e5ad5ca06a
Reason for revert: Intend to fix perf failures and reland. Original issue's description: > Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ ) > > Reason for revert: > Reverting since this seems to break multiple WebRTC Perf buildbots > > Original issue's description: > > Don't hardcode MediaType::ANY in FakeNetworkPipe. > > > > Instead let each test set the appropriate media type. This simplifies > > demuxing in Call and later in RtpTransportController. > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2774463003 > > Cr-Commit-Position: refs/heads/master@{#17418} > > Committed:9c47b00e24> > TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2784543002 > Cr-Commit-Position: refs/heads/master@{#17427} > Committed:3a3bd50610TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,lliuu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2783853002 Cr-Commit-Position: refs/heads/master@{#17459}
77 lines
2.2 KiB
C++
77 lines
2.2 KiB
C++
/*
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* Copyright (c) 2013 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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*/
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#ifndef WEBRTC_TEST_DIRECT_TRANSPORT_H_
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#define WEBRTC_TEST_DIRECT_TRANSPORT_H_
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#include <assert.h>
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#include <deque>
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#include "webrtc/api/call/transport.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/platform_thread.h"
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#include "webrtc/call/call.h"
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#include "webrtc/test/fake_network_pipe.h"
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namespace webrtc {
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class Clock;
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class PacketReceiver;
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namespace test {
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class DirectTransport : public Transport {
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public:
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DirectTransport(Call* send_call, MediaType media_type);
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DirectTransport(const FakeNetworkPipe::Config& config, Call* send_call,
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MediaType media_type);
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// These deprecated variants always use MediaType::VIDEO.
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RTC_DEPRECATED explicit DirectTransport(Call* send_call)
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: DirectTransport(send_call, MediaType::VIDEO) {}
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RTC_DEPRECATED DirectTransport(const FakeNetworkPipe::Config& config,
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Call* send_call)
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: DirectTransport(config, send_call, MediaType::VIDEO) {}
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~DirectTransport();
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void SetConfig(const FakeNetworkPipe::Config& config);
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virtual void StopSending();
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// TODO(holmer): Look into moving this to the constructor.
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virtual void SetReceiver(PacketReceiver* receiver);
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bool SendRtp(const uint8_t* data,
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size_t length,
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const PacketOptions& options) override;
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bool SendRtcp(const uint8_t* data, size_t length) override;
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int GetAverageDelayMs();
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private:
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static bool NetworkProcess(void* transport);
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bool SendPackets();
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rtc::CriticalSection lock_;
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Call* const send_call_;
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rtc::Event packet_event_;
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rtc::PlatformThread thread_;
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Clock* const clock_;
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bool shutting_down_;
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FakeNetworkPipe fake_network_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_TEST_DIRECT_TRANSPORT_H_
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