9d3ab61325
Trying to submit all changes at once proved impossible since there were too many changes in too many files. The changes to PRESUBMIT.py will be uploaded in the last CL. (original CL: https://codereview.webrtc.org/1528503003/) BUG=webrtc:5309 TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1543503002 Cr-Commit-Position: refs/heads/master@{#11102}
93 lines
3.2 KiB
C++
93 lines
3.2 KiB
C++
/*
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* Copyright (c) 2011 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_MODULES_VIDEO_CODING_RECEIVER_H_
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#define WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_
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#include <vector>
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#include "webrtc/modules/video_coding/jitter_buffer.h"
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#include "webrtc/modules/video_coding/packet.h"
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#include "webrtc/modules/video_coding/timing.h"
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#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
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#include "webrtc/modules/video_coding/include/video_coding.h"
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#include "webrtc/modules/video_coding/include/video_coding_defines.h"
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namespace webrtc {
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class Clock;
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class VCMEncodedFrame;
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class VCMReceiver {
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public:
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VCMReceiver(VCMTiming* timing, Clock* clock, EventFactory* event_factory);
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// Using this constructor, you can specify a different event factory for the
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// jitter buffer. Useful for unit tests when you want to simulate incoming
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// packets, in which case the jitter buffer's wait event is different from
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// that of VCMReceiver itself.
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VCMReceiver(VCMTiming* timing,
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Clock* clock,
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rtc::scoped_ptr<EventWrapper> receiver_event,
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rtc::scoped_ptr<EventWrapper> jitter_buffer_event);
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~VCMReceiver();
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void Reset();
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void UpdateRtt(int64_t rtt);
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int32_t InsertPacket(const VCMPacket& packet,
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uint16_t frame_width,
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uint16_t frame_height);
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VCMEncodedFrame* FrameForDecoding(uint16_t max_wait_time_ms,
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int64_t* next_render_time_ms,
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bool prefer_late_decoding);
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void ReleaseFrame(VCMEncodedFrame* frame);
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void ReceiveStatistics(uint32_t* bitrate, uint32_t* framerate);
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uint32_t DiscardedPackets() const;
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// NACK.
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void SetNackMode(VCMNackMode nackMode,
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int64_t low_rtt_nack_threshold_ms,
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int64_t high_rtt_nack_threshold_ms);
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void SetNackSettings(size_t max_nack_list_size,
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int max_packet_age_to_nack,
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int max_incomplete_time_ms);
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VCMNackMode NackMode() const;
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std::vector<uint16_t> NackList(bool* request_key_frame);
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// Receiver video delay.
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int SetMinReceiverDelay(int desired_delay_ms);
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// Decoding with errors.
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void SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode);
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VCMDecodeErrorMode DecodeErrorMode() const;
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// Returns size in time (milliseconds) of complete continuous frames in the
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// jitter buffer. The render time is estimated based on the render delay at
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// the time this function is called.
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int RenderBufferSizeMs();
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void RegisterStatsCallback(VCMReceiveStatisticsCallback* callback);
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void TriggerDecoderShutdown();
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private:
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CriticalSectionWrapper* crit_sect_;
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Clock* const clock_;
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VCMJitterBuffer jitter_buffer_;
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VCMTiming* timing_;
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rtc::scoped_ptr<EventWrapper> render_wait_event_;
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int max_video_delay_ms_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_
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