From ec8410796af7f7f67cd2018e85aaefc160e4bfa3 Mon Sep 17 00:00:00 2001 From: Lu Liu Date: Wed, 20 Dec 2017 18:37:17 +0000 Subject: [PATCH] Revert "Add Alpha Channel Support For WebRTC Unity Plugin" This reverts commit 7ed2af5b461387191de2456cba906dd5d25766b6. Reason for revert: breaking buildbot Original change's description: > Add Alpha Channel Support For WebRTC Unity Plugin > > This CL make webrtc unity plugin compatible with alpha channel support. > > Bug: webrtc:8645 > Change-Id: I3250aede47b31c4685e57d11fb2b2e86b824f9c4 > Reviewed-on: https://webrtc-review.googlesource.com/32325 > Commit-Queue: Qiang Chen > Reviewed-by: Magnus Jedvert > Reviewed-by: George Zhou > Cr-Commit-Position: refs/heads/master@{#21394} TBR=magjed@webrtc.org,gyzhou@chromium.org,qiangchen@chromium.org Change-Id: I6994d7e87170f97216886a747548a988ca71b7d0 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8645 Reviewed-on: https://webrtc-review.googlesource.com/35420 Reviewed-by: Lu Liu Commit-Queue: Lu Liu Cr-Commit-Position: refs/heads/master@{#21396} --- examples/BUILD.gn | 5 ---- examples/DEPS | 1 - examples/unityplugin/OWNERS | 1 - .../unityplugin/simple_peer_connection.cc | 17 +----------- examples/unityplugin/unity_plugin_apis.cc | 5 ++-- examples/unityplugin/unity_plugin_apis.h | 5 +--- examples/unityplugin/video_observer.cc | 27 ++++--------------- 7 files changed, 9 insertions(+), 52 deletions(-) diff --git a/examples/BUILD.gn b/examples/BUILD.gn index 9785c17229..4e733d06bc 100644 --- a/examples/BUILD.gn +++ b/examples/BUILD.gn @@ -7,7 +7,6 @@ # be found in the AUTHORS file in the root of the source tree. import("../webrtc.gni") - if (is_android) { import("//build/config/android/config.gni") import("//build/config/android/rules.gni") @@ -653,7 +652,6 @@ if (is_win || is_android) { "unityplugin/classreferenceholder.h", "unityplugin/jni_onload.cc", ] - suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ] } if (!build_with_chromium && is_clang) { @@ -672,11 +670,8 @@ if (is_win || is_android) { "../api:video_frame_api", "../api/audio_codecs:builtin_audio_decoder_factory", "../api/audio_codecs:builtin_audio_encoder_factory", - "../media:rtc_internal_video_codecs", "../media:rtc_media", "../media:rtc_media_base", - "../modules/audio_device:audio_device", - "../modules/audio_processing:audio_processing", "../modules/video_capture:video_capture_module", "../pc:libjingle_peerconnection", "../rtc_base:rtc_base", diff --git a/examples/DEPS b/examples/DEPS index 96066528da..4b6aa075d0 100644 --- a/examples/DEPS +++ b/examples/DEPS @@ -4,7 +4,6 @@ include_rules = [ "+media", "+modules/audio_device", "+modules/video_capture", - "+modules/audio_processing", "+p2p", "+pc", "+third_party/libyuv", diff --git a/examples/unityplugin/OWNERS b/examples/unityplugin/OWNERS index 343f8600f1..61ea9a97f5 100644 --- a/examples/unityplugin/OWNERS +++ b/examples/unityplugin/OWNERS @@ -1,2 +1 @@ gyzhou@chromium.org -qiangchen@chromium.org diff --git a/examples/unityplugin/simple_peer_connection.cc b/examples/unityplugin/simple_peer_connection.cc index bffd5f6538..2ea8227b12 100644 --- a/examples/unityplugin/simple_peer_connection.cc +++ b/examples/unityplugin/simple_peer_connection.cc @@ -16,16 +16,8 @@ #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/test/fakeconstraints.h" #include "api/videosourceproxy.h" -#include "media/engine/stereocodecfactory.h" #include "media/engine/webrtcvideocapturerfactory.h" -#include "media/engine/webrtcvideodecoderfactory.h" -#include "media/engine/webrtcvideoencoderfactory.h" -#include "media/engine/internaldecoderfactory.h" -#include "media/engine/internalencoderfactory.h" -#include "modules/audio_device/include/audio_device.h" -#include "modules/audio_processing/include/audio_processing.h" #include "modules/video_capture/video_capture_factory.h" -#include "rtc_base/ptr_util.h" #if defined(WEBRTC_ANDROID) #include "examples/unityplugin/classreferenceholder.h" @@ -102,14 +94,7 @@ bool SimplePeerConnection::InitializePeerConnection(const char** turn_urls, g_peer_connection_factory = webrtc::CreatePeerConnectionFactory( g_worker_thread.get(), g_worker_thread.get(), g_signaling_thread.get(), nullptr, webrtc::CreateBuiltinAudioEncoderFactory(), - webrtc::CreateBuiltinAudioDecoderFactory(), - std::unique_ptr( - new webrtc::StereoEncoderFactory( - rtc::MakeUnique())), - std::unique_ptr( - new webrtc::StereoDecoderFactory( - rtc::MakeUnique())), - nullptr, nullptr); + webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, nullptr); } if (!g_peer_connection_factory.get()) { DeletePeerConnection(); diff --git a/examples/unityplugin/unity_plugin_apis.cc b/examples/unityplugin/unity_plugin_apis.cc index 34c28d926a..ae98a833f5 100644 --- a/examples/unityplugin/unity_plugin_apis.cc +++ b/examples/unityplugin/unity_plugin_apis.cc @@ -24,13 +24,12 @@ static std::map> int CreatePeerConnection(const char** turn_urls, const int no_of_urls, const char* username, - const char* credential, - bool mandatory_receive_video) { + const char* credential) { g_peer_connection_map[g_peer_connection_id] = new rtc::RefCountedObject(); if (!g_peer_connection_map[g_peer_connection_id]->InitializePeerConnection( - turn_urls, no_of_urls, username, credential, mandatory_receive_video)) + turn_urls, no_of_urls, username, credential, false)) return -1; return g_peer_connection_id++; diff --git a/examples/unityplugin/unity_plugin_apis.h b/examples/unityplugin/unity_plugin_apis.h index b32f9e2caf..814b9675fb 100644 --- a/examples/unityplugin/unity_plugin_apis.h +++ b/examples/unityplugin/unity_plugin_apis.h @@ -19,11 +19,9 @@ typedef void (*I420FRAMEREADY_CALLBACK)(const uint8_t* data_y, const uint8_t* data_u, const uint8_t* data_v, - const uint8_t* data_a, int stride_y, int stride_u, int stride_v, - int stride_a, uint32_t width, uint32_t height); typedef void (*LOCALDATACHANNELREADY_CALLBACK)(); @@ -49,8 +47,7 @@ extern "C" { WEBRTC_PLUGIN_API int CreatePeerConnection(const char** turn_urls, const int no_of_urls, const char* username, - const char* credential, - bool mandatory_receive_video); + const char* credential); // Close a peerconnection. WEBRTC_PLUGIN_API bool ClosePeerConnection(int peer_connection_id); // Add a audio stream. If audio_only is true, the stream only has an audio diff --git a/examples/unityplugin/video_observer.cc b/examples/unityplugin/video_observer.cc index a78ef57e41..821acd66aa 100644 --- a/examples/unityplugin/video_observer.cc +++ b/examples/unityplugin/video_observer.cc @@ -17,28 +17,11 @@ void VideoObserver::SetVideoCallback(I420FRAMEREADY_CALLBACK callback) { void VideoObserver::OnFrame(const webrtc::VideoFrame& frame) { std::unique_lock lock(mutex); - if (!OnI420FrameReady) - return; - - rtc::scoped_refptr buffer( - frame.video_frame_buffer()); - - if (buffer->type() != webrtc::VideoFrameBuffer::Type::kI420A) { - rtc::scoped_refptr i420_buffer = - buffer->ToI420(); - OnI420FrameReady(i420_buffer->DataY(), i420_buffer->DataU(), - i420_buffer->DataV(), nullptr, i420_buffer->StrideY(), - i420_buffer->StrideU(), i420_buffer->StrideV(), 0, - frame.width(), frame.height()); - - } else { - // The buffer has alpha channel. - webrtc::I420ABufferInterface* i420a_buffer = buffer->GetI420A(); - - OnI420FrameReady(i420a_buffer->DataY(), i420a_buffer->DataU(), - i420a_buffer->DataV(), i420a_buffer->DataA(), - i420a_buffer->StrideY(), i420a_buffer->StrideU(), - i420a_buffer->StrideV(), i420a_buffer->StrideA(), + rtc::scoped_refptr buffer( + frame.video_frame_buffer()->ToI420()); + if (OnI420FrameReady) { + OnI420FrameReady(buffer->DataY(), buffer->DataU(), buffer->DataV(), + buffer->StrideY(), buffer->StrideU(), buffer->StrideV(), frame.width(), frame.height()); } }