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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}
117 lines
3.5 KiB
C++
117 lines
3.5 KiB
C++
/*
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* Copyright (c) 2015 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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*/
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#ifndef MODULES_VIDEO_CODING_CODECS_H264_H264_ENCODER_IMPL_H_
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#define MODULES_VIDEO_CODING_CODECS_H264_H264_ENCODER_IMPL_H_
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// Everything declared in this header is only required when WebRTC is
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// build with H264 support, please do not move anything out of the
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// #ifdef unless needed and tested.
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#ifdef WEBRTC_USE_H264
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#if defined(WEBRTC_WIN) && !defined(__clang__)
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#error "See: bugs.webrtc.org/9213#c13."
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#endif
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#include <memory>
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#include <vector>
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#include "api/video/i420_buffer.h"
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#include "api/video_codecs/video_encoder.h"
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#include "common_video/h264/h264_bitstream_parser.h"
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#include "modules/video_coding/codecs/h264/include/h264.h"
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#include "modules/video_coding/utility/quality_scaler.h"
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#include "third_party/openh264/src/codec/api/svc/codec_app_def.h"
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class ISVCEncoder;
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namespace webrtc {
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class H264EncoderImpl : public H264Encoder {
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public:
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struct LayerConfig {
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int simulcast_idx = 0;
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int width = -1;
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int height = -1;
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bool sending = true;
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bool key_frame_request = false;
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float max_frame_rate = 0;
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uint32_t target_bps = 0;
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uint32_t max_bps = 0;
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bool frame_dropping_on = false;
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int key_frame_interval = 0;
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int num_temporal_layers = 1;
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void SetStreamState(bool send_stream);
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};
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public:
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explicit H264EncoderImpl(const cricket::VideoCodec& codec);
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~H264EncoderImpl() override;
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// |settings.max_payload_size| is ignored.
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// The following members of |codec_settings| are used. The rest are ignored.
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// - codecType (must be kVideoCodecH264)
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// - targetBitrate
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// - maxFramerate
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// - width
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// - height
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int32_t InitEncode(const VideoCodec* codec_settings,
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const VideoEncoder::Settings& settings) override;
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int32_t Release() override;
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int32_t RegisterEncodeCompleteCallback(
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EncodedImageCallback* callback) override;
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void SetRates(const RateControlParameters& parameters) override;
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// The result of encoding - an EncodedImage and RTPFragmentationHeader - are
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// passed to the encode complete callback.
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int32_t Encode(const VideoFrame& frame,
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const std::vector<VideoFrameType>* frame_types) override;
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EncoderInfo GetEncoderInfo() const override;
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// Exposed for testing.
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H264PacketizationMode PacketizationModeForTesting() const {
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return packetization_mode_;
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}
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private:
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SEncParamExt CreateEncoderParams(size_t i) const;
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webrtc::H264BitstreamParser h264_bitstream_parser_;
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// Reports statistics with histograms.
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void ReportInit();
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void ReportError();
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std::vector<ISVCEncoder*> encoders_;
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std::vector<SSourcePicture> pictures_;
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std::vector<rtc::scoped_refptr<I420Buffer>> downscaled_buffers_;
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std::vector<LayerConfig> configurations_;
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std::vector<EncodedImage> encoded_images_;
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VideoCodec codec_;
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H264PacketizationMode packetization_mode_;
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size_t max_payload_size_;
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int32_t number_of_cores_;
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EncodedImageCallback* encoded_image_callback_;
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bool has_reported_init_;
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bool has_reported_error_;
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std::vector<uint8_t> tl0sync_limit_;
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};
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} // namespace webrtc
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#endif // WEBRTC_USE_H264
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#endif // MODULES_VIDEO_CODING_CODECS_H264_H264_ENCODER_IMPL_H_
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