Update/delete old TODOs

Bug: webrtc:10198
Change-Id: I0341e068d792bc0b143db86e675988f4cd07ff2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267822
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37454}

call/rtp_stream_receiver_controller.h

@@ -22,10 +22,8 @@ class RtpPacketReceived;
 
 // This class represents the RTP receive parsing and demuxing, for a
 // single RTP session.
-// TODO(nisse): Add RTCP processing, we should aim to terminate RTCP
-// and not leave any RTCP processing to individual receive streams.
-// TODO(nisse): Extract per-packet processing, including parsing and
-// demuxing, into a separate class.
+// TODO(bugs.webrtc.org/7135): Add RTCP processing, we should aim to terminate
+// RTCP and not leave any RTCP processing to individual receive streams.
 class RtpStreamReceiverController
     : public RtpStreamReceiverControllerInterface {
  public:
@@ -41,7 +39,7 @@ class RtpStreamReceiverController
   bool AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) override;
   size_t RemoveSink(const RtpPacketSinkInterface* sink) override;
 
-  // TODO(nisse): Not yet responsible for parsing.
+  // TODO(bugs.webrtc.org/7135): Not yet responsible for parsing.
   bool OnRtpPacket(const RtpPacketReceived& packet);
 
  private:

call/rtp_stream_receiver_controller_interface.h

@@ -18,12 +18,11 @@ namespace webrtc {
 
 // An RtpStreamReceiver is responsible for the rtp-specific but
 // media-independent state needed for receiving an RTP stream.
-// TODO(nisse): Currently, only owns the association between ssrc and
-// the stream's RtpPacketSinkInterface. Ownership of corresponding
-// objects from modules/rtp_rtcp/ should move to this class (or
-// rather, the corresponding implementation class). We should add
-// methods for getting rtp receive stats, and for sending RTCP
-// messages related to the receive stream.
+// TODO(bugs.webrtc.org/7135): Currently, only owns the association between ssrc
+// and the stream's RtpPacketSinkInterface. Ownership of corresponding objects
+// from modules/rtp_rtcp/ should move to this class (or rather, the
+// corresponding implementation class). We should add methods for getting rtp
+// receive stats, and for sending RTCP messages related to the receive stream.
 class RtpStreamReceiverInterface {
  public:
   virtual ~RtpStreamReceiverInterface() {}

call/rtp_transport_controller_send.h

@@ -41,9 +41,6 @@ class Clock;
 class FrameEncryptorInterface;
 class RtcEventLog;
 
-// TODO(nisse): When we get the underlying transports here, we should
-// have one object implementing RtpTransportControllerSendInterface
-// per transport, sharing the same congestion controller.
 class RtpTransportControllerSend final
     : public RtpTransportControllerSendInterface,
       public RtcpBandwidthObserver,

modules/audio_device/include/fake_audio_device.h

@@ -19,8 +19,9 @@ namespace webrtc {
 class FakeAudioDeviceModule
     : public webrtc_impl::AudioDeviceModuleDefault<AudioDeviceModule> {
  public:
-  // TODO(nisse): Fix all users of this class to managed references using
-  // scoped_refptr. Current code doesn't always use refcounting for this class.
+  // TODO(bugs.webrtc.org/12701): Fix all users of this class to managed
+  // references using scoped_refptr. Current code doesn't always use refcounting
+  // for this class.
   void AddRef() const override {}
   rtc::RefCountReleaseStatus Release() const override {
     return rtc::RefCountReleaseStatus::kDroppedLastRef;

modules/rtp_rtcp/source/receive_statistics_impl.cc

@@ -176,7 +176,6 @@ void StreamStatisticianImpl::EnableRetransmitDetection(bool enable) {
 RtpReceiveStats StreamStatisticianImpl::GetStats() const {
   RtpReceiveStats stats;
   stats.packets_lost = cumulative_loss_;
-  // TODO(nisse): Can we return a float instead?
   // Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
   stats.jitter = jitter_q4_ >> 4;
   if (receive_counters_.last_packet_received_timestamp_ms.has_value()) {

modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -308,9 +308,6 @@ RTCPSender::FeedbackState ModuleRtpRtcpImpl::GetFeedbackState() {
   return state;
 }
 
-// TODO(nisse): This method shouldn't be called for a receive-only
-// stream. Delete rtp_sender_ check as soon as all applications are
-// updated.
 int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) {
   if (rtcp_sender_.Sending() != sending) {
     // Sends RTCP BYE when going from true to false

modules/video_capture/video_capture_impl.cc

@@ -152,8 +152,6 @@ int32_t VideoCaptureImpl::IncomingFrame(uint8_t* videoFrame,
   // Setting absolute height (in case it was negative).
   // In Windows, the image starts bottom left, instead of top left.
   // Setting a negative source height, inverts the image (within LibYuv).
-
-  // TODO(nisse): Use a pool?
   rtc::scoped_refptr<I420Buffer> buffer = I420Buffer::Create(
       target_width, target_height, stride_y, stride_uv, stride_uv);
 

modules/video_coding/codecs/h264/h264_encoder_impl.cc

@@ -110,7 +110,6 @@ static void RtpFragmentize(EncodedImage* encoded_image, SFrameBSInfo* info) {
       required_capacity += layerInfo.pNalLengthInByte[nal];
     }
   }
-  // TODO(nisse): Use a cache or buffer pool to avoid allocation?
   auto buffer = EncodedImageBuffer::Create(required_capacity);
   encoded_image->SetEncodedData(buffer);
 

modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.cc

@@ -116,7 +116,7 @@ MultiplexImageComponentHeader UnpackFrameHeader(const uint8_t* buffer) {
       ByteReader<uint32_t>::ReadBigEndian(buffer + offset);
   offset += sizeof(uint32_t);
 
-  // TODO(nisse): This makes the wire format depend on the numeric values of the
+  // This makes the wire format depend on the numeric values of the
   // VideoCodecType and VideoFrameType enum constants.
   frame_header.codec_type = static_cast<VideoCodecType>(
       ByteReader<uint8_t>::ReadBigEndian(buffer + offset));

modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc

@@ -1125,8 +1125,6 @@ int LibvpxVp8Encoder::GetEncodedPartitions(const VideoFrame& input_image,
       }
     }
 
-    // TODO(nisse): Introduce some buffer cache or buffer pool, to reduce
-    // allocations and/or copy operations.
     auto buffer = EncodedImageBuffer::Create(encoded_size);
 
     iter = NULL;

modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc

@@ -1706,8 +1706,6 @@ void LibvpxVp9Encoder::GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt) {
     DeliverBufferedFrame(end_of_picture);
   }
 
-  // TODO(nisse): Introduce some buffer cache or buffer pool, to reduce
-  // allocations and/or copy operations.
   encoded_image_.SetEncodedData(EncodedImageBuffer::Create(
       static_cast<const uint8_t*>(pkt->data.frame.buf), pkt->data.frame.sz));
 

modules/video_coding/packet.cc

@@ -58,7 +58,6 @@ VCMPacket::VCMPacket(const uint8_t* ptr,
     completeNALU = kNaluIncomplete;
   }
 
-  // TODO(nisse): Delete?
   // Playout decisions are made entirely based on first packet in a frame.
   if (!is_first_packet_in_frame()) {
     video_header.playout_delay = {-1, -1};

modules/video_coding/session_info.h

@@ -49,7 +49,6 @@ class VCMSessionInfo {
   // Returns the number of bytes deleted from the session.
   size_t MakeDecodable();
 
-  // TODO(nisse): Used by tests only.
   size_t SessionLength() const;
   int NumPackets() const;
   bool HaveFirstPacket() const;

modules/video_coding/utility/quality_scaler.cc

@@ -31,7 +31,6 @@
 namespace webrtc {
 
 namespace {
-// TODO(nisse): Delete, delegate to encoders.
 // Threshold constant used until first downscale (to permit fast rampup).
 static const int kMeasureMs = 2000;
 static const float kSamplePeriodScaleFactor = 2.5;

video/adaptation/overuse_frame_detector.cc

@@ -526,7 +526,7 @@ OveruseFrameDetector::OveruseFrameDetector(
     : options_(field_trials),
       metrics_observer_(metrics_observer),
       num_process_times_(0),
-      // TODO(nisse): Use absl::optional
+      // TODO(bugs.webrtc.org/9078): Use absl::optional
       last_capture_time_us_(-1),
       num_pixels_(0),
       max_framerate_(kDefaultFrameRate),