory
#include "third_party/webrtc/video_encoder.h"
^
Original issue's description:
> Move video_encoder.h and video_decoder.h to /api and create GN targets for them
>
> BUG=webrtc:5881
> # Because PRESUBMIT ignores LINT blacklist for moved files and these
> # headers have some not easy to resolve issues.
> NOPRESUBMIT=True
>
> Review-Url: https://codereview.webrtc.org/2780943003
> Cr-Commit-Position: refs/heads/master@{#17511}
> Committed: c42f540570
TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5881
Review-Url: https://codereview.webrtc.org/2794033002
Cr-Commit-Position: refs/heads/master@{#17514}
This commit is contained in:
parent
880c85ba7e
commit
716d7ac5c1
29 changed files with 61 additions and 100 deletions
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@ -16,8 +16,6 @@ import sys
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# Files and directories that are *skipped* by cpplint in the presubmit script.
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CPPLINT_BLACKLIST = [
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'tools-webrtc',
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'webrtc/api/video_codecs/video_decoder.h',
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'webrtc/api/video_codecs/video_encoder.h',
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'webrtc/base',
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'webrtc/examples/objc',
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'webrtc/media',
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@ -40,6 +38,8 @@ CPPLINT_BLACKLIST = [
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'webrtc/call.h',
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'webrtc/common_types.h',
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'webrtc/common_types.cc',
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'webrtc/video_decoder.h',
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'webrtc/video_encoder.h',
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'webrtc/video_send_stream.h',
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]
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@ -14,6 +14,8 @@ include_rules = [
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"+webrtc/config.h",
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"+webrtc/transport.h",
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"+webrtc/typedefs.h",
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"+webrtc/video_decoder.h",
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"+webrtc/video_encoder.h",
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"+webrtc/video_frame.h",
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"+webrtc/video_receive_stream.h",
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"+webrtc/video_send_stream.h",
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@ -1,28 +0,0 @@
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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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import("../../webrtc.gni")
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if (is_android) {
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import("//build/config/android/config.gni")
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import("//build/config/android/rules.gni")
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}
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rtc_source_set("video_codecs_api") {
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sources = [
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"video_decoder.h",
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"video_encoder.h",
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]
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deps = [
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# TODO(ilnik): Add dependency on webrtc/video_frame.h when it will have it's
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# own build target.
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"../..:webrtc_common",
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"../../base:rtc_base_approved",
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]
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}
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@ -226,7 +226,6 @@ rtc_static_library("rtc_media") {
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"../api:video_frame_api",
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"../api/audio_codecs:audio_codecs_api",
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"../api/audio_codecs:builtin_audio_decoder_factory",
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"../api/video_codecs:video_codecs_api",
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"../base:rtc_base",
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"../base:rtc_base_approved",
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"../call",
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@ -306,7 +305,6 @@ if (rtc_include_tests) {
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"..:webrtc_common",
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"../api:call_api",
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"../api:video_frame_api",
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"../api/video_codecs:video_codecs_api",
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"../base:rtc_base",
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"../base:rtc_base_approved",
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"../base:rtc_base_tests_main",
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@ -440,7 +438,6 @@ if (rtc_include_tests) {
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":rtc_unittest_main",
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"../api:video_frame_api",
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"../api/audio_codecs:builtin_audio_decoder_factory",
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"../api/video_codecs:video_codecs_api",
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"../audio",
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"../base:rtc_base",
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"../base:rtc_base_approved",
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@ -14,10 +14,7 @@
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#include <map>
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#include <set>
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#include <vector>
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#include <string>
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#include "webrtc/api/video_codecs/video_decoder.h"
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#include "webrtc/api/video_codecs/video_encoder.h"
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#include "webrtc/base/basictypes.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/gunit.h"
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@ -27,6 +24,8 @@
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#include "webrtc/media/engine/webrtcvideodecoderfactory.h"
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#include "webrtc/media/engine/webrtcvideoencoderfactory.h"
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#include "webrtc/modules/video_coding/include/video_error_codes.h"
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#include "webrtc/video_decoder.h"
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#include "webrtc/video_encoder.h"
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namespace cricket {
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static const int kEventTimeoutMs = 10000;
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@ -14,7 +14,7 @@
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#include <memory>
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#include <string>
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#include "webrtc/api/video_codecs/video_decoder.h"
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#include "webrtc/video_decoder.h"
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namespace webrtc {
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@ -8,7 +8,8 @@
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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 "webrtc/api/video_codecs/video_decoder.h"
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#include "webrtc/video_decoder.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/media/engine/videodecodersoftwarefallbackwrapper.h"
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#include "webrtc/modules/video_coding/include/video_error_codes.h"
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@ -15,8 +15,8 @@
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#include <string>
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#include <vector>
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#include "webrtc/api/video_codecs/video_encoder.h"
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#include "webrtc/media/base/codec.h"
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#include "webrtc/video_encoder.h"
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namespace webrtc {
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@ -17,8 +17,6 @@
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#include <utility>
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#include "webrtc/api/video/i420_buffer.h"
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#include "webrtc/api/video_codecs/video_decoder.h"
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#include "webrtc/api/video_codecs/video_encoder.h"
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#include "webrtc/base/copyonwritebuffer.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/base/stringutils.h"
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@ -37,6 +35,8 @@
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#include "webrtc/media/engine/webrtcvoiceengine.h"
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#include "webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h"
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#include "webrtc/system_wrappers/include/field_trial.h"
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#include "webrtc/video_decoder.h"
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#include "webrtc/video_encoder.h"
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using DegradationPreference = webrtc::VideoSendStream::DegradationPreference;
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@ -13,7 +13,6 @@
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#include <memory>
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#include <vector>
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#include "webrtc/api/video_codecs/video_encoder.h"
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#include "webrtc/base/arraysize.h"
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#include "webrtc/base/gunit.h"
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#include "webrtc/base/stringutils.h"
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@ -31,6 +30,7 @@
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#include "webrtc/media/engine/webrtcvideoengine2.h"
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#include "webrtc/media/engine/webrtcvoiceengine.h"
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#include "webrtc/test/field_trial.h"
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#include "webrtc/video_encoder.h"
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using webrtc::RtpExtension;
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@ -3803,6 +3803,7 @@ TEST_F(WebRtcVideoChannel2Test, RedRtxPacketDoesntCreateUnsignalledStream) {
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// The first unsignalled SSRC received will create a default receive stream.
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// Any different unsignalled SSRC received will replace the default.
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TEST_F(WebRtcVideoChannel2Test, ReceiveDifferentUnsignaledSsrc) {
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// Allow receiving VP8, VP9, H264 (if enabled).
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cricket::VideoRecvParameters parameters;
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parameters.codecs.push_back(GetEngineCodec("VP8"));
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@ -129,7 +129,6 @@ rtc_static_library("video_coding_utility") {
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deps = [
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"../..:webrtc_common",
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"../../api/video_codecs:video_codecs_api",
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"../../base:rtc_base_approved",
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"../../base:rtc_numerics",
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"../../base:rtc_task_queue",
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@ -225,7 +224,6 @@ rtc_static_library("webrtc_vp8") {
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deps = [
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":video_coding_utility",
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"../..:webrtc_common",
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"../../api/video_codecs:video_codecs_api",
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"../../base:rtc_base_approved",
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"../../common_video",
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"../../system_wrappers",
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@ -324,7 +322,6 @@ if (rtc_include_tests) {
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":video_coding_utility",
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":webrtc_vp8",
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"../..:webrtc_common",
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"../../api/video_codecs:video_codecs_api",
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"../../base:rtc_base_approved",
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"../../common_video:common_video",
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"../../system_wrappers:system_wrappers",
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@ -531,7 +528,6 @@ if (rtc_include_tests) {
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":webrtc_vp9",
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"../..:webrtc_common",
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"../../api:video_frame_api",
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"../../api/video_codecs:video_codecs_api",
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"../../base:rtc_base",
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"../../base:rtc_base_approved",
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"../../base:rtc_task_queue",
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@ -13,12 +13,12 @@
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#include <memory>
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#include "webrtc/api/video_codecs/video_decoder.h"
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#include "webrtc/api/video_codecs/video_encoder.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/event.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/test/gtest.h"
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#include "webrtc/video_decoder.h"
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#include "webrtc/video_encoder.h"
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namespace webrtc {
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@ -16,10 +16,10 @@
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#include <map>
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#include <memory>
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#include "webrtc/api/video_codecs/video_encoder.h"
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/common_video/include/video_bitrate_allocator.h"
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#include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
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#include "webrtc/video_encoder.h"
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namespace webrtc {
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@ -14,12 +14,12 @@
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#include <vector>
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#include "webrtc/api/video/video_frame.h"
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#include "webrtc/api/video_codecs/video_decoder.h"
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#include "webrtc/api/video_codecs/video_encoder.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/modules/video_coding/include/video_error_codes.h"
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#include "webrtc/typedefs.h"
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#include "webrtc/video_decoder.h"
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#include "webrtc/video_encoder.h"
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namespace webrtc {
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#include <utility>
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#include "webrtc/common_types.h"
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#include "webrtc/api/video_codecs/video_encoder.h"
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#include "webrtc/video_encoder.h"
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#include "webrtc/base/optional.h"
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#include "webrtc/base/sequenced_task_checker.h"
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#include "webrtc/modules/video_coding/utility/moving_average.h"
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@ -8,12 +8,12 @@
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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 "webrtc/api/video_codecs/video_encoder.h"
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#include "webrtc/common_video/include/video_bitrate_allocator.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
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#include "webrtc/modules/video_coding/include/video_codec_initializer.h"
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#include "webrtc/test/gtest.h"
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#include "webrtc/video_encoder.h"
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namespace webrtc {
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@ -145,7 +145,6 @@ rtc_static_library("libjingle_peerconnection") {
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":rtc_pc",
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"../api:call_api",
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"../api:rtc_stats_api",
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"../api/video_codecs:video_codecs_api",
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"../call",
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"../media",
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"../stats",
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@ -13,15 +13,15 @@
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#include "webrtc/sdk/android/src/jni/androidmediaencoder_jni.h"
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#include <algorithm>
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#include <list>
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#include <memory>
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#include <string>
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#include <utility>
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#include <list>
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#include "third_party/libyuv/include/libyuv/convert.h"
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#include "third_party/libyuv/include/libyuv/convert_from.h"
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#include "third_party/libyuv/include/libyuv/video_common.h"
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#include "webrtc/api/video_codecs/video_encoder.h"
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#include "webrtc/sdk/android/src/jni/androidmediacodeccommon.h"
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#include "webrtc/sdk/android/src/jni/classreferenceholder.h"
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#include "webrtc/sdk/android/src/jni/native_handle_impl.h"
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#include "webrtc/base/bind.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/logging.h"
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@ -38,11 +38,9 @@
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#include "webrtc/modules/video_coding/include/video_codec_interface.h"
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#include "webrtc/modules/video_coding/utility/quality_scaler.h"
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#include "webrtc/modules/video_coding/utility/vp8_header_parser.h"
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#include "webrtc/sdk/android/src/jni/androidmediacodeccommon.h"
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#include "webrtc/sdk/android/src/jni/classreferenceholder.h"
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#include "webrtc/sdk/android/src/jni/native_handle_impl.h"
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#include "webrtc/system_wrappers/include/field_trial.h"
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#include "webrtc/system_wrappers/include/logcat_trace_context.h"
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#include "webrtc/video_encoder.h"
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using rtc::Bind;
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using rtc::Thread;
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@ -125,7 +123,7 @@ class MediaCodecVideoEncoder : public webrtc::VideoEncoder {
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private:
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class EncodeTask : public rtc::QueuedTask {
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public:
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explicit EncodeTask(rtc::WeakPtr<MediaCodecVideoEncoder> encoder);
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EncodeTask(rtc::WeakPtr<MediaCodecVideoEncoder> encoder);
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bool Run() override;
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private:
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@ -236,7 +234,7 @@ class MediaCodecVideoEncoder : public webrtc::VideoEncoder {
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int64_t stat_start_time_ms_; // Start time for statistics.
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int current_frames_; // Number of frames in the current statistics interval.
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int current_bytes_; // Encoded bytes in the current statistics interval.
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int current_acc_qp_; // Accumulated QP in the current statistics interval.
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int current_acc_qp_; // Accumulated QP in the current statistics interval.
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int current_encoding_time_ms_; // Overall encoding time in the current second
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int64_t last_input_timestamp_ms_; // Timestamp of last received yuv frame.
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int64_t last_output_timestamp_ms_; // Timestamp of last encoded frame.
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@ -261,10 +259,10 @@ class MediaCodecVideoEncoder : public webrtc::VideoEncoder {
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const webrtc::VideoRotation rotation;
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};
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std::list<InputFrameInfo> input_frame_infos_;
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int32_t output_timestamp_; // Last output frame timestamp from
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// |input_frame_infos_|.
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int64_t output_render_time_ms_; // Last output frame render time from
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// |input_frame_infos_|.
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int32_t output_timestamp_; // Last output frame timestamp from
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// |input_frame_infos_|.
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int64_t output_render_time_ms_; // Last output frame render time from
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// |input_frame_infos_|.
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webrtc::VideoRotation output_rotation_; // Last output frame rotation from
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// |input_frame_infos_|.
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// Frame size in bytes fed to MediaCodec.
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@ -278,8 +276,8 @@ class MediaCodecVideoEncoder : public webrtc::VideoEncoder {
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webrtc::H264BitstreamParser h264_bitstream_parser_;
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// VP9 variables to populate codec specific structure.
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webrtc::GofInfoVP9 gof_; // Contains each frame's temporal information for
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// non-flexible VP9 mode.
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webrtc::GofInfoVP9 gof_; // Contains each frame's temporal information for
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// non-flexible VP9 mode.
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uint8_t tl0_pic_idx_;
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size_t gof_idx_;
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@ -521,9 +519,8 @@ int32_t MediaCodecVideoEncoder::InitEncodeInternal(int width,
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ScopedLocalRefFrame local_ref_frame(jni);
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const VideoCodecType codec_type = GetCodecType();
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ALOGD << "InitEncodeInternal Type: " << static_cast<int>(codec_type) << ", "
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<< width << " x " << height << ". Bitrate: " << kbps
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<< " kbps. Fps: " << fps;
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ALOGD << "InitEncodeInternal Type: " << (int)codec_type << ", " << width
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<< " x " << height << ". Bitrate: " << kbps << " kbps. Fps: " << fps;
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if (kbps == 0) {
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kbps = last_set_bitrate_kbps_;
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||||
}
|
||||
|
@ -553,10 +550,9 @@ int32_t MediaCodecVideoEncoder::InitEncodeInternal(int width,
|
|||
input_frame_infos_.clear();
|
||||
drop_next_input_frame_ = false;
|
||||
use_surface_ = use_surface;
|
||||
// TODO(ilnik): Use rand_r() instead to avoid LINT warnings below.
|
||||
picture_id_ = static_cast<uint16_t>(rand()) & 0x7FFF; // NOLINT
|
||||
picture_id_ = static_cast<uint16_t>(rand()) & 0x7FFF;
|
||||
gof_.SetGofInfoVP9(webrtc::TemporalStructureMode::kTemporalStructureMode1);
|
||||
tl0_pic_idx_ = static_cast<uint8_t>(rand()); // NOLINT
|
||||
tl0_pic_idx_ = static_cast<uint8_t>(rand());
|
||||
gof_idx_ = 0;
|
||||
last_frame_received_ms_ = -1;
|
||||
frames_received_since_last_key_ = kMinKeyFrameInterval;
|
||||
|
@ -680,7 +676,7 @@ int32_t MediaCodecVideoEncoder::Encode(
|
|||
}
|
||||
if (frames_encoded_ < kMaxEncodedLogFrames) {
|
||||
ALOGD << "Encoder frame in # " << (frames_received_ - 1)
|
||||
<< ". TS: " << static_cast<int>(current_timestamp_us_ / 1000)
|
||||
<< ". TS: " << (int)(current_timestamp_us_ / 1000)
|
||||
<< ". Q: " << input_frame_infos_.size() << ". Fps: " << last_set_fps_
|
||||
<< ". Kbps: " << last_set_bitrate_kbps_;
|
||||
}
|
||||
|
@ -700,7 +696,7 @@ int32_t MediaCodecVideoEncoder::Encode(
|
|||
if (input_frame_infos_.size() > MAX_ENCODER_Q_SIZE) {
|
||||
ALOGD << "Already " << input_frame_infos_.size()
|
||||
<< " frames in the queue, dropping"
|
||||
<< ". TS: " << static_cast<int>(current_timestamp_us_ / 1000)
|
||||
<< ". TS: " << (int)(current_timestamp_us_ / 1000)
|
||||
<< ". Fps: " << last_set_fps_
|
||||
<< ". Consecutive drops: " << consecutive_full_queue_frame_drops_;
|
||||
current_timestamp_us_ += rtc::kNumMicrosecsPerSec / last_set_fps_;
|
||||
|
@ -1143,13 +1139,14 @@ bool MediaCodecVideoEncoder::DeliverPendingOutputs(JNIEnv* jni) {
|
|||
frame_encoding_time_ms = rtc::TimeMillis() - encoding_start_time_ms;
|
||||
}
|
||||
if (frames_encoded_ < kMaxEncodedLogFrames) {
|
||||
int current_latency = static_cast<int>(last_input_timestamp_ms_ -
|
||||
last_output_timestamp_ms_);
|
||||
ALOGD << "Encoder frame out # " << frames_encoded_
|
||||
<< ". Key: " << key_frame << ". Size: " << payload_size
|
||||
<< ". TS: " << static_cast<int>(last_output_timestamp_ms_)
|
||||
<< ". Latency: " << current_latency
|
||||
<< ". EncTime: " << frame_encoding_time_ms;
|
||||
int current_latency =
|
||||
(int)(last_input_timestamp_ms_ - last_output_timestamp_ms_);
|
||||
ALOGD << "Encoder frame out # " << frames_encoded_ <<
|
||||
". Key: " << key_frame <<
|
||||
". Size: " << payload_size <<
|
||||
". TS: " << (int)last_output_timestamp_ms_ <<
|
||||
". Latency: " << current_latency <<
|
||||
". EncTime: " << frame_encoding_time_ms;
|
||||
}
|
||||
|
||||
// Calculate and print encoding statistics - every 3 seconds.
|
||||
|
|
|
@ -403,7 +403,6 @@ rtc_source_set("test_common") {
|
|||
":test_support",
|
||||
":video_test_common",
|
||||
"..:webrtc_common",
|
||||
"../api/video_codecs:video_codecs_api",
|
||||
"../audio",
|
||||
"../base:rtc_base_approved",
|
||||
"../call",
|
||||
|
|
|
@ -14,7 +14,7 @@
|
|||
#include <memory>
|
||||
#include <vector>
|
||||
|
||||
#include "webrtc/api/video_codecs/video_encoder.h"
|
||||
#include "webrtc/video_encoder.h"
|
||||
|
||||
namespace webrtc {
|
||||
namespace test {
|
||||
|
|
|
@ -14,12 +14,12 @@
|
|||
#include <vector>
|
||||
#include <memory>
|
||||
|
||||
#include "webrtc/api/video_codecs/video_encoder.h"
|
||||
#include "webrtc/base/criticalsection.h"
|
||||
#include "webrtc/base/sequenced_task_checker.h"
|
||||
#include "webrtc/base/task_queue.h"
|
||||
#include "webrtc/common_types.h"
|
||||
#include "webrtc/system_wrappers/include/clock.h"
|
||||
#include "webrtc/video_encoder.h"
|
||||
|
||||
namespace webrtc {
|
||||
namespace test {
|
||||
|
|
|
@ -58,7 +58,6 @@ rtc_static_library("video") {
|
|||
deps = [
|
||||
"..:webrtc_common",
|
||||
"../api:transport_api",
|
||||
"../api/video_codecs:video_codecs_api",
|
||||
"../base:rtc_base_approved",
|
||||
"../base:rtc_numerics",
|
||||
"../base:rtc_task_queue",
|
||||
|
@ -169,7 +168,6 @@ if (rtc_include_tests) {
|
|||
"replay.cc",
|
||||
]
|
||||
deps = [
|
||||
"../api/video_codecs:video_codecs_api",
|
||||
"../system_wrappers:metrics_default",
|
||||
"../test:field_trial",
|
||||
"../test:run_test",
|
||||
|
|
|
@ -15,7 +15,6 @@
|
|||
#include <string>
|
||||
#include <vector>
|
||||
|
||||
#include "webrtc/api/video_codecs/video_encoder.h"
|
||||
#include "webrtc/base/checks.h"
|
||||
#include "webrtc/base/event.h"
|
||||
#include "webrtc/base/file.h"
|
||||
|
@ -55,6 +54,7 @@
|
|||
#include "webrtc/test/testsupport/fileutils.h"
|
||||
#include "webrtc/test/testsupport/perf_test.h"
|
||||
#include "webrtc/video/transport_adapter.h"
|
||||
#include "webrtc/video_encoder.h"
|
||||
|
||||
#if defined(MEMORY_SANITIZER)
|
||||
// Flaky under MemorySanitizer, see
|
||||
|
|
|
@ -13,12 +13,12 @@
|
|||
|
||||
#include <vector>
|
||||
|
||||
#include "webrtc/api/video_codecs/video_encoder.h"
|
||||
#include "webrtc/base/constructormagic.h"
|
||||
#include "webrtc/base/criticalsection.h"
|
||||
#include "webrtc/base/thread_annotations.h"
|
||||
#include "webrtc/common_types.h"
|
||||
#include "webrtc/config.h"
|
||||
#include "webrtc/video_encoder.h"
|
||||
#include "webrtc/system_wrappers/include/atomic32.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
|
|
@ -15,7 +15,6 @@
|
|||
#include <sstream>
|
||||
|
||||
#include "gflags/gflags.h"
|
||||
#include "webrtc/api/video_codecs/video_decoder.h"
|
||||
#include "webrtc/base/checks.h"
|
||||
#include "webrtc/call/call.h"
|
||||
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
|
||||
|
@ -33,6 +32,7 @@
|
|||
#include "webrtc/test/video_capturer.h"
|
||||
#include "webrtc/test/video_renderer.h"
|
||||
#include "webrtc/typedefs.h"
|
||||
#include "webrtc/video_decoder.h"
|
||||
|
||||
namespace webrtc {
|
||||
namespace flags {
|
||||
|
|
|
@ -13,7 +13,6 @@
|
|||
#include "webrtc/test/gtest.h"
|
||||
#include "webrtc/test/gmock.h"
|
||||
|
||||
#include "webrtc/api/video_codecs/video_decoder.h"
|
||||
#include "webrtc/base/criticalsection.h"
|
||||
#include "webrtc/base/event.h"
|
||||
#include "webrtc/media/base/fakevideorenderer.h"
|
||||
|
@ -25,6 +24,7 @@
|
|||
#include "webrtc/system_wrappers/include/clock.h"
|
||||
#include "webrtc/system_wrappers/include/sleep.h"
|
||||
#include "webrtc/test/field_trial.h"
|
||||
#include "webrtc/video_decoder.h"
|
||||
|
||||
using testing::_;
|
||||
using testing::Invoke;
|
||||
|
|
|
@ -17,7 +17,6 @@
|
|||
#include <vector>
|
||||
|
||||
#include "webrtc/api/video/video_rotation.h"
|
||||
#include "webrtc/api/video_codecs/video_encoder.h"
|
||||
#include "webrtc/base/criticalsection.h"
|
||||
#include "webrtc/base/event.h"
|
||||
#include "webrtc/base/sequenced_task_checker.h"
|
||||
|
@ -32,6 +31,7 @@
|
|||
#include "webrtc/system_wrappers/include/atomic32.h"
|
||||
#include "webrtc/typedefs.h"
|
||||
#include "webrtc/video/overuse_frame_detector.h"
|
||||
#include "webrtc/video_encoder.h"
|
||||
#include "webrtc/video_send_stream.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
|
|
@ -8,8 +8,8 @@
|
|||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#ifndef WEBRTC_API_VIDEO_CODECS_VIDEO_DECODER_H_
|
||||
#define WEBRTC_API_VIDEO_CODECS_VIDEO_DECODER_H_
|
||||
#ifndef WEBRTC_VIDEO_DECODER_H_
|
||||
#define WEBRTC_VIDEO_DECODER_H_
|
||||
|
||||
#include <memory>
|
||||
#include <string>
|
||||
|
@ -83,4 +83,4 @@ class VideoDecoder {
|
|||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // WEBRTC_API_VIDEO_CODECS_VIDEO_DECODER_H_
|
||||
#endif // WEBRTC_VIDEO_DECODER_H_
|
|
@ -8,8 +8,8 @@
|
|||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#ifndef WEBRTC_API_VIDEO_CODECS_VIDEO_ENCODER_H_
|
||||
#define WEBRTC_API_VIDEO_CODECS_VIDEO_ENCODER_H_
|
||||
#ifndef WEBRTC_VIDEO_ENCODER_H_
|
||||
#define WEBRTC_VIDEO_ENCODER_H_
|
||||
|
||||
#include <memory>
|
||||
#include <string>
|
||||
|
@ -189,4 +189,4 @@ class VideoEncoder {
|
|||
};
|
||||
|
||||
} // namespace webrtc
|
||||
#endif // WEBRTC_API_VIDEO_CODECS_VIDEO_ENCODER_H_
|
||||
#endif // WEBRTC_VIDEO_ENCODER_H_
|
Loading…
Reference in a new issue