This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
202 lines
7.0 KiB
C++
202 lines
7.0 KiB
C++
/*
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* Copyright (c) 2012 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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#ifndef WEBRTC_VIDEO_ENGINE_VIE_ENCODER_H_
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#define WEBRTC_VIDEO_ENGINE_VIE_ENCODER_H_
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#include <map>
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#include <vector>
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/base/scoped_ref_ptr.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/common_types.h"
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#include "webrtc/frame_callback.h"
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#include "webrtc/modules/bitrate_controller/include/bitrate_allocator.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
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#include "webrtc/modules/video_processing/main/interface/video_processing.h"
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#include "webrtc/typedefs.h"
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#include "webrtc/video/video_capture_input.h"
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#include "webrtc/video_engine/vie_defines.h"
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namespace webrtc {
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class Config;
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class CriticalSectionWrapper;
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class EncodedImageCallback;
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class PacedSender;
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class PayloadRouter;
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class ProcessThread;
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class QMVideoSettingsCallback;
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class SendStatisticsProxy;
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class ViEBitrateObserver;
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class ViEEffectFilter;
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class VideoCodingModule;
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class ViEEncoder : public RtcpIntraFrameObserver,
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public VideoEncoderRateObserver,
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public VCMPacketizationCallback,
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public VCMSendStatisticsCallback,
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public VideoCaptureCallback {
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public:
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friend class ViEBitrateObserver;
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ViEEncoder(uint32_t number_of_cores,
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ProcessThread* module_process_thread,
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SendStatisticsProxy* stats_proxy,
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I420FrameCallback* pre_encode_callback,
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PacedSender* pacer,
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BitrateAllocator* bitrate_allocator);
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~ViEEncoder();
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bool Init();
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// This function is assumed to be called before any frames are delivered and
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// only once.
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// Ideally this would be done in Init, but the dependencies between ViEEncoder
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// and ViEChannel makes it really hard to do in a good way.
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void StartThreadsAndSetSharedMembers(
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rtc::scoped_refptr<PayloadRouter> send_payload_router,
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VCMProtectionCallback* vcm_protection_callback);
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// This function must be called before the corresponding ViEChannel is
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// deleted.
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void StopThreadsAndRemoveSharedMembers();
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void SetNetworkTransmissionState(bool is_transmitting);
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// Returns the id of the owning channel.
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int Owner() const;
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// Drops incoming packets before they get to the encoder.
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void Pause();
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void Restart();
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// Codec settings.
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uint8_t NumberOfCodecs();
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int32_t GetCodec(uint8_t list_index, VideoCodec* video_codec);
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int32_t RegisterExternalEncoder(VideoEncoder* encoder,
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uint8_t pl_type,
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bool internal_source);
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int32_t DeRegisterExternalEncoder(uint8_t pl_type);
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int32_t SetEncoder(const VideoCodec& video_codec);
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int32_t GetEncoder(VideoCodec* video_codec);
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// Scale or crop/pad image.
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int32_t ScaleInputImage(bool enable);
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// Implementing VideoCaptureCallback.
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void DeliverFrame(VideoFrame video_frame) override;
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int32_t SendKeyFrame();
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uint32_t LastObservedBitrateBps() const;
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int CodecTargetBitrate(uint32_t* bitrate) const;
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// Loss protection.
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int32_t UpdateProtectionMethod(bool nack, bool fec);
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bool nack_enabled() const { return nack_enabled_; }
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// Buffering mode.
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void SetSenderBufferingMode(int target_delay_ms);
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// Implements VideoEncoderRateObserver.
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void OnSetRates(uint32_t bitrate_bps, int framerate) override;
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// Implements VCMPacketizationCallback.
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int32_t SendData(uint8_t payload_type,
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const EncodedImage& encoded_image,
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const RTPFragmentationHeader& fragmentation_header,
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const RTPVideoHeader* rtp_video_hdr) override;
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// Implements VideoSendStatisticsCallback.
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int32_t SendStatistics(const uint32_t bit_rate,
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const uint32_t frame_rate) override;
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// Implements RtcpIntraFrameObserver.
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void OnReceivedIntraFrameRequest(uint32_t ssrc) override;
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void OnReceivedSLI(uint32_t ssrc, uint8_t picture_id) override;
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void OnReceivedRPSI(uint32_t ssrc, uint64_t picture_id) override;
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void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) override;
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// Sets SSRCs for all streams.
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bool SetSsrcs(const std::vector<uint32_t>& ssrcs);
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void SetMinTransmitBitrate(int min_transmit_bitrate_kbps);
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// Lets the sender suspend video when the rate drops below
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// |threshold_bps|, and turns back on when the rate goes back up above
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// |threshold_bps| + |window_bps|.
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void SuspendBelowMinBitrate();
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// New-style callbacks, used by VideoSendStream.
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void RegisterPostEncodeImageCallback(
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EncodedImageCallback* post_encode_callback);
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int GetPaddingNeededBps() const;
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protected:
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// Called by BitrateObserver.
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void OnNetworkChanged(uint32_t bitrate_bps,
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uint8_t fraction_lost,
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int64_t round_trip_time_ms);
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private:
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bool EncoderPaused() const EXCLUSIVE_LOCKS_REQUIRED(data_cs_);
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void TraceFrameDropStart() EXCLUSIVE_LOCKS_REQUIRED(data_cs_);
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void TraceFrameDropEnd() EXCLUSIVE_LOCKS_REQUIRED(data_cs_);
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const uint32_t number_of_cores_;
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const rtc::scoped_ptr<VideoProcessingModule> vpm_;
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const rtc::scoped_ptr<QMVideoSettingsCallback> qm_callback_;
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const rtc::scoped_ptr<VideoCodingModule> vcm_;
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rtc::scoped_refptr<PayloadRouter> send_payload_router_;
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rtc::scoped_ptr<CriticalSectionWrapper> data_cs_;
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rtc::scoped_ptr<BitrateObserver> bitrate_observer_;
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SendStatisticsProxy* const stats_proxy_;
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I420FrameCallback* const pre_encode_callback_;
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PacedSender* const pacer_;
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BitrateAllocator* const bitrate_allocator_;
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// The time we last received an input frame or encoded frame. This is used to
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// track when video is stopped long enough that we also want to stop sending
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// padding.
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int64_t time_of_last_frame_activity_ms_ GUARDED_BY(data_cs_);
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bool simulcast_enabled_ GUARDED_BY(data_cs_);
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int min_transmit_bitrate_kbps_ GUARDED_BY(data_cs_);
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uint32_t last_observed_bitrate_bps_ GUARDED_BY(data_cs_);
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int target_delay_ms_ GUARDED_BY(data_cs_);
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bool network_is_transmitting_ GUARDED_BY(data_cs_);
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bool encoder_paused_ GUARDED_BY(data_cs_);
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bool encoder_paused_and_dropped_frame_ GUARDED_BY(data_cs_);
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std::map<unsigned int, int64_t> time_last_intra_request_ms_
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GUARDED_BY(data_cs_);
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bool fec_enabled_;
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bool nack_enabled_;
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ProcessThread* module_process_thread_;
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bool has_received_sli_ GUARDED_BY(data_cs_);
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uint8_t picture_id_sli_ GUARDED_BY(data_cs_);
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bool has_received_rpsi_ GUARDED_BY(data_cs_);
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uint64_t picture_id_rpsi_ GUARDED_BY(data_cs_);
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std::map<uint32_t, int> ssrc_streams_ GUARDED_BY(data_cs_);
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bool video_suspended_ GUARDED_BY(data_cs_);
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_VIE_ENCODER_H_
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