From f2bfc2b8ef3d774658b9ce3dcd6757f932d071fb Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Peter=20Bostr=C3=B6m?= Date: Thu, 17 Sep 2015 13:04:11 +0200 Subject: [PATCH] Remove some dead code. WebRtcPassthroughRender has been dead since webrtcvideoengine.cc was removed, FakeExternalTransport has probably been unused for a long time. BUG=webrtc:1695 R=henrika@webrtc.org Review URL: https://codereview.webrtc.org/1343393003 . Cr-Commit-Position: refs/heads/master@{#9968} --- talk/libjingle.gyp | 2 - talk/libjingle_tests.gyp | 1 - talk/media/webrtc/webrtcpassthroughrender.cc | 172 ---------------- talk/media/webrtc/webrtcpassthroughrender.h | 185 ------------------ .../webrtcpassthroughrender_unittest.cc | 172 ---------------- .../fakes/fake_external_transport.cc | 96 --------- .../auto_test/fakes/fake_external_transport.h | 46 ----- webrtc/voice_engine/voice_engine.gyp | 2 - 8 files changed, 676 deletions(-) delete mode 100644 talk/media/webrtc/webrtcpassthroughrender.cc delete mode 100644 talk/media/webrtc/webrtcpassthroughrender.h delete mode 100644 talk/media/webrtc/webrtcpassthroughrender_unittest.cc delete mode 100644 webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.cc delete mode 100644 webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.h diff --git a/talk/libjingle.gyp b/talk/libjingle.gyp index 6198ffe26e..2d4a27a80f 100755 --- a/talk/libjingle.gyp +++ b/talk/libjingle.gyp @@ -499,8 +499,6 @@ 'media/webrtc/webrtcmediaengine.cc', 'media/webrtc/webrtcmediaengine.h', 'media/webrtc/webrtcmediaengine.cc', - 'media/webrtc/webrtcpassthroughrender.cc', - 'media/webrtc/webrtcpassthroughrender.h', 'media/webrtc/webrtcvideocapturer.cc', 'media/webrtc/webrtcvideocapturer.h', 'media/webrtc/webrtcvideocapturerfactory.h', diff --git a/talk/libjingle_tests.gyp b/talk/libjingle_tests.gyp index 8100287610..49111f59e8 100755 --- a/talk/libjingle_tests.gyp +++ b/talk/libjingle_tests.gyp @@ -97,7 +97,6 @@ 'media/devices/filevideocapturer_unittest.cc', 'media/sctp/sctpdataengine_unittest.cc', 'media/webrtc/simulcast_unittest.cc', - 'media/webrtc/webrtcpassthroughrender_unittest.cc', 'media/webrtc/webrtcvideocapturer_unittest.cc', 'media/base/videoframe_unittest.h', 'media/webrtc/webrtcvideoframe_unittest.cc', diff --git a/talk/media/webrtc/webrtcpassthroughrender.cc b/talk/media/webrtc/webrtcpassthroughrender.cc deleted file mode 100644 index e317860762..0000000000 --- a/talk/media/webrtc/webrtcpassthroughrender.cc +++ /dev/null @@ -1,172 +0,0 @@ -/* - * libjingle - * Copyright 2004 Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#include "talk/media/webrtc/webrtcpassthroughrender.h" - -#include "webrtc/base/common.h" -#include "webrtc/base/logging.h" - -namespace cricket { - -#define LOG_FIND_STREAM_ERROR(func, id) LOG(LS_ERROR) \ - << "" << func << " - Failed to find stream: " << id - -class PassthroughStream: public webrtc::VideoRenderCallback { - public: - PassthroughStream() : running_(false) {} - virtual ~PassthroughStream() { - } - virtual int32_t RenderFrame(const uint32_t stream_id, - const webrtc::VideoFrame& videoFrame) { - rtc::CritScope cs(&stream_critical_); - // Send frame for rendering directly - if (running_ && renderer_) { - renderer_->RenderFrame(stream_id, videoFrame); - } - return 0; - } - int32_t SetRenderer(VideoRenderCallback* renderer) { - rtc::CritScope cs(&stream_critical_); - renderer_ = renderer; - return 0; - } - - int32_t StartRender() { - rtc::CritScope cs(&stream_critical_); - running_ = true; - return 0; - } - - int32_t StopRender() { - rtc::CritScope cs(&stream_critical_); - running_ = false; - return 0; - } - - private: - VideoRenderCallback* renderer_; - rtc::CriticalSection stream_critical_; - bool running_; -}; - -WebRtcPassthroughRender::WebRtcPassthroughRender() - : window_(NULL) { -} - -WebRtcPassthroughRender::~WebRtcPassthroughRender() { - while (!stream_render_map_.empty()) { - PassthroughStream* stream = stream_render_map_.begin()->second; - stream_render_map_.erase(stream_render_map_.begin()); - delete stream; - } -} - -webrtc::VideoRenderCallback* WebRtcPassthroughRender::AddIncomingRenderStream( - const uint32_t stream_id, - const uint32_t zOrder, - const float left, const float top, - const float right, const float bottom) { - rtc::CritScope cs(&render_critical_); - // Stream already exist. - if (FindStream(stream_id) != NULL) { - LOG(LS_ERROR) << "AddIncomingRenderStream - Stream already exists: " - << stream_id; - return NULL; - } - - PassthroughStream* stream = new PassthroughStream(); - // Store the stream - stream_render_map_[stream_id] = stream; - return stream; -} - -int32_t WebRtcPassthroughRender::DeleteIncomingRenderStream( - const uint32_t stream_id) { - rtc::CritScope cs(&render_critical_); - PassthroughStream* stream = FindStream(stream_id); - if (stream == NULL) { - LOG_FIND_STREAM_ERROR("DeleteIncomingRenderStream", stream_id); - return -1; - } - delete stream; - stream_render_map_.erase(stream_id); - return 0; -} - -int32_t WebRtcPassthroughRender::AddExternalRenderCallback( - const uint32_t stream_id, - webrtc::VideoRenderCallback* render_object) { - rtc::CritScope cs(&render_critical_); - PassthroughStream* stream = FindStream(stream_id); - if (stream == NULL) { - LOG_FIND_STREAM_ERROR("AddExternalRenderCallback", stream_id); - return -1; - } - return stream->SetRenderer(render_object); -} - -bool WebRtcPassthroughRender::HasIncomingRenderStream( - const uint32_t stream_id) const { - return (FindStream(stream_id) != NULL); -} - -webrtc::RawVideoType WebRtcPassthroughRender::PreferredVideoType() const { - return webrtc::kVideoI420; -} - -int32_t WebRtcPassthroughRender::StartRender(const uint32_t stream_id) { - rtc::CritScope cs(&render_critical_); - PassthroughStream* stream = FindStream(stream_id); - if (stream == NULL) { - LOG_FIND_STREAM_ERROR("StartRender", stream_id); - return -1; - } - return stream->StartRender(); -} - -int32_t WebRtcPassthroughRender::StopRender(const uint32_t stream_id) { - rtc::CritScope cs(&render_critical_); - PassthroughStream* stream = FindStream(stream_id); - if (stream == NULL) { - LOG_FIND_STREAM_ERROR("StopRender", stream_id); - return -1; - } - return stream->StopRender(); -} - -// TODO(ronghuawu): Is it ok to return non-const pointer to PassthroughStream -// from this const function FindStream. -PassthroughStream* WebRtcPassthroughRender::FindStream( - const uint32_t stream_id) const { - StreamMap::const_iterator it = stream_render_map_.find(stream_id); - if (it == stream_render_map_.end()) { - return NULL; - } - return it->second; -} - -} // namespace cricket diff --git a/talk/media/webrtc/webrtcpassthroughrender.h b/talk/media/webrtc/webrtcpassthroughrender.h deleted file mode 100644 index 685cfe881c..0000000000 --- a/talk/media/webrtc/webrtcpassthroughrender.h +++ /dev/null @@ -1,185 +0,0 @@ -/* - * libjingle - * Copyright 2004 Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -#ifndef TALK_MEDIA_WEBRTCPASSTHROUGHRENDER_H_ -#define TALK_MEDIA_WEBRTCPASSTHROUGHRENDER_H_ - -#include - -#include "webrtc/base/criticalsection.h" -#include "webrtc/modules/video_render/include/video_render.h" - -namespace cricket { -class PassthroughStream; - -class WebRtcPassthroughRender : public webrtc::VideoRender { - public: - WebRtcPassthroughRender(); - virtual ~WebRtcPassthroughRender(); - - int64_t TimeUntilNextProcess() override { return 0; } - - int32_t Process() override { return 0; } - - void* Window() override { - rtc::CritScope cs(&render_critical_); - return window_; - } - - int32_t ChangeWindow(void* window) override { - rtc::CritScope cs(&render_critical_); - window_ = window; - return 0; - } - - webrtc::VideoRenderCallback* AddIncomingRenderStream( - const uint32_t stream_id, - const uint32_t zOrder, - const float left, - const float top, - const float right, - const float bottom) override; - - int32_t DeleteIncomingRenderStream(const uint32_t stream_id) override; - - int32_t AddExternalRenderCallback( - const uint32_t stream_id, - webrtc::VideoRenderCallback* render_object) override; - - int32_t GetIncomingRenderStreamProperties(const uint32_t stream_id, - uint32_t& zOrder, - float& left, - float& top, - float& right, - float& bottom) const override { - return -1; - } - - uint32_t GetIncomingFrameRate(const uint32_t stream_id) override { return 0; } - - uint32_t GetNumIncomingRenderStreams() const override { - return static_cast(stream_render_map_.size()); - } - - bool HasIncomingRenderStream(const uint32_t stream_id) const override; - - int32_t RegisterRawFrameCallback( - const uint32_t stream_id, - webrtc::VideoRenderCallback* callback_obj) override { - return -1; - } - - int32_t StartRender(const uint32_t stream_id) override; - - int32_t StopRender(const uint32_t stream_id) override; - - int32_t ResetRender() override { return 0; } - - webrtc::RawVideoType PreferredVideoType() const override; - - bool IsFullScreen() override { return false; } - - int32_t GetScreenResolution(uint32_t& screenWidth, - uint32_t& screenHeight) const override { - return -1; - } - - uint32_t RenderFrameRate(const uint32_t stream_id) override { return 0; } - - int32_t SetStreamCropping(const uint32_t stream_id, - const float left, - const float top, - const float right, - const float bottom) override { - return -1; - } - - int32_t SetExpectedRenderDelay(uint32_t stream_id, - int32_t delay_ms) override { - return -1; - } - - int32_t ConfigureRenderer(const uint32_t stream_id, - const unsigned int zOrder, - const float left, - const float top, - const float right, - const float bottom) override { - return -1; - } - - int32_t SetTransparentBackground(const bool enable) override { return -1; } - - int32_t FullScreenRender(void* window, const bool enable) override { - return -1; - } - - int32_t SetBitmap(const void* bitMap, - const uint8_t pictureId, - const void* colorKey, - const float left, - const float top, - const float right, - const float bottom) override { - return -1; - } - - int32_t SetText(const uint8_t textId, - const uint8_t* text, - const int32_t textLength, - const uint32_t textColorRef, - const uint32_t backgroundColorRef, - const float left, - const float top, - const float right, - const float bottom) override { - return -1; - } - - int32_t SetStartImage(const uint32_t stream_id, - const webrtc::VideoFrame& videoFrame) override { - return -1; - } - - int32_t SetTimeoutImage(const uint32_t stream_id, - const webrtc::VideoFrame& videoFrame, - const uint32_t timeout) override { - return -1; - } - - private: - typedef std::map StreamMap; - - PassthroughStream* FindStream(const uint32_t stream_id) const; - - void* window_; - StreamMap stream_render_map_; - rtc::CriticalSection render_critical_; -}; -} // namespace cricket - -#endif // TALK_MEDIA_WEBRTCPASSTHROUGHRENDER_H_ diff --git a/talk/media/webrtc/webrtcpassthroughrender_unittest.cc b/talk/media/webrtc/webrtcpassthroughrender_unittest.cc deleted file mode 100644 index 65eed05f8e..0000000000 --- a/talk/media/webrtc/webrtcpassthroughrender_unittest.cc +++ /dev/null @@ -1,172 +0,0 @@ -/* - * libjingle - * Copyright 2008 Google Inc. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions are met: - * - * 1. Redistributions of source code must retain the above copyright notice, - * this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. - * 3. The name of the author may not be used to endorse or promote products - * derived from this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO - * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, - * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; - * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, - * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR - * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF - * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - */ - -// Author: Ronghua Wu (ronghuawu@google.com) - -#include - -#include "talk/media/base/testutils.h" -#include "talk/media/webrtc/webrtcpassthroughrender.h" -#include "webrtc/base/gunit.h" - -class WebRtcPassthroughRenderTest : public testing::Test { - public: - class ExternalRenderer : public webrtc::VideoRenderCallback { - public: - ExternalRenderer() : frame_num_(0) { - } - - virtual ~ExternalRenderer() { - } - - virtual int32_t RenderFrame(const uint32_t stream_id, - const webrtc::VideoFrame& videoFrame) { - ++frame_num_; - LOG(INFO) << "RenderFrame stream_id: " << stream_id - << " frame_num: " << frame_num_; - return 0; - } - - int frame_num() const { - return frame_num_; - } - - private: - int frame_num_; - }; - - WebRtcPassthroughRenderTest() - : renderer_(new cricket::WebRtcPassthroughRender()) { - } - - ~WebRtcPassthroughRenderTest() { - } - - webrtc::VideoRenderCallback* AddIncomingRenderStream(int stream_id) { - return renderer_->AddIncomingRenderStream(stream_id, 0, 0, 0, 0, 0); - } - - bool HasIncomingRenderStream(int stream_id) { - return renderer_->HasIncomingRenderStream(stream_id); - } - - bool DeleteIncomingRenderStream(int stream_id) { - return (renderer_->DeleteIncomingRenderStream(stream_id) == 0); - } - - bool AddExternalRenderCallback(int stream_id, - webrtc::VideoRenderCallback* renderer) { - return (renderer_->AddExternalRenderCallback(stream_id, renderer) == 0); - } - - bool StartRender(int stream_id) { - return (renderer_->StartRender(stream_id) == 0); - } - - bool StopRender(int stream_id) { - return (renderer_->StopRender(stream_id) == 0); - } - - private: - rtc::scoped_ptr renderer_; -}; - -TEST_F(WebRtcPassthroughRenderTest, Streams) { - const int stream_id1 = 1234; - const int stream_id2 = 5678; - const int stream_id3 = 9012; // A stream that doesn't exist. - webrtc::VideoRenderCallback* stream = NULL; - // Add a new stream - stream = AddIncomingRenderStream(stream_id1); - EXPECT_TRUE(stream != NULL); - EXPECT_TRUE(HasIncomingRenderStream(stream_id1)); - // Tried to add a already existed stream should return null - stream =AddIncomingRenderStream(stream_id1); - EXPECT_TRUE(stream == NULL); - stream = AddIncomingRenderStream(stream_id2); - EXPECT_TRUE(stream != NULL); - EXPECT_TRUE(HasIncomingRenderStream(stream_id2)); - // Remove the stream - EXPECT_FALSE(DeleteIncomingRenderStream(stream_id3)); - EXPECT_TRUE(DeleteIncomingRenderStream(stream_id2)); - EXPECT_TRUE(!HasIncomingRenderStream(stream_id2)); - // Add back the removed stream - stream = AddIncomingRenderStream(stream_id2); - EXPECT_TRUE(stream != NULL); - EXPECT_TRUE(HasIncomingRenderStream(stream_id2)); -} - -TEST_F(WebRtcPassthroughRenderTest, Renderer) { - webrtc::VideoFrame frame; - const int stream_id1 = 1234; - const int stream_id2 = 5678; - const int stream_id3 = 9012; // A stream that doesn't exist. - webrtc::VideoRenderCallback* stream1 = NULL; - webrtc::VideoRenderCallback* stream2 = NULL; - // Add two new stream - stream1 = AddIncomingRenderStream(stream_id1); - EXPECT_TRUE(stream1 != NULL); - EXPECT_TRUE(HasIncomingRenderStream(stream_id1)); - stream2 = AddIncomingRenderStream(stream_id2); - EXPECT_TRUE(stream2 != NULL); - EXPECT_TRUE(HasIncomingRenderStream(stream_id2)); - // Register the external renderer - WebRtcPassthroughRenderTest::ExternalRenderer renderer1; - WebRtcPassthroughRenderTest::ExternalRenderer renderer2; - EXPECT_FALSE(AddExternalRenderCallback(stream_id3, &renderer1)); - EXPECT_TRUE(AddExternalRenderCallback(stream_id1, &renderer1)); - EXPECT_TRUE(AddExternalRenderCallback(stream_id2, &renderer2)); - int test_frame_num = 10; - // RenderFrame without starting the render - for (int i = 0; i < test_frame_num; ++i) { - stream1->RenderFrame(stream_id1, frame); - } - EXPECT_EQ(0, renderer1.frame_num()); - // Start the render and test again. - EXPECT_FALSE(StartRender(stream_id3)); - EXPECT_TRUE(StartRender(stream_id1)); - for (int i = 0; i < test_frame_num; ++i) { - stream1->RenderFrame(stream_id1, frame); - } - EXPECT_EQ(test_frame_num, renderer1.frame_num()); - // Stop the render and test again. - EXPECT_FALSE(StopRender(stream_id3)); - EXPECT_TRUE(StopRender(stream_id1)); - for (int i = 0; i < test_frame_num; ++i) { - stream1->RenderFrame(stream_id1, frame); - } - // The frame number should not have changed. - EXPECT_EQ(test_frame_num, renderer1.frame_num()); - - // Test on stream2 with a differnt number. - EXPECT_TRUE(StartRender(stream_id2)); - test_frame_num = 30; - for (int i = 0; i < test_frame_num; ++i) { - stream2->RenderFrame(stream_id2, frame); - } - EXPECT_EQ(test_frame_num, renderer2.frame_num()); -} diff --git a/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.cc b/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.cc deleted file mode 100644 index c825ea5861..0000000000 --- a/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.cc +++ /dev/null @@ -1,96 +0,0 @@ -/* - * Copyright (c) 2012 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. - */ - -#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/event_wrapper.h" -#include "webrtc/system_wrappers/interface/sleep.h" -#include "webrtc/system_wrappers/interface/thread_wrapper.h" -#include "webrtc/voice_engine/include/voe_network.h" -#include "webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.h" -#include "webrtc/voice_engine/voice_engine_defines.h" - -FakeExternalTransport::FakeExternalTransport(webrtc::VoENetwork* ptr) - : my_network_(ptr), - lock_(NULL), - event_(NULL), - length_(0), - channel_(0), - delay_is_enabled_(0), - delay_time_in_ms_(0) { - const char* thread_name = "external_thread"; - lock_ = webrtc::CriticalSectionWrapper::CreateCriticalSection(); - event_ = webrtc::EventWrapper::Create(); - thread_ = webrtc::ThreadWrapper::CreateThread(Run, this, thread_name); - if (thread_) { - thread_->Start(); - thread_->SetPriority(webrtc::kHighPriority); - } -} - -FakeExternalTransport::~FakeExternalTransport() { - if (thread_) { - event_->Set(); - thread_->Stop(); - delete event_; - event_ = NULL; - delete lock_; - lock_ = NULL; - } -} - -bool FakeExternalTransport::Run(void* ptr) { - return static_cast (ptr)->Process(); -} - -bool FakeExternalTransport::Process() { - switch (event_->Wait(500)) { - case webrtc::kEventSignaled: - lock_->Enter(); - my_network_->ReceivedRTPPacket(channel_, packet_buffer_, length_, - webrtc::PacketTime()); - lock_->Leave(); - return true; - case webrtc::kEventTimeout: - return true; - case webrtc::kEventError: - break; - } - return true; -} - -int FakeExternalTransport::SendPacket(int channel, - const void *data, - size_t len) { - lock_->Enter(); - if (len < 1612) { - memcpy(packet_buffer_, (const unsigned char*) data, len); - length_ = len; - channel_ = channel; - } - lock_->Leave(); - event_->Set(); // Triggers ReceivedRTPPacket() from worker thread. - return static_cast(len); -} - -int FakeExternalTransport::SendRTCPPacket(int channel, - const void *data, - size_t len) { - if (delay_is_enabled_) { - webrtc::SleepMs(delay_time_in_ms_); - } - my_network_->ReceivedRTCPPacket(channel, data, len); - return static_cast(len); -} - -void FakeExternalTransport::SetDelayStatus(bool enable, - unsigned int delayInMs) { - delay_is_enabled_ = enable; - delay_time_in_ms_ = delayInMs; -} diff --git a/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.h b/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.h deleted file mode 100644 index aecc58264a..0000000000 --- a/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.h +++ /dev/null @@ -1,46 +0,0 @@ -/* - * Copyright (c) 2012 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. - */ -#ifndef VOICE_ENGINE_MAIN_TEST_AUTO_TEST_FAKES_FAKE_EXTERNAL_TRANSPORT_H_ -#define VOICE_ENGINE_MAIN_TEST_AUTO_TEST_FAKES_FAKE_EXTERNAL_TRANSPORT_H_ - -#include "webrtc/common_types.h" - -namespace webrtc { -class CriticalSectionWrapper; -class EventWrapper; -class ThreadWrapper; -class VoENetwork; -} - -class FakeExternalTransport : public webrtc::Transport { - public: - explicit FakeExternalTransport(webrtc::VoENetwork* ptr); - virtual ~FakeExternalTransport(); - int SendPacket(int channel, const void* data, size_t len) override; - int SendRTCPPacket(int channel, const void* data, size_t len) override; - void SetDelayStatus(bool enabled, unsigned int delayInMs = 100); - - webrtc::VoENetwork* my_network_; - private: - static bool Run(void* ptr); - bool Process(); - private: - rtc::scoped_ptr thread_; - webrtc::CriticalSectionWrapper* lock_; - webrtc::EventWrapper* event_; - private: - unsigned char packet_buffer_[1612]; - size_t length_; - int channel_; - bool delay_is_enabled_; - int delay_time_in_ms_; -}; - -#endif // VOICE_ENGINE_MAIN_TEST_AUTO_TEST_FAKES_FAKE_EXTERNAL_TRANSPORT_H_ diff --git a/webrtc/voice_engine/voice_engine.gyp b/webrtc/voice_engine/voice_engine.gyp index 37ffc5390d..221b2aa681 100644 --- a/webrtc/voice_engine/voice_engine.gyp +++ b/webrtc/voice_engine/voice_engine.gyp @@ -162,8 +162,6 @@ 'test/auto_test/extended/ec_metrics_test.cc', 'test/auto_test/fakes/conference_transport.cc', 'test/auto_test/fakes/conference_transport.h', - 'test/auto_test/fakes/fake_external_transport.cc', - 'test/auto_test/fakes/fake_external_transport.h', 'test/auto_test/fakes/loudest_filter.cc', 'test/auto_test/fakes/loudest_filter.h', 'test/auto_test/fixtures/after_initialization_fixture.cc',