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1 /* | 1 /* |
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2012 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 | 10 |
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568 | 568 |
569 DestroyAudioRtpSender(); | 569 DestroyAudioRtpSender(); |
570 } | 570 } |
571 | 571 |
572 TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) { | 572 TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) { |
573 CreateAudioRtpSender(); | 573 CreateAudioRtpSender(); |
574 | 574 |
575 EXPECT_EQ(-1, voice_media_channel_->max_bps()); | 575 EXPECT_EQ(-1, voice_media_channel_->max_bps()); |
576 webrtc::RtpParameters params = audio_rtp_sender_->GetParameters(); | 576 webrtc::RtpParameters params = audio_rtp_sender_->GetParameters(); |
577 EXPECT_EQ(1, params.encodings.size()); | 577 EXPECT_EQ(1, params.encodings.size()); |
578 EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps); | 578 EXPECT_FALSE(params.encodings[0].max_bitrate_bps); |
579 params.encodings[0].max_bitrate_bps = 1000; | 579 params.encodings[0].max_bitrate_bps = rtc::Optional<int>(1000); |
580 EXPECT_TRUE(audio_rtp_sender_->SetParameters(params)); | 580 EXPECT_TRUE(audio_rtp_sender_->SetParameters(params)); |
581 | 581 |
582 // Read back the parameters and verify they have been changed. | 582 // Read back the parameters and verify they have been changed. |
583 params = audio_rtp_sender_->GetParameters(); | 583 params = audio_rtp_sender_->GetParameters(); |
584 EXPECT_EQ(1, params.encodings.size()); | 584 EXPECT_EQ(1, params.encodings.size()); |
585 EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps); | 585 EXPECT_EQ(rtc::Optional<int>(1000), params.encodings[0].max_bitrate_bps); |
586 | 586 |
587 // Verify that the audio channel received the new parameters. | 587 // Verify that the audio channel received the new parameters. |
588 params = voice_media_channel_->GetRtpSendParameters(kAudioSsrc); | 588 params = voice_media_channel_->GetRtpSendParameters(kAudioSsrc); |
589 EXPECT_EQ(1, params.encodings.size()); | 589 EXPECT_EQ(1, params.encodings.size()); |
590 EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps); | 590 EXPECT_EQ(rtc::Optional<int>(1000), params.encodings[0].max_bitrate_bps); |
591 | 591 |
592 // Verify that the global bitrate limit has not been changed. | 592 // Verify that the global bitrate limit has not been changed. |
593 EXPECT_EQ(-1, voice_media_channel_->max_bps()); | 593 EXPECT_EQ(-1, voice_media_channel_->max_bps()); |
594 | 594 |
595 DestroyAudioRtpSender(); | 595 DestroyAudioRtpSender(); |
596 } | 596 } |
597 | 597 |
598 TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) { | 598 TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) { |
599 CreateVideoRtpSender(); | 599 CreateVideoRtpSender(); |
600 | 600 |
601 RtpParameters params = video_rtp_sender_->GetParameters(); | 601 RtpParameters params = video_rtp_sender_->GetParameters(); |
602 EXPECT_EQ(1u, params.encodings.size()); | 602 EXPECT_EQ(1u, params.encodings.size()); |
603 EXPECT_TRUE(video_rtp_sender_->SetParameters(params)); | 603 EXPECT_TRUE(video_rtp_sender_->SetParameters(params)); |
604 | 604 |
605 DestroyVideoRtpSender(); | 605 DestroyVideoRtpSender(); |
606 } | 606 } |
607 | 607 |
608 TEST_F(RtpSenderReceiverTest, SetVideoMaxSendBitrate) { | 608 TEST_F(RtpSenderReceiverTest, SetVideoMaxSendBitrate) { |
609 CreateVideoRtpSender(); | 609 CreateVideoRtpSender(); |
610 | 610 |
611 EXPECT_EQ(-1, video_media_channel_->max_bps()); | 611 EXPECT_EQ(-1, video_media_channel_->max_bps()); |
612 webrtc::RtpParameters params = video_rtp_sender_->GetParameters(); | 612 webrtc::RtpParameters params = video_rtp_sender_->GetParameters(); |
613 EXPECT_EQ(1, params.encodings.size()); | 613 EXPECT_EQ(1, params.encodings.size()); |
614 EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps); | 614 EXPECT_FALSE(params.encodings[0].max_bitrate_bps); |
615 params.encodings[0].max_bitrate_bps = 1000; | 615 params.encodings[0].max_bitrate_bps = rtc::Optional<int>(1000); |
616 EXPECT_TRUE(video_rtp_sender_->SetParameters(params)); | 616 EXPECT_TRUE(video_rtp_sender_->SetParameters(params)); |
617 | 617 |
618 // Read back the parameters and verify they have been changed. | 618 // Read back the parameters and verify they have been changed. |
619 params = video_rtp_sender_->GetParameters(); | 619 params = video_rtp_sender_->GetParameters(); |
620 EXPECT_EQ(1, params.encodings.size()); | 620 EXPECT_EQ(1, params.encodings.size()); |
621 EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps); | 621 EXPECT_EQ(rtc::Optional<int>(1000), params.encodings[0].max_bitrate_bps); |
622 | 622 |
623 // Verify that the video channel received the new parameters. | 623 // Verify that the video channel received the new parameters. |
624 params = video_media_channel_->GetRtpSendParameters(kVideoSsrc); | 624 params = video_media_channel_->GetRtpSendParameters(kVideoSsrc); |
625 EXPECT_EQ(1, params.encodings.size()); | 625 EXPECT_EQ(1, params.encodings.size()); |
626 EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps); | 626 EXPECT_EQ(rtc::Optional<int>(1000), params.encodings[0].max_bitrate_bps); |
627 | 627 |
628 // Verify that the global bitrate limit has not been changed. | 628 // Verify that the global bitrate limit has not been changed. |
629 EXPECT_EQ(-1, video_media_channel_->max_bps()); | 629 EXPECT_EQ(-1, video_media_channel_->max_bps()); |
630 | 630 |
631 DestroyVideoRtpSender(); | 631 DestroyVideoRtpSender(); |
632 } | 632 } |
633 | 633 |
634 TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetParameters) { | 634 TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetParameters) { |
635 CreateAudioRtpReceiver(); | 635 CreateAudioRtpReceiver(); |
636 | 636 |
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798 // Make sure the signal from "GetOnDestroyedSignal()" fires when the sender is | 798 // Make sure the signal from "GetOnDestroyedSignal()" fires when the sender is |
799 // destroyed, which is needed for the DTMF sender. | 799 // destroyed, which is needed for the DTMF sender. |
800 TEST_F(RtpSenderReceiverTest, TestOnDestroyedSignal) { | 800 TEST_F(RtpSenderReceiverTest, TestOnDestroyedSignal) { |
801 CreateAudioRtpSender(); | 801 CreateAudioRtpSender(); |
802 EXPECT_FALSE(audio_sender_destroyed_signal_fired_); | 802 EXPECT_FALSE(audio_sender_destroyed_signal_fired_); |
803 audio_rtp_sender_ = nullptr; | 803 audio_rtp_sender_ = nullptr; |
804 EXPECT_TRUE(audio_sender_destroyed_signal_fired_); | 804 EXPECT_TRUE(audio_sender_destroyed_signal_fired_); |
805 } | 805 } |
806 | 806 |
807 } // namespace webrtc | 807 } // namespace webrtc |
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