51238e6c28
After a send stream is stopped, it can still be re-used and implicitly restarted by activating layers. This change removes marking the flag we use for async operations as 'not alive' inside Stop() and only doing so when the send stream is stopped permanently. The effect this has is that an implicit start via UpdateActiveSimulcastLayers() will run and correctly update the states. Before, if a stream had been stopped, the safety flag would prevent the async operation from running. Bug: chromium:1241213 Change-Id: Iebdfabba3e1955aafa364760eebd4f66281bcc60 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229304 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34809}
325 lines
12 KiB
C++
325 lines
12 KiB
C++
/*
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* Copyright (c) 2013 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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#include "video/video_send_stream.h"
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#include <utility>
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#include "api/array_view.h"
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#include "api/video/video_stream_encoder_settings.h"
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "modules/rtp_rtcp/source/rtp_header_extension_size.h"
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#include "modules/rtp_rtcp/source/rtp_sender.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/strings/string_builder.h"
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#include "rtc_base/task_utils/to_queued_task.h"
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#include "system_wrappers/include/clock.h"
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#include "system_wrappers/include/field_trial.h"
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#include "video/adaptation/overuse_frame_detector.h"
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#include "video/video_stream_encoder.h"
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namespace webrtc {
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namespace {
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size_t CalculateMaxHeaderSize(const RtpConfig& config) {
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size_t header_size = kRtpHeaderSize;
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size_t extensions_size = 0;
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size_t fec_extensions_size = 0;
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if (!config.extensions.empty()) {
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RtpHeaderExtensionMap extensions_map(config.extensions);
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extensions_size = RtpHeaderExtensionSize(RTPSender::VideoExtensionSizes(),
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extensions_map);
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fec_extensions_size =
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RtpHeaderExtensionSize(RTPSender::FecExtensionSizes(), extensions_map);
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}
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header_size += extensions_size;
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if (config.flexfec.payload_type >= 0) {
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// All FEC extensions again plus maximum FlexFec overhead.
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header_size += fec_extensions_size + 32;
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} else {
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if (config.ulpfec.ulpfec_payload_type >= 0) {
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// Header with all the FEC extensions will be repeated plus maximum
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// UlpFec overhead.
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header_size += fec_extensions_size + 18;
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}
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if (config.ulpfec.red_payload_type >= 0) {
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header_size += 1; // RED header.
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}
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}
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// Additional room for Rtx.
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if (config.rtx.payload_type >= 0)
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header_size += kRtxHeaderSize;
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return header_size;
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}
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VideoStreamEncoder::BitrateAllocationCallbackType
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GetBitrateAllocationCallbackType(const VideoSendStream::Config& config) {
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if (webrtc::RtpExtension::FindHeaderExtensionByUri(
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config.rtp.extensions,
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webrtc::RtpExtension::kVideoLayersAllocationUri,
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config.crypto_options.srtp.enable_encrypted_rtp_header_extensions
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? RtpExtension::Filter::kPreferEncryptedExtension
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: RtpExtension::Filter::kDiscardEncryptedExtension)) {
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return VideoStreamEncoder::BitrateAllocationCallbackType::
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kVideoLayersAllocation;
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}
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if (field_trial::IsEnabled("WebRTC-Target-Bitrate-Rtcp")) {
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return VideoStreamEncoder::BitrateAllocationCallbackType::
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kVideoBitrateAllocation;
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}
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return VideoStreamEncoder::BitrateAllocationCallbackType::
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kVideoBitrateAllocationWhenScreenSharing;
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}
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RtpSenderFrameEncryptionConfig CreateFrameEncryptionConfig(
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const VideoSendStream::Config* config) {
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RtpSenderFrameEncryptionConfig frame_encryption_config;
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frame_encryption_config.frame_encryptor = config->frame_encryptor;
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frame_encryption_config.crypto_options = config->crypto_options;
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return frame_encryption_config;
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}
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RtpSenderObservers CreateObservers(RtcpRttStats* call_stats,
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EncoderRtcpFeedback* encoder_feedback,
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SendStatisticsProxy* stats_proxy,
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SendDelayStats* send_delay_stats) {
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RtpSenderObservers observers;
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observers.rtcp_rtt_stats = call_stats;
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observers.intra_frame_callback = encoder_feedback;
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observers.rtcp_loss_notification_observer = encoder_feedback;
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observers.report_block_data_observer = stats_proxy;
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observers.rtp_stats = stats_proxy;
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observers.bitrate_observer = stats_proxy;
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observers.frame_count_observer = stats_proxy;
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observers.rtcp_type_observer = stats_proxy;
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observers.send_delay_observer = stats_proxy;
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observers.send_packet_observer = send_delay_stats;
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return observers;
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}
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} // namespace
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namespace internal {
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VideoSendStream::VideoSendStream(
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Clock* clock,
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int num_cpu_cores,
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TaskQueueFactory* task_queue_factory,
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RtcpRttStats* call_stats,
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RtpTransportControllerSendInterface* transport,
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BitrateAllocatorInterface* bitrate_allocator,
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SendDelayStats* send_delay_stats,
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RtcEventLog* event_log,
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VideoSendStream::Config config,
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VideoEncoderConfig encoder_config,
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const std::map<uint32_t, RtpState>& suspended_ssrcs,
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const std::map<uint32_t, RtpPayloadState>& suspended_payload_states,
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std::unique_ptr<FecController> fec_controller)
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: rtp_transport_queue_(transport->GetWorkerQueue()),
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transport_(transport),
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stats_proxy_(clock, config, encoder_config.content_type),
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config_(std::move(config)),
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content_type_(encoder_config.content_type),
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video_stream_encoder_(std::make_unique<VideoStreamEncoder>(
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clock,
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num_cpu_cores,
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&stats_proxy_,
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config_.encoder_settings,
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std::make_unique<OveruseFrameDetector>(&stats_proxy_),
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task_queue_factory,
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GetBitrateAllocationCallbackType(config_))),
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encoder_feedback_(
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clock,
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config_.rtp.ssrcs,
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video_stream_encoder_.get(),
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[this](uint32_t ssrc, const std::vector<uint16_t>& seq_nums) {
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return rtp_video_sender_->GetSentRtpPacketInfos(ssrc, seq_nums);
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}),
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rtp_video_sender_(
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transport->CreateRtpVideoSender(suspended_ssrcs,
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suspended_payload_states,
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config_.rtp,
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config_.rtcp_report_interval_ms,
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config_.send_transport,
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CreateObservers(call_stats,
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&encoder_feedback_,
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&stats_proxy_,
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send_delay_stats),
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event_log,
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std::move(fec_controller),
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CreateFrameEncryptionConfig(&config_),
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config_.frame_transformer)),
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send_stream_(clock,
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&stats_proxy_,
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rtp_transport_queue_,
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transport,
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bitrate_allocator,
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video_stream_encoder_.get(),
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&config_,
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encoder_config.max_bitrate_bps,
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encoder_config.bitrate_priority,
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encoder_config.content_type,
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rtp_video_sender_) {
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RTC_DCHECK(config_.encoder_settings.encoder_factory);
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RTC_DCHECK(config_.encoder_settings.bitrate_allocator_factory);
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video_stream_encoder_->SetFecControllerOverride(rtp_video_sender_);
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ReconfigureVideoEncoder(std::move(encoder_config));
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}
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VideoSendStream::~VideoSendStream() {
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RTC_DCHECK_RUN_ON(&thread_checker_);
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RTC_DCHECK(!running_);
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transport_->DestroyRtpVideoSender(rtp_video_sender_);
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}
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void VideoSendStream::UpdateActiveSimulcastLayers(
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const std::vector<bool> active_layers) {
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RTC_DCHECK_RUN_ON(&thread_checker_);
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// Keep our `running_` flag expected state in sync with active layers since
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// the `send_stream_` will be implicitly stopped/started depending on the
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// state of the layers.
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bool running = false;
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rtc::StringBuilder active_layers_string;
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active_layers_string << "{";
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for (size_t i = 0; i < active_layers.size(); ++i) {
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if (active_layers[i]) {
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running = true;
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active_layers_string << "1";
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} else {
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active_layers_string << "0";
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}
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if (i < active_layers.size() - 1) {
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active_layers_string << ", ";
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}
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}
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active_layers_string << "}";
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RTC_LOG(LS_INFO) << "UpdateActiveSimulcastLayers: "
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<< active_layers_string.str();
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rtp_transport_queue_->PostTask(
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ToQueuedTask(transport_queue_safety_, [this, active_layers] {
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send_stream_.UpdateActiveSimulcastLayers(active_layers);
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}));
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running_ = running;
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}
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void VideoSendStream::Start() {
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RTC_DCHECK_RUN_ON(&thread_checker_);
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RTC_DLOG(LS_INFO) << "VideoSendStream::Start";
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if (running_)
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return;
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running_ = true;
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rtp_transport_queue_->PostTask(ToQueuedTask([this] {
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transport_queue_safety_->SetAlive();
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send_stream_.Start();
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thread_sync_event_.Set();
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}));
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// It is expected that after VideoSendStream::Start has been called, incoming
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// frames are not dropped in VideoStreamEncoder. To ensure this, Start has to
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// be synchronized.
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// TODO(tommi): ^^^ Validate if this still holds.
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thread_sync_event_.Wait(rtc::Event::kForever);
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}
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void VideoSendStream::Stop() {
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RTC_DCHECK_RUN_ON(&thread_checker_);
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if (!running_)
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return;
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RTC_DLOG(LS_INFO) << "VideoSendStream::Stop";
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running_ = false;
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rtp_transport_queue_->PostTask(ToQueuedTask(transport_queue_safety_, [this] {
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// As the stream can get re-used and implicitly restarted via changing
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// the state of the active layers, we do not mark the
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// `transport_queue_safety_` flag with `SetNotAlive()` here. That's only
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// done when we stop permanently via `StopPermanentlyAndGetRtpStates()`.
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send_stream_.Stop();
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}));
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}
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bool VideoSendStream::started() {
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RTC_DCHECK_RUN_ON(&thread_checker_);
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return running_;
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}
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void VideoSendStream::AddAdaptationResource(
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rtc::scoped_refptr<Resource> resource) {
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RTC_DCHECK_RUN_ON(&thread_checker_);
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video_stream_encoder_->AddAdaptationResource(resource);
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}
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std::vector<rtc::scoped_refptr<Resource>>
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VideoSendStream::GetAdaptationResources() {
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RTC_DCHECK_RUN_ON(&thread_checker_);
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return video_stream_encoder_->GetAdaptationResources();
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}
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void VideoSendStream::SetSource(
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rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
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const DegradationPreference& degradation_preference) {
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RTC_DCHECK_RUN_ON(&thread_checker_);
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video_stream_encoder_->SetSource(source, degradation_preference);
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}
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void VideoSendStream::ReconfigureVideoEncoder(VideoEncoderConfig config) {
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RTC_DCHECK_RUN_ON(&thread_checker_);
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RTC_DCHECK_EQ(content_type_, config.content_type);
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video_stream_encoder_->ConfigureEncoder(
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std::move(config),
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config_.rtp.max_packet_size - CalculateMaxHeaderSize(config_.rtp));
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}
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VideoSendStream::Stats VideoSendStream::GetStats() {
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// TODO(perkj, solenberg): Some test cases in EndToEndTest call GetStats from
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// a network thread. See comment in Call::GetStats().
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// RTC_DCHECK_RUN_ON(&thread_checker_);
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return stats_proxy_.GetStats();
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}
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absl::optional<float> VideoSendStream::GetPacingFactorOverride() const {
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return send_stream_.configured_pacing_factor();
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}
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void VideoSendStream::StopPermanentlyAndGetRtpStates(
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VideoSendStream::RtpStateMap* rtp_state_map,
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VideoSendStream::RtpPayloadStateMap* payload_state_map) {
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RTC_DCHECK_RUN_ON(&thread_checker_);
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video_stream_encoder_->Stop();
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running_ = false;
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// Always run these cleanup steps regardless of whether running_ was set
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// or not. This will unregister callbacks before destruction.
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// See `VideoSendStreamImpl::StopVideoSendStream` for more.
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rtp_transport_queue_->PostTask([this, rtp_state_map, payload_state_map]() {
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transport_queue_safety_->SetNotAlive();
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send_stream_.Stop();
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*rtp_state_map = send_stream_.GetRtpStates();
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*payload_state_map = send_stream_.GetRtpPayloadStates();
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thread_sync_event_.Set();
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});
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thread_sync_event_.Wait(rtc::Event::kForever);
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}
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void VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
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// Called on a network thread.
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send_stream_.DeliverRtcp(packet, length);
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}
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} // namespace internal
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} // namespace webrtc
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