| Index: content/renderer/media/webrtc/rtc_rtp_contributing_source.cc
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| diff --git a/content/renderer/media/webrtc/rtc_rtp_contributing_source.cc b/content/renderer/media/webrtc/rtc_rtp_contributing_source.cc
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| deleted file mode 100644
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| index b259b9c52f4672e7bf5940cd6cfd9533c46d8845..0000000000000000000000000000000000000000
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| --- a/content/renderer/media/webrtc/rtc_rtp_contributing_source.cc
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| +++ /dev/null
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| @@ -1,40 +0,0 @@
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| -// Copyright (c) 2017 The Chromium Authors. All rights reserved.
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| -// Use of this source code is governed by a BSD-style license that can be
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| -// found in the LICENSE file.
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| -
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| -#include "content/renderer/media/webrtc/rtc_rtp_contributing_source.h"
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| -
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| -#include "base/logging.h"
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| -#include "base/time/time.h"
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| -#include "third_party/webrtc/base/scoped_ref_ptr.h"
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| -
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| -namespace content {
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| -
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| -RTCRtpContributingSource::RTCRtpContributingSource(
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| -    const webrtc::RtpSource& source)
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| -    : source_(source) {}
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| -
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| -RTCRtpContributingSource::~RTCRtpContributingSource() {}
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| -
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| -blink::WebRTCRtpContributingSourceType RTCRtpContributingSource::SourceType()
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| -    const {
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| -  switch (source_.source_type()) {
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| -    case webrtc::RtpSourceType::SSRC:
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| -      return blink::WebRTCRtpContributingSourceType::SSRC;
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| -    case webrtc::RtpSourceType::CSRC:
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| -      return blink::WebRTCRtpContributingSourceType::CSRC;
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| -    default:
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| -      NOTREACHED();
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| -      return blink::WebRTCRtpContributingSourceType::SSRC;
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| -  }
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| -}
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| -
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| -double RTCRtpContributingSource::TimestampMs() const {
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| -  return source_.timestamp_ms();
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| -}
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| -
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| -uint32_t RTCRtpContributingSource::Source() const {
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| -  return source_.source_id();
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| -}
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| -
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| -}  // namespace content
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| 
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