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4bac79ece2
This change is required to allow modification of Jitter Buffer delay in javascript via Origin Trial Experiment. Link to experiment description: https://groups.google.com/a/chromium.org/forum/#!topic/blink-dev/Tgm4qiNepJc Bug: webrtc:10287 Change-Id: I4f21380aad5982a4a60c55683b5173ce72ce0392 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131144 Commit-Queue: Ruslan Burakov <kuddai@google.com> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27444}
64 lines
1.9 KiB
C++
64 lines
1.9 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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: timestamp_ms_(timestamp_ms),
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source_id_(source_id),
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source_type_(source_type) {}
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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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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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void RtpReceiverInterface::SetJitterBufferMinimumDelay(
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absl::optional<double> delay_seconds) {}
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} // namespace webrtc
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