Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 344 insertions, 0 deletions

diff --git a/third_party/libwebrtc/video/video_send_stream.cc b/third_party/libwebrtc/video/video_send_stream.cc
new file mode 100644
index 0000000000..b99b08eefb
--- /dev/null
+++ b/third_party/libwebrtc/video/video_send_stream.cc
@@ -0,0 +1,344 @@
+/*
+ *  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 "video/video_send_stream.h"
+
+#include <utility>
+
+#include "api/array_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/video/video_stream_encoder_settings.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_header_extension_size.h"
+#include "modules/rtp_rtcp/source/rtp_sender.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/clock.h"
+#include "video/adaptation/overuse_frame_detector.h"
+#include "video/frame_cadence_adapter.h"
+#include "video/video_stream_encoder.h"
+
+namespace webrtc {
+
+namespace {
+
+size_t CalculateMaxHeaderSize(const RtpConfig& config) {
+  size_t header_size = kRtpHeaderSize;
+  size_t extensions_size = 0;
+  size_t fec_extensions_size = 0;
+  if (!config.extensions.empty()) {
+    RtpHeaderExtensionMap extensions_map(config.extensions);
+    extensions_size = RtpHeaderExtensionSize(RTPSender::VideoExtensionSizes(),
+                                             extensions_map);
+    fec_extensions_size =
+        RtpHeaderExtensionSize(RTPSender::FecExtensionSizes(), extensions_map);
+  }
+  header_size += extensions_size;
+  if (config.flexfec.payload_type >= 0) {
+    // All FEC extensions again plus maximum FlexFec overhead.
+    header_size += fec_extensions_size + 32;
+  } else {
+    if (config.ulpfec.ulpfec_payload_type >= 0) {
+      // Header with all the FEC extensions will be repeated plus maximum
+      // UlpFec overhead.
+      header_size += fec_extensions_size + 18;
+    }
+    if (config.ulpfec.red_payload_type >= 0) {
+      header_size += 1;  // RED header.
+    }
+  }
+  // Additional room for Rtx.
+  if (config.rtx.payload_type >= 0)
+    header_size += kRtxHeaderSize;
+  return header_size;
+}
+
+VideoStreamEncoder::BitrateAllocationCallbackType
+GetBitrateAllocationCallbackType(const VideoSendStream::Config& config,
+                                 const FieldTrialsView& field_trials) {
+  if (webrtc::RtpExtension::FindHeaderExtensionByUri(
+          config.rtp.extensions,
+          webrtc::RtpExtension::kVideoLayersAllocationUri,
+          config.crypto_options.srtp.enable_encrypted_rtp_header_extensions
+              ? RtpExtension::Filter::kPreferEncryptedExtension
+              : RtpExtension::Filter::kDiscardEncryptedExtension)) {
+    return VideoStreamEncoder::BitrateAllocationCallbackType::
+        kVideoLayersAllocation;
+  }
+  if (field_trials.IsEnabled("WebRTC-Target-Bitrate-Rtcp")) {
+    return VideoStreamEncoder::BitrateAllocationCallbackType::
+        kVideoBitrateAllocation;
+  }
+  return VideoStreamEncoder::BitrateAllocationCallbackType::
+      kVideoBitrateAllocationWhenScreenSharing;
+}
+
+RtpSenderFrameEncryptionConfig CreateFrameEncryptionConfig(
+    const VideoSendStream::Config* config) {
+  RtpSenderFrameEncryptionConfig frame_encryption_config;
+  frame_encryption_config.frame_encryptor = config->frame_encryptor.get();
+  frame_encryption_config.crypto_options = config->crypto_options;
+  return frame_encryption_config;
+}
+
+RtpSenderObservers CreateObservers(RtcpRttStats* call_stats,
+                                   EncoderRtcpFeedback* encoder_feedback,
+                                   SendStatisticsProxy* stats_proxy,
+                                   SendPacketObserver* send_packet_observer) {
+  RtpSenderObservers observers;
+  observers.rtcp_rtt_stats = call_stats;
+  observers.intra_frame_callback = encoder_feedback;
+  observers.rtcp_loss_notification_observer = encoder_feedback;
+  observers.report_block_data_observer = stats_proxy;
+  observers.rtp_stats = stats_proxy;
+  observers.bitrate_observer = stats_proxy;
+  observers.frame_count_observer = stats_proxy;
+  observers.rtcp_type_observer = stats_proxy;
+  observers.send_packet_observer = send_packet_observer;
+  return observers;
+}
+
+std::unique_ptr<VideoStreamEncoder> CreateVideoStreamEncoder(
+    Clock* clock,
+    int num_cpu_cores,
+    TaskQueueFactory* task_queue_factory,
+    SendStatisticsProxy* stats_proxy,
+    const VideoStreamEncoderSettings& encoder_settings,
+    VideoStreamEncoder::BitrateAllocationCallbackType
+        bitrate_allocation_callback_type,
+    const FieldTrialsView& field_trials,
+    webrtc::VideoEncoderFactory::EncoderSelectorInterface* encoder_selector) {
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> encoder_queue =
+      task_queue_factory->CreateTaskQueue("EncoderQueue",
+                                          TaskQueueFactory::Priority::NORMAL);
+  TaskQueueBase* encoder_queue_ptr = encoder_queue.get();
+  return std::make_unique<VideoStreamEncoder>(
+      clock, num_cpu_cores, stats_proxy, encoder_settings,
+      std::make_unique<OveruseFrameDetector>(stats_proxy),
+      FrameCadenceAdapterInterface::Create(clock, encoder_queue_ptr,
+                                           field_trials),
+      std::move(encoder_queue), bitrate_allocation_callback_type, field_trials,
+      encoder_selector);
+}
+
+}  // namespace
+
+namespace internal {
+
+VideoSendStream::VideoSendStream(
+    Clock* clock,
+    int num_cpu_cores,
+    TaskQueueFactory* task_queue_factory,
+    TaskQueueBase* network_queue,
+    RtcpRttStats* call_stats,
+    RtpTransportControllerSendInterface* transport,
+    BitrateAllocatorInterface* bitrate_allocator,
+    SendDelayStats* send_delay_stats,
+    RtcEventLog* event_log,
+    VideoSendStream::Config config,
+    VideoEncoderConfig encoder_config,
+    const std::map<uint32_t, RtpState>& suspended_ssrcs,
+    const std::map<uint32_t, RtpPayloadState>& suspended_payload_states,
+    std::unique_ptr<FecController> fec_controller,
+    const FieldTrialsView& field_trials)
+    : transport_(transport),
+      stats_proxy_(clock, config, encoder_config.content_type, field_trials),
+      send_packet_observer_(&stats_proxy_, send_delay_stats),
+      config_(std::move(config)),
+      content_type_(encoder_config.content_type),
+      video_stream_encoder_(CreateVideoStreamEncoder(
+          clock,
+          num_cpu_cores,
+          task_queue_factory,
+          &stats_proxy_,
+          config_.encoder_settings,
+          GetBitrateAllocationCallbackType(config_, field_trials),
+          field_trials,
+          config_.encoder_selector)),
+      encoder_feedback_(
+          clock,
+          config_.rtp.ssrcs,
+          video_stream_encoder_.get(),
+          [this](uint32_t ssrc, const std::vector<uint16_t>& seq_nums) {
+            return rtp_video_sender_->GetSentRtpPacketInfos(ssrc, seq_nums);
+          }),
+      rtp_video_sender_(transport->CreateRtpVideoSender(
+          suspended_ssrcs,
+          suspended_payload_states,
+          config_.rtp,
+          config_.rtcp_report_interval_ms,
+          config_.send_transport,
+          CreateObservers(call_stats,
+                          &encoder_feedback_,
+                          &stats_proxy_,
+                          &send_packet_observer_),
+          event_log,
+          std::move(fec_controller),
+          CreateFrameEncryptionConfig(&config_),
+          config_.frame_transformer)),
+      send_stream_(clock,
+                   &stats_proxy_,
+                   transport,
+                   bitrate_allocator,
+                   video_stream_encoder_.get(),
+                   &config_,
+                   encoder_config.max_bitrate_bps,
+                   encoder_config.bitrate_priority,
+                   encoder_config.content_type,
+                   rtp_video_sender_,
+                   field_trials) {
+  RTC_DCHECK(config_.encoder_settings.encoder_factory);
+  RTC_DCHECK(config_.encoder_settings.bitrate_allocator_factory);
+
+  video_stream_encoder_->SetFecControllerOverride(rtp_video_sender_);
+
+  ReconfigureVideoEncoder(std::move(encoder_config));
+}
+
+VideoSendStream::~VideoSendStream() {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK(!running_);
+  transport_->DestroyRtpVideoSender(rtp_video_sender_);
+}
+
+void VideoSendStream::Start() {
+  const std::vector<bool> active_layers(config_.rtp.ssrcs.size(), true);
+  StartPerRtpStream(active_layers);
+}
+
+void VideoSendStream::StartPerRtpStream(const std::vector<bool> active_layers) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  // Keep our `running_` flag expected state in sync with active layers since
+  // the `send_stream_` will be implicitly stopped/started depending on the
+  // state of the layers.
+  bool running = false;
+
+  rtc::StringBuilder active_layers_string;
+  active_layers_string << "{";
+  for (size_t i = 0; i < active_layers.size(); ++i) {
+    if (active_layers[i]) {
+      running = true;
+      active_layers_string << "1";
+    } else {
+      active_layers_string << "0";
+    }
+    if (i < active_layers.size() - 1) {
+      active_layers_string << ", ";
+    }
+  }
+  active_layers_string << "}";
+  RTC_LOG(LS_INFO) << "StartPerRtpStream: " << active_layers_string.str();
+  send_stream_.StartPerRtpStream(active_layers);
+  running_ = running;
+}
+
+void VideoSendStream::Stop() {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  if (!running_)
+    return;
+  RTC_DLOG(LS_INFO) << "VideoSendStream::Stop";
+  running_ = false;
+  send_stream_.Stop();
+}
+
+bool VideoSendStream::started() {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return running_;
+}
+
+void VideoSendStream::AddAdaptationResource(
+    rtc::scoped_refptr<Resource> resource) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  video_stream_encoder_->AddAdaptationResource(resource);
+}
+
+std::vector<rtc::scoped_refptr<Resource>>
+VideoSendStream::GetAdaptationResources() {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return video_stream_encoder_->GetAdaptationResources();
+}
+
+void VideoSendStream::SetSource(
+    rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
+    const DegradationPreference& degradation_preference) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  video_stream_encoder_->SetSource(source, degradation_preference);
+}
+
+void VideoSendStream::ReconfigureVideoEncoder(VideoEncoderConfig config) {
+  ReconfigureVideoEncoder(std::move(config), nullptr);
+}
+
+void VideoSendStream::ReconfigureVideoEncoder(VideoEncoderConfig config,
+                                              SetParametersCallback callback) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK_EQ(content_type_, config.content_type);
+  RTC_LOG(LS_VERBOSE) << "Encoder config: " << config.ToString()
+                      << " VideoSendStream config: " << config_.ToString();
+  video_stream_encoder_->ConfigureEncoder(
+      std::move(config),
+      config_.rtp.max_packet_size - CalculateMaxHeaderSize(config_.rtp),
+      std::move(callback));
+}
+
+VideoSendStream::Stats VideoSendStream::GetStats() {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return stats_proxy_.GetStats();
+}
+
+absl::optional<float> VideoSendStream::GetPacingFactorOverride() const {
+  return send_stream_.configured_pacing_factor();
+}
+
+void VideoSendStream::StopPermanentlyAndGetRtpStates(
+    VideoSendStream::RtpStateMap* rtp_state_map,
+    VideoSendStream::RtpPayloadStateMap* payload_state_map) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  video_stream_encoder_->Stop();
+
+  running_ = false;
+  // Always run these cleanup steps regardless of whether running_ was set
+  // or not. This will unregister callbacks before destruction.
+  // See `VideoSendStreamImpl::StopVideoSendStream` for more.
+  send_stream_.Stop();
+  *rtp_state_map = send_stream_.GetRtpStates();
+  *payload_state_map = send_stream_.GetRtpPayloadStates();
+}
+
+void VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  send_stream_.DeliverRtcp(packet, length);
+}
+
+void VideoSendStream::GenerateKeyFrame(const std::vector<std::string>& rids) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  // Map rids to layers. If rids is empty, generate a keyframe for all layers.
+  std::vector<VideoFrameType> next_frames(config_.rtp.ssrcs.size(),
+                                          VideoFrameType::kVideoFrameKey);
+  if (!config_.rtp.rids.empty() && !rids.empty()) {
+    std::fill(next_frames.begin(), next_frames.end(),
+              VideoFrameType::kVideoFrameDelta);
+    for (const auto& rid : rids) {
+      for (size_t i = 0; i < config_.rtp.rids.size(); i++) {
+        if (config_.rtp.rids[i] == rid) {
+          next_frames[i] = VideoFrameType::kVideoFrameKey;
+          break;
+        }
+      }
+    }
+  }
+  if (video_stream_encoder_) {
+    video_stream_encoder_->SendKeyFrame(next_frames);
+  }
+}
+
+}  // namespace internal
+}  // namespace webrtc