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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
54 lines
1.9 KiB
C++
54 lines
1.9 KiB
C++
/*
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* Copyright 2019 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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// This is EXPERIMENTAL interface for media transport.
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//
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// The goal is to refactor WebRTC code so that audio and video frames
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// are sent / received through the media transport interface. This will
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// enable different media transport implementations, including QUIC-based
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// media transport.
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#include "api/transport/media/audio_transport.h"
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#include <utility>
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namespace webrtc {
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MediaTransportEncodedAudioFrame::~MediaTransportEncodedAudioFrame() {}
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MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
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int sampling_rate_hz,
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int starting_sample_index,
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int samples_per_channel,
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int sequence_number,
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FrameType frame_type,
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int payload_type,
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std::vector<uint8_t> encoded_data)
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: sampling_rate_hz_(sampling_rate_hz),
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starting_sample_index_(starting_sample_index),
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samples_per_channel_(samples_per_channel),
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sequence_number_(sequence_number),
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frame_type_(frame_type),
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payload_type_(payload_type),
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encoded_data_(std::move(encoded_data)) {}
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MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
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const MediaTransportEncodedAudioFrame&) = default;
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MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
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MediaTransportEncodedAudioFrame&&) = default;
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MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
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const MediaTransportEncodedAudioFrame&) = default;
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MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
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MediaTransportEncodedAudioFrame&&) = default;
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} // namespace webrtc
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