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 <qiangchen@chromium.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: George Zhou <gyzhou@chromium.org>
> 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 <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21396}

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",

examples/DEPS

@@ -4,7 +4,6 @@ include_rules = [
   "+media",
   "+modules/audio_device",
   "+modules/video_capture",
-  "+modules/audio_processing",
   "+p2p",
   "+pc",
   "+third_party/libyuv",

examples/unityplugin/OWNERS

@@ -1,2 +1 @@
 gyzhou@chromium.org
-qiangchen@chromium.org

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(),
+        webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, nullptr);
-        std::unique_ptr<webrtc::VideoEncoderFactory>(
-            new webrtc::StereoEncoderFactory(
-                rtc::MakeUnique<webrtc::InternalEncoderFactory>())),
-        std::unique_ptr<webrtc::VideoDecoderFactory>(
-            new webrtc::StereoDecoderFactory(
-                rtc::MakeUnique<webrtc::InternalDecoderFactory>())),
-        nullptr, nullptr);
   }
   if (!g_peer_connection_factory.get()) {
     DeletePeerConnection();

examples/unityplugin/unity_plugin_apis.cc

@@ -24,13 +24,12 @@ static std::map<int, rtc::scoped_refptr<SimplePeerConnection>>
 int CreatePeerConnection(const char** turn_urls,
                          const int no_of_urls,
                          const char* username,
-                         const char* credential,
+                         const char* credential) {
-                         bool mandatory_receive_video) {
   g_peer_connection_map[g_peer_connection_id] =
       new rtc::RefCountedObject<SimplePeerConnection>();
 
   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++;

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,
+                                           const char* credential);
-                                           bool mandatory_receive_video);
 // 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

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<std::mutex> lock(mutex);
-  if (!OnI420FrameReady)
+  rtc::scoped_refptr<webrtc::PlanarYuvBuffer> buffer(
-    return;
+      frame.video_frame_buffer()->ToI420());
-
+  if (OnI420FrameReady) {
-  rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer(
+    OnI420FrameReady(buffer->DataY(), buffer->DataU(), buffer->DataV(),
-      frame.video_frame_buffer());
+                     buffer->StrideY(), buffer->StrideU(), buffer->StrideV(),
-
-  if (buffer->type() != webrtc::VideoFrameBuffer::Type::kI420A) {
-    rtc::scoped_refptr<webrtc::I420BufferInterface> 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(),
                      frame.width(), frame.height());
   }
 }