blob: 06fbb3676953eadf4ed776dddf5eb3362633402b [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 <map>
#include <memory>
#include <vector>
#include "webrtc/call/bitrate_allocator.h"
#include "webrtc/call/video_receive_stream.h"
#include "webrtc/call/video_send_stream.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/modules/video_coding/protection_bitrate_calculator.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/rtc_base/event.h"
#include "webrtc/rtc_base/task_queue.h"
#include "webrtc/video/encoder_rtcp_feedback.h"
#include "webrtc/video/send_delay_stats.h"
#include "webrtc/video/send_statistics_proxy.h"
#include "webrtc/video/video_stream_encoder.h"
namespace webrtc {
class CallStats;
class SendSideCongestionController;
class IvfFileWriter;
class ProcessThread;
class RtpRtcp;
class RtpTransportControllerSendInterface;
class RtcEventLog;
namespace internal {
class VideoSendStreamImpl;
// VideoSendStream implements webrtc::VideoSendStream.
// Internally, it delegates all public methods to VideoSendStreamImpl and / or
// VideoStreamEncoder. VideoSendStreamInternal is created and deleted on
// |worker_queue|.
class VideoSendStream : public webrtc::VideoSendStream {
VideoSendStream(int num_cpu_cores,
ProcessThread* module_process_thread,
rtc::TaskQueue* worker_queue,
CallStats* call_stats,
RtpTransportControllerSendInterface* transport,
BitrateAllocator* bitrate_allocator,
SendDelayStats* send_delay_stats,
RtcEventLog* event_log,
VideoSendStream::Config config,
VideoEncoderConfig encoder_config,
const std::map<uint32_t, RtpState>& suspended_ssrcs);
~VideoSendStream() override;
void SignalNetworkState(NetworkState state);
bool DeliverRtcp(const uint8_t* packet, size_t length);
// webrtc::VideoSendStream implementation.
void Start() override;
void Stop() override;
void SetSource(rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
const DegradationPreference& degradation_preference) override;
void ReconfigureVideoEncoder(VideoEncoderConfig) override;
Stats GetStats() override;
typedef std::map<uint32_t, RtpState> RtpStateMap;
// Takes ownership of each file, is responsible for closing them later.
// Calling this method will close and finalize any current logs.
// Giving rtc::kInvalidPlatformFileValue in any position disables logging
// for the corresponding stream.
// 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(const std::vector<rtc::PlatformFile>& files,
size_t byte_limit) override;
RtpStateMap StopPermanentlyAndGetRtpStates();
void SetTransportOverhead(size_t transport_overhead_per_packet);
class ConstructionTask;
class DestructAndGetRtpStateTask;
rtc::ThreadChecker thread_checker_;
rtc::TaskQueue* const worker_queue_;
rtc::Event thread_sync_event_;
SendStatisticsProxy stats_proxy_;
const VideoSendStream::Config config_;
const VideoEncoderConfig::ContentType content_type_;
std::unique_ptr<VideoSendStreamImpl> send_stream_;
std::unique_ptr<VideoStreamEncoder> video_stream_encoder_;
} // namespace internal
} // namespace webrtc