blob: 416cfe993cc29d959d94e318d85da626e994b697 [file] [log] [blame]
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include <memory>
#include <vector>
#include "webrtc/call.h"
#include "webrtc/call/transport_adapter.h"
#include "webrtc/common_video/include/incoming_video_stream.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/modules/video_coding/video_coding_impl.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/video/receive_statistics_proxy.h"
#include "webrtc/video/rtp_stream_receiver.h"
#include "webrtc/video/rtp_streams_synchronizer.h"
#include "webrtc/video/video_stream_decoder.h"
#include "webrtc/video_receive_stream.h"
namespace webrtc {
class CallStats;
class CongestionController;
class IvfFileWriter;
class ProcessThread;
class RTPFragmentationHeader;
class VoiceEngine;
class VieRemb;
namespace internal {
class VideoReceiveStream : public webrtc::VideoReceiveStream,
public rtc::VideoSinkInterface<VideoFrame>,
public EncodedImageCallback,
public NackSender,
public KeyFrameRequestSender {
VideoReceiveStream(int num_cpu_cores,
CongestionController* congestion_controller,
VideoReceiveStream::Config config,
webrtc::VoiceEngine* voice_engine,
ProcessThread* process_thread,
CallStats* call_stats,
VieRemb* remb);
~VideoReceiveStream() override;
void SignalNetworkState(NetworkState state);
bool DeliverRtcp(const uint8_t* packet, size_t length);
bool DeliverRtp(const uint8_t* packet,
size_t length,
const PacketTime& packet_time);
// webrtc::VideoReceiveStream implementation.
void Start() override;
void Stop() override;
webrtc::VideoReceiveStream::Stats GetStats() const override;
// Overrides rtc::VideoSinkInterface<VideoFrame>.
void OnFrame(const VideoFrame& video_frame) override;
// Overrides EncodedImageCallback.
EncodedImageCallback::Result OnEncodedImage(
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info,
const RTPFragmentationHeader* fragmentation) override;
const Config& config() const { return config_; }
void SetSyncChannel(VoiceEngine* voice_engine, int audio_channel_id);
// Implements NackSender.
void SendNack(const std::vector<uint16_t>& sequence_numbers) override;
// Implements KeyFrameRequestSender.
void RequestKeyFrame() override;
// Takes ownership of the file, is responsible for closing it later.
// Calling this method will close and finalize any current log.
// Giving rtc::kInvalidPlatformFileValue disables logging.
// If a frame to be written would make the log too large the write fails and
// the log is closed and finalized. A |byte_limit| of 0 means no limit.
void EnableEncodedFrameRecording(rtc::PlatformFile file,
size_t byte_limit) override;
static bool DecodeThreadFunction(void* ptr);
void Decode();
TransportAdapter transport_adapter_;
const VideoReceiveStream::Config config_;
ProcessThread* const process_thread_;
Clock* const clock_;
rtc::PlatformThread decode_thread_;
CongestionController* const congestion_controller_;
CallStats* const call_stats_;
vcm::VideoReceiver video_receiver_;
std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> incoming_video_stream_;
ReceiveStatisticsProxy stats_proxy_;
RtpStreamReceiver rtp_stream_receiver_;
std::unique_ptr<VideoStreamDecoder> video_stream_decoder_;
RtpStreamsSynchronizer rtp_stream_sync_;
rtc::CriticalSection ivf_writer_lock_;
std::unique_ptr<IvfFileWriter> ivf_writer_ GUARDED_BY(ivf_writer_lock_);
} // namespace internal
} // namespace webrtc