02b3d275a0
Reason for revert: A fix is being prepared downstream so this can now go in. Original issue's description: > Revert of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #5 id:80001 of https://codereview.webrtc.org/1897233002/ ) > > Reason for revert: > API changes broke downstream. > > Original issue's description: > > Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. > > EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport. > > EncodedImageCallback can of course be cleaned up in the future. > > > > This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter. > > > > BUG=webrtc::5687 > > > > Committed: https://crrev.com/f5d55aaecdc39e9cc66eb6e87614f04afe28f6eb > > Cr-Commit-Position: refs/heads/master@{#12436} > > TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:5687 > > Committed: https://crrev.com/a261e6136655af33f283eda8e60a6dd93dd746a4 > Cr-Commit-Position: refs/heads/master@{#12441} TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5687 Review URL: https://codereview.webrtc.org/1905583002 Cr-Commit-Position: refs/heads/master@{#12442}
212 lines
8.0 KiB
C++
212 lines
8.0 KiB
C++
/*
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* Copyright (c) 2015 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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#include <memory>
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#include "testing/gmock/include/gmock/gmock.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
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#include "webrtc/modules/video_coding/include/video_codec_interface.h"
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#include "webrtc/video/payload_router.h"
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using ::testing::_;
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using ::testing::AnyNumber;
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using ::testing::NiceMock;
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using ::testing::Return;
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namespace webrtc {
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TEST(PayloadRouterTest, SendOnOneModule) {
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MockRtpRtcp rtp;
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std::vector<RtpRtcp*> modules(1, &rtp);
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uint8_t payload = 'a';
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int8_t payload_type = 96;
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EncodedImage encoded_image;
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encoded_image._timeStamp = 1;
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encoded_image.capture_time_ms_ = 2;
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encoded_image._frameType = kVideoFrameKey;
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encoded_image._buffer = &payload;
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encoded_image._length = 1;
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PayloadRouter payload_router(modules, payload_type);
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payload_router.SetSendingRtpModules(modules.size());
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EXPECT_CALL(rtp, SendOutgoingData(encoded_image._frameType, payload_type,
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encoded_image._timeStamp,
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encoded_image.capture_time_ms_, &payload,
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encoded_image._length, nullptr, _))
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.Times(0);
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EXPECT_EQ(-1, payload_router.Encoded(encoded_image, nullptr, nullptr));
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payload_router.set_active(true);
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EXPECT_CALL(rtp, SendOutgoingData(encoded_image._frameType, payload_type,
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encoded_image._timeStamp,
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encoded_image.capture_time_ms_, &payload,
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encoded_image._length, nullptr, _))
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.Times(1);
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EXPECT_EQ(0, payload_router.Encoded(encoded_image, nullptr, nullptr));
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payload_router.set_active(false);
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EXPECT_CALL(rtp, SendOutgoingData(encoded_image._frameType, payload_type,
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encoded_image._timeStamp,
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encoded_image.capture_time_ms_, &payload,
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encoded_image._length, nullptr, _))
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.Times(0);
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EXPECT_EQ(-1, payload_router.Encoded(encoded_image, nullptr, nullptr));
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payload_router.set_active(true);
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EXPECT_CALL(rtp, SendOutgoingData(encoded_image._frameType, payload_type,
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encoded_image._timeStamp,
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encoded_image.capture_time_ms_, &payload,
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encoded_image._length, nullptr, _))
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.Times(1);
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EXPECT_EQ(0, payload_router.Encoded(encoded_image, nullptr, nullptr));
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payload_router.SetSendingRtpModules(0);
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EXPECT_CALL(rtp, SendOutgoingData(encoded_image._frameType, payload_type,
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encoded_image._timeStamp,
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encoded_image.capture_time_ms_, &payload,
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encoded_image._length, nullptr, _))
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.Times(0);
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EXPECT_EQ(-1, payload_router.Encoded(encoded_image, nullptr, nullptr));
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}
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TEST(PayloadRouterTest, SendSimulcast) {
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MockRtpRtcp rtp_1;
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MockRtpRtcp rtp_2;
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std::vector<RtpRtcp*> modules;
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modules.push_back(&rtp_1);
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modules.push_back(&rtp_2);
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int8_t payload_type = 96;
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uint8_t payload = 'a';
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EncodedImage encoded_image;
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encoded_image._timeStamp = 1;
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encoded_image.capture_time_ms_ = 2;
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encoded_image._frameType = kVideoFrameKey;
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encoded_image._buffer = &payload;
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encoded_image._length = 1;
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PayloadRouter payload_router(modules, payload_type);
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payload_router.SetSendingRtpModules(modules.size());
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CodecSpecificInfo codec_info_1;
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memset(&codec_info_1, 0, sizeof(CodecSpecificInfo));
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codec_info_1.codecType = kVideoCodecVP8;
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codec_info_1.codecSpecific.VP8.simulcastIdx = 0;
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payload_router.set_active(true);
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EXPECT_CALL(rtp_1, SendOutgoingData(encoded_image._frameType, payload_type,
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encoded_image._timeStamp,
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encoded_image.capture_time_ms_, &payload,
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encoded_image._length, nullptr, _))
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.Times(1);
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EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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EXPECT_EQ(0, payload_router.Encoded(encoded_image, &codec_info_1, nullptr));
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CodecSpecificInfo codec_info_2;
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memset(&codec_info_2, 0, sizeof(CodecSpecificInfo));
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codec_info_2.codecType = kVideoCodecVP8;
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codec_info_2.codecSpecific.VP8.simulcastIdx = 1;
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EXPECT_CALL(rtp_2, SendOutgoingData(encoded_image._frameType, payload_type,
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encoded_image._timeStamp,
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encoded_image.capture_time_ms_, &payload,
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encoded_image._length, nullptr, _))
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.Times(1);
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EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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EXPECT_EQ(0, payload_router.Encoded(encoded_image, &codec_info_2, nullptr));
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// Inactive.
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payload_router.set_active(false);
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EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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EXPECT_EQ(-1, payload_router.Encoded(encoded_image, &codec_info_1, nullptr));
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EXPECT_EQ(-1, payload_router.Encoded(encoded_image, &codec_info_2, nullptr));
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// Invalid simulcast index.
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payload_router.SetSendingRtpModules(1);
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payload_router.set_active(true);
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EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
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.Times(0);
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codec_info_2.codecSpecific.VP8.simulcastIdx = 1;
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EXPECT_EQ(-1, payload_router.Encoded(encoded_image, &codec_info_2, nullptr));
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}
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TEST(PayloadRouterTest, MaxPayloadLength) {
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// Without any limitations from the modules, verify we get the max payload
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// length for IP/UDP/SRTP with a MTU of 150 bytes.
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const size_t kDefaultMaxLength = 1500 - 20 - 8 - 12 - 4;
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MockRtpRtcp rtp_1;
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MockRtpRtcp rtp_2;
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std::vector<RtpRtcp*> modules;
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modules.push_back(&rtp_1);
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modules.push_back(&rtp_2);
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PayloadRouter payload_router(modules, 42);
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EXPECT_EQ(kDefaultMaxLength, PayloadRouter::DefaultMaxPayloadLength());
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payload_router.SetSendingRtpModules(modules.size());
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// Modules return a higher length than the default value.
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EXPECT_CALL(rtp_1, MaxDataPayloadLength())
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.Times(1)
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.WillOnce(Return(kDefaultMaxLength + 10));
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EXPECT_CALL(rtp_2, MaxDataPayloadLength())
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.Times(1)
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.WillOnce(Return(kDefaultMaxLength + 10));
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EXPECT_EQ(kDefaultMaxLength, payload_router.MaxPayloadLength());
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// The modules return a value lower than default.
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const size_t kTestMinPayloadLength = 1001;
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EXPECT_CALL(rtp_1, MaxDataPayloadLength())
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.Times(1)
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.WillOnce(Return(kTestMinPayloadLength + 10));
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EXPECT_CALL(rtp_2, MaxDataPayloadLength())
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.Times(1)
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.WillOnce(Return(kTestMinPayloadLength));
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EXPECT_EQ(kTestMinPayloadLength, payload_router.MaxPayloadLength());
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}
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TEST(PayloadRouterTest, SetTargetSendBitrates) {
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MockRtpRtcp rtp_1;
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MockRtpRtcp rtp_2;
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std::vector<RtpRtcp*> modules;
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modules.push_back(&rtp_1);
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modules.push_back(&rtp_2);
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PayloadRouter payload_router(modules, 42);
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payload_router.SetSendingRtpModules(modules.size());
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const uint32_t bitrate_1 = 10000;
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const uint32_t bitrate_2 = 76543;
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std::vector<uint32_t> bitrates(2, bitrate_1);
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bitrates[1] = bitrate_2;
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EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
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.Times(1);
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EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
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.Times(1);
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payload_router.SetTargetSendBitrates(bitrates);
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bitrates.resize(1);
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EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
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.Times(1);
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EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
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.Times(0);
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payload_router.SetTargetSendBitrates(bitrates);
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}
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} // namespace webrtc
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