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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
commit | 40a355a42d4a9444dc753c04c6608dade2f06a23 (patch) | |
tree | 871fc667d2de662f171103ce5ec067014ef85e61 /third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc | |
parent | Adding upstream version 124.0.1. (diff) | |
download | firefox-40a355a42d4a9444dc753c04c6608dade2f06a23.tar.xz firefox-40a355a42d4a9444dc753c04c6608dade2f06a23.zip |
Adding upstream version 125.0.1.upstream/125.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc | 15 |
1 files changed, 15 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc index 0db610c149..724cd3a5e0 100644 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc @@ -222,4 +222,19 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) { EXPECT_FALSE(transport_.last_sent_packet().Marker()); } +TEST_F(RtpSenderAudioTest, SendsCsrcs) { + const char payload_name[] = "audio"; + const uint8_t payload_type = 127; + ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload( + payload_name, payload_type, 48000, 0, 1500)); + uint8_t payload[] = {47, 11, 32, 93, 89}; + + std::vector<uint32_t> csrcs({123, 456, 789}); + + ASSERT_TRUE(rtp_sender_audio_->SendAudio( + {.payload = payload, .payload_id = payload_type, .csrcs = csrcs})); + + EXPECT_EQ(transport_.last_sent_packet().Csrcs(), csrcs); +} + } // namespace webrtc |