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authorChih-hung Hsieh <chh@google.com>2015-12-01 17:07:48 +0000
committerandroid-build-merger <android-build-merger@google.com>2015-12-01 17:07:48 +0000
commita4acd9d6bc9b3b033d7d274316e75ee067df8d20 (patch)
tree672a185b294789cf991f385c3e395dd63bea9063 /webrtc/video_engine/vie_channel.h
parent3681b90ba4fe7a27232dd3e27897d5d7ed9d651c (diff)
parentfe8b4a657979b49e1701bd92f6d5814a99e0b2be (diff)
downloadwebrtc-a4acd9d6bc9b3b033d7d274316e75ee067df8d20.tar.gz
Merge changes I7bbf776e,I1b827825
am: fe8b4a6579 * commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits) WIP: Changes after merge commit 'cb3f9bd' Make the nonlinear beamformer steerable Utilize bitrate above codec max to protect video. Enable VP9 internal resize by default. Filter overlapping RTP header extensions. Make VCMEncodedFrameCallback const. MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback. Remove redudant encoder rate calls. Create isolate files for nonparallel tests. Register header extensions in RtpRtcpObserver to avoid log spam. Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table ACM: Move NACK functionality inside NetEq Fix chromium-style warnings in webrtc/sound/. Create a 'webrtc_nonparallel_tests' target. Update scalability structure data according to updates in the RTP payload profile. audio_coding: rename interface -> include Rewrote perform_action_on_all_files to be parallell. Update reference indices according to updates in the RTP payload profile. Disable P2PTransport...TestFailoverControlledSide on Memcheck pass clangcl compile options to ignore warnings in gflags.cc ...
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diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
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+/*
+ * Copyright (c) 2012 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.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
+#define WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
+
+#include <list>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/typedefs.h"
+#include "webrtc/video_engine/vie_defines.h"
+#include "webrtc/video_engine/vie_receiver.h"
+#include "webrtc/video_engine/vie_sync_module.h"
+
+namespace webrtc {
+
+class CallStatsObserver;
+class ChannelStatsObserver;
+class Config;
+class CriticalSectionWrapper;
+class EncodedImageCallback;
+class I420FrameCallback;
+class IncomingVideoStream;
+class PacedSender;
+class PacketRouter;
+class PayloadRouter;
+class ProcessThread;
+class ReceiveStatisticsProxy;
+class ReportBlockStats;
+class RtcpRttStats;
+class ThreadWrapper;
+class ViEChannelProtectionCallback;
+class ViERTPObserver;
+class VideoCodingModule;
+class VideoDecoder;
+class VideoRenderCallback;
+class VoEVideoSync;
+
+enum StreamType {
+ kViEStreamTypeNormal = 0, // Normal media stream
+ kViEStreamTypeRtx = 1 // Retransmission media stream
+};
+
+class ViEChannel : public VCMFrameTypeCallback,
+ public VCMReceiveCallback,
+ public VCMReceiveStatisticsCallback,
+ public VCMDecoderTimingCallback,
+ public VCMPacketRequestCallback,
+ public RtpFeedback {
+ public:
+ friend class ChannelStatsObserver;
+ friend class ViEChannelProtectionCallback;
+
+ ViEChannel(uint32_t number_of_cores,
+ Transport* transport,
+ ProcessThread* module_process_thread,
+ RtcpIntraFrameObserver* intra_frame_observer,
+ RtcpBandwidthObserver* bandwidth_observer,
+ TransportFeedbackObserver* transport_feedback_observer,
+ RemoteBitrateEstimator* remote_bitrate_estimator,
+ RtcpRttStats* rtt_stats,
+ PacedSender* paced_sender,
+ PacketRouter* packet_router,
+ size_t max_rtp_streams,
+ bool sender);
+ ~ViEChannel();
+
+ int32_t Init();
+
+ // Sets the encoder to use for the channel. |new_stream| indicates the encoder
+ // type has changed and we should start a new RTP stream.
+ int32_t SetSendCodec(const VideoCodec& video_codec, bool new_stream = true);
+ int32_t SetReceiveCodec(const VideoCodec& video_codec);
+ // Registers an external decoder. |buffered_rendering| means that the decoder
+ // will render frames after decoding according to the render timestamp
+ // provided by the video coding module. |render_delay| indicates the time
+ // needed to decode and render a frame.
+ int32_t RegisterExternalDecoder(const uint8_t pl_type,
+ VideoDecoder* decoder,
+ bool buffered_rendering,
+ int32_t render_delay);
+ int32_t DeRegisterExternalDecoder(const uint8_t pl_type);
+ int32_t ReceiveCodecStatistics(uint32_t* num_key_frames,
+ uint32_t* num_delta_frames);
+ uint32_t DiscardedPackets() const;
+
+ // Returns the estimated delay in milliseconds.
+ int ReceiveDelay() const;
+
+ void SetRTCPMode(const RtcpMode rtcp_mode);
+ void SetProtectionMode(bool enable_nack,
+ bool enable_fec,
+ int payload_type_red,
+ int payload_type_fec);
+ bool IsSendingFecEnabled();
+ int SetSenderBufferingMode(int target_delay_ms);
+ int SetReceiverBufferingMode(int target_delay_ms);
+ int SetSendTimestampOffsetStatus(bool enable, int id);
+ int SetReceiveTimestampOffsetStatus(bool enable, int id);
+ int SetSendAbsoluteSendTimeStatus(bool enable, int id);
+ int SetReceiveAbsoluteSendTimeStatus(bool enable, int id);
+ int SetSendVideoRotationStatus(bool enable, int id);
+ int SetReceiveVideoRotationStatus(bool enable, int id);
+ int SetSendTransportSequenceNumber(bool enable, int id);
+ int SetReceiveTransportSequenceNumber(bool enable, int id);
+ void SetRtcpXrRrtrStatus(bool enable);
+ void EnableTMMBR(bool enable);
+
+ // Sets SSRC for outgoing stream.
+ int32_t SetSSRC(const uint32_t SSRC,
+ const StreamType usage,
+ const unsigned char simulcast_idx);
+
+ // Gets SSRC for outgoing stream number |idx|.
+ int32_t GetLocalSSRC(uint8_t idx, unsigned int* ssrc);
+
+ // Gets SSRC for the incoming stream.
+ uint32_t GetRemoteSSRC();
+
+ int SetRtxSendPayloadType(int payload_type, int associated_payload_type);
+ void SetRtxReceivePayloadType(int payload_type, int associated_payload_type);
+ // If set to true, the RTX payload type mapping supplied in
+ // |SetRtxReceivePayloadType| will be used when restoring RTX packets. Without
+ // it, RTX packets will always be restored to the last non-RTX packet payload
+ // type received.
+ void SetUseRtxPayloadMappingOnRestore(bool val);
+
+ void SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state);
+ RtpState GetRtpStateForSsrc(uint32_t ssrc);
+
+ // Sets the CName for the outgoing stream on the channel.
+ int32_t SetRTCPCName(const char* rtcp_cname);
+
+ // Gets the CName of the incoming stream.
+ int32_t GetRemoteRTCPCName(char rtcp_cname[]);
+
+ // Returns statistics reported by the remote client in an RTCP packet.
+ // TODO(pbos): Remove this along with VideoSendStream::GetRtt().
+ int32_t GetSendRtcpStatistics(uint16_t* fraction_lost,
+ uint32_t* cumulative_lost,
+ uint32_t* extended_max,
+ uint32_t* jitter_samples,
+ int64_t* rtt_ms);
+
+ // Called on receipt of RTCP report block from remote side.
+ void RegisterSendChannelRtcpStatisticsCallback(
+ RtcpStatisticsCallback* callback);
+
+ // Called on generation of RTCP stats
+ void RegisterReceiveChannelRtcpStatisticsCallback(
+ RtcpStatisticsCallback* callback);
+
+ // Gets send statistics for the rtp and rtx stream.
+ void GetSendStreamDataCounters(StreamDataCounters* rtp_counters,
+ StreamDataCounters* rtx_counters) const;
+
+ // Gets received stream data counters.
+ void GetReceiveStreamDataCounters(StreamDataCounters* rtp_counters,
+ StreamDataCounters* rtx_counters) const;
+
+ // Called on update of RTP statistics.
+ void RegisterSendChannelRtpStatisticsCallback(
+ StreamDataCountersCallback* callback);
+
+ // Called on update of RTP statistics.
+ void RegisterReceiveChannelRtpStatisticsCallback(
+ StreamDataCountersCallback* callback);
+
+ void GetSendRtcpPacketTypeCounter(
+ RtcpPacketTypeCounter* packet_counter) const;
+
+ void GetReceiveRtcpPacketTypeCounter(
+ RtcpPacketTypeCounter* packet_counter) const;
+
+ void RegisterSendSideDelayObserver(SendSideDelayObserver* observer);
+
+ // Called on any new send bitrate estimate.
+ void RegisterSendBitrateObserver(BitrateStatisticsObserver* observer);
+
+ // Implements RtpFeedback.
+ int32_t OnInitializeDecoder(const int8_t payload_type,
+ const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+ const int frequency,
+ const uint8_t channels,
+ const uint32_t rate) override;
+ void OnIncomingSSRCChanged(const uint32_t ssrc) override;
+ void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override;
+
+ int32_t SetRemoteSSRCType(const StreamType usage, const uint32_t SSRC);
+
+ int32_t StartSend();
+ int32_t StopSend();
+ bool Sending();
+ void StartReceive();
+ void StopReceive();
+
+ int32_t ReceivedRTPPacket(const void* rtp_packet,
+ const size_t rtp_packet_length,
+ const PacketTime& packet_time);
+ int32_t ReceivedRTCPPacket(const void* rtcp_packet,
+ const size_t rtcp_packet_length);
+
+ // Sets the maximum transfer unit size for the network link, i.e. including
+ // IP, UDP and RTP headers.
+ int32_t SetMTU(uint16_t mtu);
+
+ // Gets the modules used by the channel.
+ RtpRtcp* rtp_rtcp();
+ rtc::scoped_refptr<PayloadRouter> send_payload_router();
+ VCMProtectionCallback* vcm_protection_callback();
+
+
+ CallStatsObserver* GetStatsObserver();
+
+ // Implements VCMReceiveCallback.
+ virtual int32_t FrameToRender(VideoFrame& video_frame); // NOLINT
+
+ // Implements VCMReceiveCallback.
+ virtual int32_t ReceivedDecodedReferenceFrame(
+ const uint64_t picture_id);
+
+ // Implements VCMReceiveCallback.
+ void OnIncomingPayloadType(int payload_type) override;
+
+ // Implements VCMReceiveStatisticsCallback.
+ void OnReceiveRatesUpdated(uint32_t bit_rate, uint32_t frame_rate) override;
+ void OnDiscardedPacketsUpdated(int discarded_packets) override;
+ void OnFrameCountsUpdated(const FrameCounts& frame_counts) override;
+
+ // Implements VCMDecoderTimingCallback.
+ virtual void OnDecoderTiming(int decode_ms,
+ int max_decode_ms,
+ int current_delay_ms,
+ int target_delay_ms,
+ int jitter_buffer_ms,
+ int min_playout_delay_ms,
+ int render_delay_ms);
+
+ // Implements FrameTypeCallback.
+ virtual int32_t RequestKeyFrame();
+
+ // Implements FrameTypeCallback.
+ virtual int32_t SliceLossIndicationRequest(
+ const uint64_t picture_id);
+
+ // Implements VideoPacketRequestCallback.
+ int32_t ResendPackets(const uint16_t* sequence_numbers,
+ uint16_t length) override;
+
+ int32_t SetVoiceChannel(int32_t ve_channel_id,
+ VoEVideoSync* ve_sync_interface);
+ int32_t VoiceChannel();
+
+ // New-style callbacks, used by VideoReceiveStream.
+ void RegisterPreRenderCallback(I420FrameCallback* pre_render_callback);
+ void RegisterPreDecodeImageCallback(
+ EncodedImageCallback* pre_decode_callback);
+
+ void RegisterSendFrameCountObserver(FrameCountObserver* observer);
+ void RegisterRtcpPacketTypeCounterObserver(
+ RtcpPacketTypeCounterObserver* observer);
+ void RegisterReceiveStatisticsProxy(
+ ReceiveStatisticsProxy* receive_statistics_proxy);
+ void SetIncomingVideoStream(IncomingVideoStream* incoming_video_stream);
+
+ protected:
+ static bool ChannelDecodeThreadFunction(void* obj);
+ bool ChannelDecodeProcess();
+
+ void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms);
+
+ int ProtectionRequest(const FecProtectionParams* delta_fec_params,
+ const FecProtectionParams* key_fec_params,
+ uint32_t* sent_video_rate_bps,
+ uint32_t* sent_nack_rate_bps,
+ uint32_t* sent_fec_rate_bps);
+
+ private:
+ static std::vector<RtpRtcp*> CreateRtpRtcpModules(
+ bool receiver_only,
+ ReceiveStatistics* receive_statistics,
+ Transport* outgoing_transport,
+ RtcpIntraFrameObserver* intra_frame_callback,
+ RtcpBandwidthObserver* bandwidth_callback,
+ TransportFeedbackObserver* transport_feedback_callback,
+ RtcpRttStats* rtt_stats,
+ RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
+ RemoteBitrateEstimator* remote_bitrate_estimator,
+ RtpPacketSender* paced_sender,
+ TransportSequenceNumberAllocator* transport_sequence_number_allocator,
+ BitrateStatisticsObserver* send_bitrate_observer,
+ FrameCountObserver* send_frame_count_observer,
+ SendSideDelayObserver* send_side_delay_observer,
+ size_t num_modules);
+
+ // Assumed to be protected.
+ void StartDecodeThread();
+ void StopDecodeThread();
+
+ void ProcessNACKRequest(const bool enable);
+ // Compute NACK list parameters for the buffering mode.
+ int GetRequiredNackListSize(int target_delay_ms);
+ void SetRtxSendStatus(bool enable);
+
+ void UpdateHistograms();
+
+ // ViEChannel exposes methods that allow to modify observers and callbacks
+ // to be modified. Such an API-style is cumbersome to implement and maintain
+ // at all the levels when comparing to only setting them at construction. As
+ // so this class instantiates its children with a wrapper that can be modified
+ // at a later time.
+ template <class T>
+ class RegisterableCallback : public T {
+ public:
+ RegisterableCallback()
+ : critsect_(CriticalSectionWrapper::CreateCriticalSection()),
+ callback_(NULL) {}
+
+ void Set(T* callback) {
+ CriticalSectionScoped cs(critsect_.get());
+ callback_ = callback;
+ }
+
+ protected:
+ // Note: this should be implemented with a RW-lock to allow simultaneous
+ // calls into the callback. However that doesn't seem to be needed for the
+ // current type of callbacks covered by this class.
+ rtc::scoped_ptr<CriticalSectionWrapper> critsect_;
+ T* callback_ GUARDED_BY(critsect_);
+
+ private:
+ RTC_DISALLOW_COPY_AND_ASSIGN(RegisterableCallback);
+ };
+
+ class RegisterableBitrateStatisticsObserver:
+ public RegisterableCallback<BitrateStatisticsObserver> {
+ virtual void Notify(const BitrateStatistics& total_stats,
+ const BitrateStatistics& retransmit_stats,
+ uint32_t ssrc) {
+ CriticalSectionScoped cs(critsect_.get());
+ if (callback_)
+ callback_->Notify(total_stats, retransmit_stats, ssrc);
+ }
+ } send_bitrate_observer_;
+
+ class RegisterableFrameCountObserver
+ : public RegisterableCallback<FrameCountObserver> {
+ public:
+ virtual void FrameCountUpdated(const FrameCounts& frame_counts,
+ uint32_t ssrc) {
+ CriticalSectionScoped cs(critsect_.get());
+ if (callback_)
+ callback_->FrameCountUpdated(frame_counts, ssrc);
+ }
+
+ private:
+ } send_frame_count_observer_;
+
+ class RegisterableSendSideDelayObserver :
+ public RegisterableCallback<SendSideDelayObserver> {
+ void SendSideDelayUpdated(int avg_delay_ms,
+ int max_delay_ms,
+ uint32_t ssrc) override {
+ CriticalSectionScoped cs(critsect_.get());
+ if (callback_)
+ callback_->SendSideDelayUpdated(avg_delay_ms, max_delay_ms, ssrc);
+ }
+ } send_side_delay_observer_;
+
+ class RegisterableRtcpPacketTypeCounterObserver
+ : public RegisterableCallback<RtcpPacketTypeCounterObserver> {
+ public:
+ void RtcpPacketTypesCounterUpdated(
+ uint32_t ssrc,
+ const RtcpPacketTypeCounter& packet_counter) override {
+ CriticalSectionScoped cs(critsect_.get());
+ if (callback_)
+ callback_->RtcpPacketTypesCounterUpdated(ssrc, packet_counter);
+ counter_map_[ssrc] = packet_counter;
+ }
+
+ virtual std::map<uint32_t, RtcpPacketTypeCounter> GetPacketTypeCounterMap()
+ const {
+ CriticalSectionScoped cs(critsect_.get());
+ return counter_map_;
+ }
+
+ private:
+ std::map<uint32_t, RtcpPacketTypeCounter> counter_map_
+ GUARDED_BY(critsect_);
+ } rtcp_packet_type_counter_observer_;
+
+ const uint32_t number_of_cores_;
+ const bool sender_;
+
+ ProcessThread* const module_process_thread_;
+
+ // Used for all registered callbacks except rendering.
+ rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+
+ // Owned modules/classes.
+ rtc::scoped_refptr<PayloadRouter> send_payload_router_;
+ rtc::scoped_ptr<ViEChannelProtectionCallback> vcm_protection_callback_;
+
+ VideoCodingModule* const vcm_;
+ ViEReceiver vie_receiver_;
+ ViESyncModule vie_sync_;
+
+ // Helper to report call statistics.
+ rtc::scoped_ptr<ChannelStatsObserver> stats_observer_;
+
+ // Not owned.
+ ReceiveStatisticsProxy* receive_stats_callback_ GUARDED_BY(crit_);
+ FrameCounts receive_frame_counts_ GUARDED_BY(crit_);
+ IncomingVideoStream* incoming_video_stream_ GUARDED_BY(crit_);
+ RtcpIntraFrameObserver* const intra_frame_observer_;
+ RtcpRttStats* const rtt_stats_;
+ PacedSender* const paced_sender_;
+ PacketRouter* const packet_router_;
+
+ const rtc::scoped_ptr<RtcpBandwidthObserver> bandwidth_observer_;
+ TransportFeedbackObserver* const transport_feedback_observer_;
+
+ rtc::scoped_ptr<ThreadWrapper> decode_thread_;
+
+ int nack_history_size_sender_;
+ int max_nack_reordering_threshold_;
+ I420FrameCallback* pre_render_callback_ GUARDED_BY(crit_);
+
+ const rtc::scoped_ptr<ReportBlockStats> report_block_stats_sender_;
+
+ int64_t time_of_first_rtt_ms_ GUARDED_BY(crit_);
+ int64_t rtt_sum_ms_ GUARDED_BY(crit_);
+ int64_t last_rtt_ms_ GUARDED_BY(crit_);
+ size_t num_rtts_ GUARDED_BY(crit_);
+
+ // RtpRtcp modules, declared last as they use other members on construction.
+ const std::vector<RtpRtcp*> rtp_rtcp_modules_;
+ size_t num_active_rtp_rtcp_modules_ GUARDED_BY(crit_);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_