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}
diff --git a/call/rtp_stream_receiver_controller.h b/call/rtp_stream_receiver_controller.h
index 284c9fa..38b2f11 100644
--- a/call/rtp_stream_receiver_controller.h
+++ b/call/rtp_stream_receiver_controller.h
@@ -22,10 +22,8 @@
// 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 @@
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:
diff --git a/call/rtp_stream_receiver_controller_interface.h b/call/rtp_stream_receiver_controller_interface.h
index a5e5295..68a9e7d 100644
--- a/call/rtp_stream_receiver_controller_interface.h
+++ b/call/rtp_stream_receiver_controller_interface.h
@@ -18,12 +18,11 @@
// 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() {}
diff --git a/call/rtp_transport_controller_send.h b/call/rtp_transport_controller_send.h
index abe13ee..37a5c1e 100644
--- a/call/rtp_transport_controller_send.h
+++ b/call/rtp_transport_controller_send.h
@@ -41,9 +41,6 @@
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,
diff --git a/modules/audio_device/include/fake_audio_device.h b/modules/audio_device/include/fake_audio_device.h
index 9949627..2322ce0 100644
--- a/modules/audio_device/include/fake_audio_device.h
+++ b/modules/audio_device/include/fake_audio_device.h
@@ -19,8 +19,9 @@
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;
diff --git a/modules/rtp_rtcp/source/receive_statistics_impl.cc b/modules/rtp_rtcp/source/receive_statistics_impl.cc
index d2dba66..1e8e399 100644
--- a/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -176,7 +176,6 @@
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()) {
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index ae4bf51..c3f7151 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -308,9 +308,6 @@
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
diff --git a/modules/video_capture/video_capture_impl.cc b/modules/video_capture/video_capture_impl.cc
index fcee2be..234c2e1 100644
--- a/modules/video_capture/video_capture_impl.cc
+++ b/modules/video_capture/video_capture_impl.cc
@@ -152,8 +152,6 @@
// 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);
diff --git a/modules/video_coding/codecs/h264/h264_encoder_impl.cc b/modules/video_coding/codecs/h264/h264_encoder_impl.cc
index ce976ee..f4bda65 100644
--- a/modules/video_coding/codecs/h264/h264_encoder_impl.cc
+++ b/modules/video_coding/codecs/h264/h264_encoder_impl.cc
@@ -110,7 +110,6 @@
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);
diff --git a/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.cc b/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.cc
index 6bc306d..0f05d1a 100644
--- a/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.cc
+++ b/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.cc
@@ -116,7 +116,7 @@
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));
diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
index cef9ee9..d6e95e0 100644
--- a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
+++ b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
@@ -1125,8 +1125,6 @@
}
}
- // TODO(nisse): Introduce some buffer cache or buffer pool, to reduce
- // allocations and/or copy operations.
auto buffer = EncodedImageBuffer::Create(encoded_size);
iter = NULL;
diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
index 35675f5..609990e 100644
--- a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
+++ b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
@@ -1706,8 +1706,6 @@
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));
diff --git a/modules/video_coding/packet.cc b/modules/video_coding/packet.cc
index 324248a..f1bac4a 100644
--- a/modules/video_coding/packet.cc
+++ b/modules/video_coding/packet.cc
@@ -58,7 +58,6 @@
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};
diff --git a/modules/video_coding/session_info.h b/modules/video_coding/session_info.h
index 846352a..6079dbb 100644
--- a/modules/video_coding/session_info.h
+++ b/modules/video_coding/session_info.h
@@ -49,7 +49,6 @@
// 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;
diff --git a/modules/video_coding/utility/quality_scaler.cc b/modules/video_coding/utility/quality_scaler.cc
index 53b770b..6f66b4d 100644
--- a/modules/video_coding/utility/quality_scaler.cc
+++ b/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;
diff --git a/video/adaptation/overuse_frame_detector.cc b/video/adaptation/overuse_frame_detector.cc
index 38cf4a2..9836a46 100644
--- a/video/adaptation/overuse_frame_detector.cc
+++ b/video/adaptation/overuse_frame_detector.cc
@@ -526,7 +526,7 @@
: 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),