Passing the estimated capture clock offset to SendVideo.
Bug: webrtc:10739 Change-Id: I491db1910fad9101c7c9087e880862e755dfc69d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182184 Reviewed-by: Chen Xing <chxg@google.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31983}
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@ -399,7 +399,8 @@ bool RTPSenderVideo::SendVideo(
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int64_t capture_time_ms,
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rtc::ArrayView<const uint8_t> payload,
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RTPVideoHeader video_header,
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absl::optional<int64_t> expected_retransmission_time_ms) {
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absl::optional<int64_t> expected_retransmission_time_ms,
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absl::optional<int64_t> estimated_capture_clock_offset_ms) {
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#if RTC_TRACE_EVENTS_ENABLED
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TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms, "Send", "type",
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FrameTypeToString(video_header.frame_type));
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@ -452,7 +453,7 @@ bool RTPSenderVideo::SendVideo(
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single_packet->Timestamp(), kVideoPayloadTypeFrequency,
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Int64MsToUQ32x32(single_packet->capture_time_ms() + NtpOffsetMs()),
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/*estimated_capture_clock_offset=*/
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include_capture_clock_offset_ ? absl::make_optional(0)
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include_capture_clock_offset_ ? estimated_capture_clock_offset_ms
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: absl::nullopt);
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auto first_packet = std::make_unique<RtpPacketToSend>(*single_packet);
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@ -91,13 +91,19 @@ class RTPSenderVideo {
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// expected_retransmission_time_ms.has_value() -> retransmission allowed.
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// Calls to this method is assumed to be externally serialized.
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// |estimated_capture_clock_offset_ms| is an estimated clock offset between
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// this sender and the original capturer, for this video packet. See
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// http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time for more
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// details. If the sender and the capture has the same clock, it is supposed
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// to be zero valued, which is given as the default.
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bool SendVideo(int payload_type,
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absl::optional<VideoCodecType> codec_type,
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uint32_t rtp_timestamp,
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int64_t capture_time_ms,
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rtc::ArrayView<const uint8_t> payload,
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RTPVideoHeader video_header,
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absl::optional<int64_t> expected_retransmission_time_ms);
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absl::optional<int64_t> expected_retransmission_time_ms,
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absl::optional<int64_t> estimated_capture_clock_offset_ms = 0);
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bool SendEncodedImage(
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int payload_type,
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@ -867,10 +867,12 @@ TEST_P(RtpSenderVideoTest, AbsoluteCaptureTimeWithCaptureClockOffset) {
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kAbsoluteCaptureTimeExtensionId);
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RTPVideoHeader hdr;
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const absl::optional<int64_t> kExpectedCaptureClockOffset =
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absl::make_optional(1234);
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hdr.frame_type = VideoFrameType::kVideoFrameKey;
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rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp,
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kAbsoluteCaptureTimestampMs, kFrame, hdr,
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kDefaultExpectedRetransmissionTimeMs);
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rtp_sender_video_->SendVideo(
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kPayload, kType, kTimestamp, kAbsoluteCaptureTimestampMs, kFrame, hdr,
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kDefaultExpectedRetransmissionTimeMs, kExpectedCaptureClockOffset);
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// It is expected that one and only one of the packets sent on this video
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// frame has absolute capture time header extension. And it includes capture
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@ -883,9 +885,8 @@ TEST_P(RtpSenderVideoTest, AbsoluteCaptureTimeWithCaptureClockOffset) {
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++packets_with_abs_capture_time;
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EXPECT_EQ(absolute_capture_time->absolute_capture_timestamp,
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Int64MsToUQ32x32(kAbsoluteCaptureTimestampMs + NtpOffsetMs()));
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EXPECT_TRUE(
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absolute_capture_time->estimated_capture_clock_offset.has_value());
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EXPECT_EQ(0, *absolute_capture_time->estimated_capture_clock_offset);
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EXPECT_EQ(kExpectedCaptureClockOffset,
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absolute_capture_time->estimated_capture_clock_offset);
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}
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}
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EXPECT_EQ(packets_with_abs_capture_time, 1);
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