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b5d918324c
RTP timestamp was recently added to contributing sources in the WebRTC specification. This CL implements that change in WebRTC. Bug: webrtc:10650 Change-Id: Ic0ccfbea7049a5b66063fa6cf60d01d5bd713132 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137515 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28020}
74 lines
2.2 KiB
C++
74 lines
2.2 KiB
C++
/*
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* Copyright 2018 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "api/rtp_receiver_interface.h"
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namespace webrtc {
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RtpSource::RtpSource(int64_t timestamp_ms,
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uint32_t source_id,
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RtpSourceType source_type,
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absl::optional<uint8_t> audio_level,
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uint32_t rtp_timestamp)
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: timestamp_ms_(timestamp_ms),
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source_id_(source_id),
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source_type_(source_type),
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audio_level_(audio_level),
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rtp_timestamp_(rtp_timestamp) {}
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RtpSource::RtpSource(int64_t timestamp_ms,
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uint32_t source_id,
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RtpSourceType source_type)
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: timestamp_ms_(timestamp_ms),
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source_id_(source_id),
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source_type_(source_type),
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rtp_timestamp_(0) {}
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RtpSource::RtpSource(int64_t timestamp_ms,
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uint32_t source_id,
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RtpSourceType source_type,
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uint8_t audio_level)
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: timestamp_ms_(timestamp_ms),
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source_id_(source_id),
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source_type_(source_type),
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audio_level_(audio_level),
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rtp_timestamp_(0) {}
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RtpSource::RtpSource(const RtpSource&) = default;
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RtpSource& RtpSource::operator=(const RtpSource&) = default;
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RtpSource::~RtpSource() = default;
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std::vector<std::string> RtpReceiverInterface::stream_ids() const {
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return {};
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}
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std::vector<rtc::scoped_refptr<MediaStreamInterface>>
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RtpReceiverInterface::streams() const {
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return {};
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}
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std::vector<RtpSource> RtpReceiverInterface::GetSources() const {
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return {};
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}
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void RtpReceiverInterface::SetFrameDecryptor(
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rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {}
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rtc::scoped_refptr<FrameDecryptorInterface>
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RtpReceiverInterface::GetFrameDecryptor() const {
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return nullptr;
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}
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rtc::scoped_refptr<DtlsTransportInterface>
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RtpReceiverInterface::dtls_transport() const {
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return nullptr;
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}
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} // namespace webrtc
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