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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 #include <algorithm> | 10 #include <algorithm> |
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1787 send_config->rtp.ssrcs[0]; | 1787 send_config->rtp.ssrcs[0]; |
1788 | 1788 |
1789 if (stream_index == 0) | 1789 if (stream_index == 0) |
1790 first_media_ssrc_ = send_config->rtp.ssrcs[0]; | 1790 first_media_ssrc_ = send_config->rtp.ssrcs[0]; |
1791 } | 1791 } |
1792 | 1792 |
1793 void UpdateReceiveConfig( | 1793 void UpdateReceiveConfig( |
1794 size_t stream_index, | 1794 size_t stream_index, |
1795 VideoReceiveStream::Config* receive_config) override { | 1795 VideoReceiveStream::Config* receive_config) override { |
1796 receive_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; | 1796 receive_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| 1797 #if 0 |
1797 receive_config->rtp.extensions.clear(); | 1798 receive_config->rtp.extensions.clear(); |
1798 receive_config->rtp.extensions.push_back(RtpExtension( | 1799 receive_config->rtp.extensions.push_back(RtpExtension( |
1799 RtpExtension::kTransportSequenceNumberUri, kExtensionId)); | 1800 RtpExtension::kTransportSequenceNumberUri, kExtensionId)); |
| 1801 #endif |
1800 receive_config->renderer = &fake_renderer_; | 1802 receive_config->renderer = &fake_renderer_; |
1801 } | 1803 } |
1802 | 1804 |
1803 test::DirectTransport* CreateSendTransport(Call* sender_call) override { | 1805 test::DirectTransport* CreateSendTransport(Call* sender_call) override { |
1804 std::map<uint8_t, MediaType> payload_type_map = | 1806 std::map<uint8_t, MediaType> payload_type_map = |
1805 MultiStreamTest::payload_type_map_; | 1807 MultiStreamTest::payload_type_map_; |
1806 RTC_DCHECK(payload_type_map.find(kSendRtxPayloadType) == | 1808 RTC_DCHECK(payload_type_map.find(kSendRtxPayloadType) == |
1807 payload_type_map.end()); | 1809 payload_type_map.end()); |
1808 payload_type_map[kSendRtxPayloadType] = MediaType::VIDEO; | 1810 payload_type_map[kSendRtxPayloadType] = MediaType::VIDEO; |
1809 observer_ = | 1811 observer_ = |
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4278 << "Reduced-size RTCP require rtcp-rsize to be negotiated."; | 4280 << "Reduced-size RTCP require rtcp-rsize to be negotiated."; |
4279 EXPECT_FALSE(default_receive_config.rtp.remb) | 4281 EXPECT_FALSE(default_receive_config.rtp.remb) |
4280 << "REMB require rtcp-fb: goog-remb to be negotiated."; | 4282 << "REMB require rtcp-fb: goog-remb to be negotiated."; |
4281 EXPECT_FALSE( | 4283 EXPECT_FALSE( |
4282 default_receive_config.rtp.rtcp_xr.receiver_reference_time_report) | 4284 default_receive_config.rtp.rtcp_xr.receiver_reference_time_report) |
4283 << "RTCP XR settings require rtcp-xr to be negotiated."; | 4285 << "RTCP XR settings require rtcp-xr to be negotiated."; |
4284 EXPECT_EQ(0U, default_receive_config.rtp.rtx_ssrc) | 4286 EXPECT_EQ(0U, default_receive_config.rtp.rtx_ssrc) |
4285 << "Enabling RTX requires ssrc-group: FID negotiation"; | 4287 << "Enabling RTX requires ssrc-group: FID negotiation"; |
4286 EXPECT_TRUE(default_receive_config.rtp.rtx_payload_types.empty()) | 4288 EXPECT_TRUE(default_receive_config.rtp.rtx_payload_types.empty()) |
4287 << "Enabling RTX requires rtpmap: rtx negotiation."; | 4289 << "Enabling RTX requires rtpmap: rtx negotiation."; |
| 4290 #if 0 |
4288 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) | 4291 EXPECT_TRUE(default_receive_config.rtp.extensions.empty()) |
4289 << "Enabling RTP extensions require negotiation."; | 4292 << "Enabling RTP extensions require negotiation."; |
4290 | 4293 #endif |
4291 VerifyEmptyNackConfig(default_receive_config.rtp.nack); | 4294 VerifyEmptyNackConfig(default_receive_config.rtp.nack); |
4292 VerifyEmptyUlpfecConfig(default_receive_config.rtp.ulpfec); | 4295 VerifyEmptyUlpfecConfig(default_receive_config.rtp.ulpfec); |
4293 } | 4296 } |
4294 | 4297 |
4295 TEST_F(EndToEndTest, VerifyDefaultFlexfecReceiveConfigParameters) { | 4298 TEST_F(EndToEndTest, VerifyDefaultFlexfecReceiveConfigParameters) { |
4296 test::NullTransport rtcp_send_transport; | 4299 test::NullTransport rtcp_send_transport; |
4297 FlexfecReceiveStream::Config default_receive_config(&rtcp_send_transport); | 4300 FlexfecReceiveStream::Config default_receive_config(&rtcp_send_transport); |
4298 EXPECT_EQ(-1, default_receive_config.payload_type) | 4301 EXPECT_EQ(-1, default_receive_config.payload_type) |
4299 << "Enabling FlexFEC requires rtpmap: flexfec negotiation."; | 4302 << "Enabling FlexFEC requires rtpmap: flexfec negotiation."; |
4300 EXPECT_EQ(0U, default_receive_config.remote_ssrc) | 4303 EXPECT_EQ(0U, default_receive_config.remote_ssrc) |
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4470 std::unique_ptr<VideoEncoder> encoder_; | 4473 std::unique_ptr<VideoEncoder> encoder_; |
4471 std::unique_ptr<VideoDecoder> decoder_; | 4474 std::unique_ptr<VideoDecoder> decoder_; |
4472 rtc::CriticalSection crit_; | 4475 rtc::CriticalSection crit_; |
4473 int recorded_frames_ GUARDED_BY(crit_); | 4476 int recorded_frames_ GUARDED_BY(crit_); |
4474 } test(this); | 4477 } test(this); |
4475 | 4478 |
4476 RunBaseTest(&test); | 4479 RunBaseTest(&test); |
4477 } | 4480 } |
4478 | 4481 |
4479 } // namespace webrtc | 4482 } // namespace webrtc |
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