d52a1a6971
This reverts commit 1811c04f22a26da3ed2832373a5c92a9786420c3. Reason for revert: Downstream projects fixed. Original change's description: > Revert "Remove RTPVideoHeader::vp8() accessors." > > This reverts commit af7afc66427b0e9109e7d492f2805d63d239b914. > > Reason for revert: Break downstream projects. > > Original change's description: > > Remove RTPVideoHeader::vp8() accessors. > > > > Bug: none > > Change-Id: Ia7d65148fb36a8f26647bee8a876ce7217ff8a68 > > Reviewed-on: https://webrtc-review.googlesource.com/93321 > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#24626} > > TBR=danilchap@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,holmer@google.com > > Change-Id: I3f7f19c0ea810c0fd988c59e6556bbea9b756b33 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: none > Reviewed-on: https://webrtc-review.googlesource.com/98864 > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24628} TBR=danilchap@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,holmer@google.com Change-Id: I9246f36e638108ae4fc46c1ae4559c8205d50fc1 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: none Reviewed-on: https://webrtc-review.googlesource.com/98841 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24629}
94 lines
2.9 KiB
C++
94 lines
2.9 KiB
C++
/*
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* Copyright (c) 2013 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 "test/configurable_frame_size_encoder.h"
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#include <string.h>
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#include "common_video/include/video_frame.h"
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#include "modules/video_coding/include/video_codec_interface.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace test {
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ConfigurableFrameSizeEncoder::ConfigurableFrameSizeEncoder(
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size_t max_frame_size)
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: callback_(NULL),
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max_frame_size_(max_frame_size),
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current_frame_size_(max_frame_size),
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buffer_(new uint8_t[max_frame_size]),
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codec_type_(kVideoCodecGeneric) {
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memset(buffer_.get(), 0, max_frame_size);
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}
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ConfigurableFrameSizeEncoder::~ConfigurableFrameSizeEncoder() {}
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int32_t ConfigurableFrameSizeEncoder::InitEncode(
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const VideoCodec* codec_settings,
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int32_t number_of_cores,
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size_t max_payload_size) {
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return WEBRTC_VIDEO_CODEC_OK;
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}
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int32_t ConfigurableFrameSizeEncoder::Encode(
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const VideoFrame& inputImage,
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const CodecSpecificInfo* codecSpecificInfo,
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const std::vector<FrameType>* frame_types) {
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EncodedImage encodedImage(buffer_.get(), current_frame_size_,
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max_frame_size_);
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encodedImage._completeFrame = true;
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encodedImage._encodedHeight = inputImage.height();
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encodedImage._encodedWidth = inputImage.width();
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encodedImage._frameType = kVideoFrameKey;
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encodedImage.SetTimestamp(inputImage.timestamp());
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encodedImage.capture_time_ms_ = inputImage.render_time_ms();
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RTPFragmentationHeader* fragmentation = NULL;
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CodecSpecificInfo specific{};
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specific.codecType = codec_type_;
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callback_->OnEncodedImage(encodedImage, &specific, fragmentation);
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return WEBRTC_VIDEO_CODEC_OK;
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}
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int32_t ConfigurableFrameSizeEncoder::RegisterEncodeCompleteCallback(
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EncodedImageCallback* callback) {
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callback_ = callback;
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return WEBRTC_VIDEO_CODEC_OK;
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}
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int32_t ConfigurableFrameSizeEncoder::Release() {
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return WEBRTC_VIDEO_CODEC_OK;
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}
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int32_t ConfigurableFrameSizeEncoder::SetChannelParameters(uint32_t packet_loss,
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int64_t rtt) {
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return WEBRTC_VIDEO_CODEC_OK;
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}
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int32_t ConfigurableFrameSizeEncoder::SetRateAllocation(
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const VideoBitrateAllocation& allocation,
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uint32_t framerate) {
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return WEBRTC_VIDEO_CODEC_OK;
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}
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int32_t ConfigurableFrameSizeEncoder::SetFrameSize(size_t size) {
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RTC_DCHECK_LE(size, max_frame_size_);
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current_frame_size_ = size;
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return WEBRTC_VIDEO_CODEC_OK;
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}
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void ConfigurableFrameSizeEncoder::SetCodecType(VideoCodecType codec_type) {
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codec_type_ = codec_type;
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}
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} // namespace test
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} // namespace webrtc
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