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/*
* Copyright 2019 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// This is EXPERIMENTAL interface for media transport.
//
// The goal is to refactor WebRTC code so that audio and video frames
// are sent / received through the media transport interface. This will
// enable different media transport implementations, including QUIC-based
// media transport.
#include "api/transport/media/audio_transport.h"
#include <utility>
namespace webrtc {
MediaTransportEncodedAudioFrame::~MediaTransportEncodedAudioFrame() {}
MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
int sampling_rate_hz,
int starting_sample_index,
int samples_per_channel,
int sequence_number,
FrameType frame_type,
int payload_type,
std::vector<uint8_t> encoded_data)
: sampling_rate_hz_(sampling_rate_hz),
starting_sample_index_(starting_sample_index),
samples_per_channel_(samples_per_channel),
sequence_number_(sequence_number),
frame_type_(frame_type),
payload_type_(payload_type),
encoded_data_(std::move(encoded_data)) {}
MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
const MediaTransportEncodedAudioFrame&) = default;
MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
MediaTransportEncodedAudioFrame&&) = default;
MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
const MediaTransportEncodedAudioFrame&) = default;
MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
MediaTransportEncodedAudioFrame&&) = default;
} // namespace webrtc
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