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authorNiels Möller <nisse@webrtc.org>2022-07-05 08:55:19 +0200
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>2022-07-05 09:59:33 +0000
commit6939f63ca149b78b2229eea7e3a69f143b3e261b (patch)
tree4a0934ba4319e8882a85a0e3762e86b4fbfddfb0
parentc8152fe4a8d62323d01924679dca8acc602ccf48 (diff)
downloadwebrtc-6939f63ca149b78b2229eea7e3a69f143b3e261b.tar.gz
Update old TODO comments
Bug: None Change-Id: I96850df6cfa19303043108a59ef60d7b686ec747 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267661 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37436}
-rw-r--r--api/transport/BUILD.gn1
-rw-r--r--api/video/encoded_image.h5
-rw-r--r--api/video/i420_buffer.h4
-rw-r--r--api/video/i422_buffer.h4
-rw-r--r--api/video/i444_buffer.h4
-rw-r--r--api/video/nv12_buffer.h4
-rw-r--r--api/video/video_frame.h12
-rw-r--r--api/video/video_stream_encoder_interface.h4
-rw-r--r--api/video/video_stream_encoder_observer.h11
-rw-r--r--api/video_codecs/video_encoder.h12
-rw-r--r--api/video_codecs/video_encoder_config.h2
-rw-r--r--audio/channel_receive.cc3
-rw-r--r--audio/channel_send.cc2
-rw-r--r--call/call.cc10
-rw-r--r--call/rtp_config.h2
-rw-r--r--modules/video_coding/codecs/h264/h264_decoder_impl.cc4
16 files changed, 36 insertions, 48 deletions
diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn
index 44d4b117b5..596710ee45 100644
--- a/api/transport/BUILD.gn
+++ b/api/transport/BUILD.gn
@@ -58,7 +58,6 @@ rtc_library("field_trial_based_config") {
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
-# TODO(nisse): Rename?
rtc_source_set("datagram_transport_interface") {
visibility = [ "*" ]
sources = [ "data_channel_transport_interface.h" ]
diff --git a/api/video/encoded_image.h b/api/video/encoded_image.h
index 30acd8dc7a..dae790c46c 100644
--- a/api/video/encoded_image.h
+++ b/api/video/encoded_image.h
@@ -78,9 +78,8 @@ class RTC_EXPORT EncodedImage {
EncodedImage& operator=(EncodedImage&&);
EncodedImage& operator=(const EncodedImage&);
- // TODO(nisse): Change style to timestamp(), set_timestamp(), for consistency
- // with the VideoFrame class.
- // Set frame timestamp (90kHz).
+ // TODO(bugs.webrtc.org/9378): Change style to timestamp(), set_timestamp(),
+ // for consistency with the VideoFrame class. Set frame timestamp (90kHz).
void SetTimestamp(uint32_t timestamp) { timestamp_rtp_ = timestamp; }
// Get frame timestamp (90kHz).
diff --git a/api/video/i420_buffer.h b/api/video/i420_buffer.h
index af52c64fb4..b337489657 100644
--- a/api/video/i420_buffer.h
+++ b/api/video/i420_buffer.h
@@ -65,8 +65,8 @@ class RTC_EXPORT I420Buffer : public I420BufferInterface {
// quirks in memory checkers
// (https://bugs.chromium.org/p/libyuv/issues/detail?id=377) and
// ffmpeg (http://crbug.com/390941).
- // TODO(nisse): Deprecated. Should be deleted if/when those issues
- // are resolved in a better way. Or in the mean time, use SetBlack.
+ // TODO(https://crbug.com/390941): Deprecated. Should be deleted if/when those
+ // issues are resolved in a better way. Or in the mean time, use SetBlack.
void InitializeData();
int width() const override;
diff --git a/api/video/i422_buffer.h b/api/video/i422_buffer.h
index 16c717469b..600b4ecea7 100644
--- a/api/video/i422_buffer.h
+++ b/api/video/i422_buffer.h
@@ -61,8 +61,8 @@ class RTC_EXPORT I422Buffer : public I422BufferInterface {
// quirks in memory checkers
// (https://bugs.chromium.org/p/libyuv/issues/detail?id=377) and
// ffmpeg (http://crbug.com/390941).
- // TODO(nisse): Deprecated. Should be deleted if/when those issues
- // are resolved in a better way. Or in the mean time, use SetBlack.
+ // TODO(https://crbug.com/390941): Deprecated. Should be deleted if/when those
+ // issues are resolved in a better way. Or in the mean time, use SetBlack.
void InitializeData();
int width() const override;
diff --git a/api/video/i444_buffer.h b/api/video/i444_buffer.h
index 557bf4f3e0..f1e3f63114 100644
--- a/api/video/i444_buffer.h
+++ b/api/video/i444_buffer.h
@@ -58,8 +58,8 @@ class RTC_EXPORT I444Buffer : public I444BufferInterface {
// quirks in memory checkers
// (https://bugs.chromium.org/p/libyuv/issues/detail?id=377) and
// ffmpeg (http://crbug.com/390941).
- // TODO(nisse): Deprecated. Should be deleted if/when those issues
- // are resolved in a better way. Or in the mean time, use SetBlack.
+ // TODO(https://crbug.com/390941): Deprecated. Should be deleted if/when those
+ // issues are resolved in a better way. Or in the mean time, use SetBlack.
void InitializeData();
int width() const override;
diff --git a/api/video/nv12_buffer.h b/api/video/nv12_buffer.h
index 7baef2aeba..46a85f82e1 100644
--- a/api/video/nv12_buffer.h
+++ b/api/video/nv12_buffer.h
@@ -52,8 +52,8 @@ class RTC_EXPORT NV12Buffer : public NV12BufferInterface {
// quirks in memory checkers
// (https://bugs.chromium.org/p/libyuv/issues/detail?id=377) and
// ffmpeg (http://crbug.com/390941).
- // TODO(nisse): Deprecated. Should be deleted if/when those issues
- // are resolved in a better way. Or in the mean time, use SetBlack.
+ // TODO(https://crbug.com/390941): Deprecated. Should be deleted if/when those
+ // issues are resolved in a better way. Or in the mean time, use SetBlack.
void InitializeData();
// Scale the cropped area of `src` to the size of `this` buffer, and
diff --git a/api/video/video_frame.h b/api/video/video_frame.h
index 3d2c78598c..0299123a7a 100644
--- a/api/video/video_frame.h
+++ b/api/video/video_frame.h
@@ -166,19 +166,15 @@ class RTC_EXPORT VideoFrame {
int64_t timestamp_us() const { return timestamp_us_; }
void set_timestamp_us(int64_t timestamp_us) { timestamp_us_ = timestamp_us; }
- // TODO(nisse): After the cricket::VideoFrame and webrtc::VideoFrame
- // merge, timestamps other than timestamp_us will likely be
- // deprecated.
-
// Set frame timestamp (90kHz).
void set_timestamp(uint32_t timestamp) { timestamp_rtp_ = timestamp; }
// Get frame timestamp (90kHz).
uint32_t timestamp() const { return timestamp_rtp_; }
- // For now, transport_frame_id and rtp timestamp are the same.
- // TODO(nisse): Must be handled differently for QUIC.
- uint32_t transport_frame_id() const { return timestamp(); }
+ [[deprecated("Use timestamp()")]] uint32_t transport_frame_id() const {
+ return timestamp();
+ }
// Set capture ntp time in milliseconds.
void set_ntp_time_ms(int64_t ntp_time_ms) { ntp_time_ms_ = ntp_time_ms; }
@@ -219,7 +215,6 @@ class RTC_EXPORT VideoFrame {
}
// Get render time in milliseconds.
- // TODO(nisse): Deprecated. Migrate all users to timestamp_us().
int64_t render_time_ms() const;
// Return the underlying buffer. Never nullptr for a properly
@@ -229,7 +224,6 @@ class RTC_EXPORT VideoFrame {
void set_video_frame_buffer(
const rtc::scoped_refptr<VideoFrameBuffer>& buffer);
- // TODO(nisse): Deprecated.
// Return true if the frame is stored in a texture.
bool is_texture() const {
return video_frame_buffer()->type() == VideoFrameBuffer::Type::kNative;
diff --git a/api/video/video_stream_encoder_interface.h b/api/video/video_stream_encoder_interface.h
index f2d7e131e6..44affa4c16 100644
--- a/api/video/video_stream_encoder_interface.h
+++ b/api/video/video_stream_encoder_interface.h
@@ -74,8 +74,8 @@ class VideoStreamEncoderInterface {
// or frame rate may be reduced. The VideoStreamEncoder registers itself with
// `source`, and signals adaptation decisions to the source in the form of
// VideoSinkWants.
- // TODO(nisse): When adaptation logic is extracted from this class,
- // it no longer needs to know the source.
+ // TODO(bugs.webrtc.org/14246): When adaptation logic is extracted from this
+ // class, it no longer needs to know the source.
virtual void SetSource(
rtc::VideoSourceInterface<VideoFrame>* source,
const DegradationPreference& degradation_preference) = 0;
diff --git a/api/video/video_stream_encoder_observer.h b/api/video/video_stream_encoder_observer.h
index ea8196ce6d..01e1f661fe 100644
--- a/api/video/video_stream_encoder_observer.h
+++ b/api/video/video_stream_encoder_observer.h
@@ -53,7 +53,6 @@ class VideoStreamEncoderObserver : public CpuOveruseMetricsObserver {
bool framerate_scaling_enabled;
};
- // TODO(nisse): Duplicates enum EncodedImageCallback::DropReason.
enum class DropReason {
kSource,
kEncoderQueue,
@@ -66,7 +65,7 @@ class VideoStreamEncoderObserver : public CpuOveruseMetricsObserver {
virtual void OnIncomingFrame(int width, int height) = 0;
- // TODO(nisse): Merge into one callback per encoded frame.
+ // TODO(bugs.webrtc.org/8504): Merge into one callback per encoded frame.
using CpuOveruseMetricsObserver::OnEncodedFrameTimeMeasured;
virtual void OnSendEncodedImage(const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_info) = 0;
@@ -105,10 +104,10 @@ class VideoStreamEncoderObserver : public CpuOveruseMetricsObserver {
// down.
virtual void OnEncoderInternalScalerUpdate(bool is_scaled) {}
- // TODO(nisse): VideoStreamEncoder wants to query the stats, which makes this
- // not a pure observer. GetInputFrameRate is needed for the cpu adaptation, so
- // can be deleted if that responsibility is moved out to a VideoStreamAdaptor
- // class.
+ // TODO(bugs.webrtc.org/14246): VideoStreamEncoder wants to query the stats,
+ // which makes this not a pure observer. GetInputFrameRate is needed for the
+ // cpu adaptation, so can be deleted if that responsibility is moved out to a
+ // VideoStreamAdaptor class.
virtual int GetInputFrameRate() const = 0;
};
diff --git a/api/video_codecs/video_encoder.h b/api/video_codecs/video_encoder.h
index 94d7287f78..30ec58e807 100644
--- a/api/video_codecs/video_encoder.h
+++ b/api/video_codecs/video_encoder.h
@@ -66,6 +66,10 @@ class RTC_EXPORT EncodedImageCallback {
// kDroppedByMediaOptimizations - dropped by MediaOptimizations (for rate
// limiting purposes).
// kDroppedByEncoder - dropped by encoder's internal rate limiter.
+ // TODO(bugs.webrtc.org/10164): Delete this enum? It duplicates the more
+ // general VideoStreamEncoderObserver::DropReason. Also,
+ // kDroppedByMediaOptimizations is not produced by any encoder, but by
+ // VideoStreamEncoder.
enum class DropReason : uint8_t {
kDroppedByMediaOptimizations,
kDroppedByEncoder
@@ -96,11 +100,9 @@ class RTC_EXPORT VideoEncoder {
struct KOff {};
public:
- // TODO(nisse): Would be nicer if kOff were a constant ScalingSettings
- // rather than a magic value. However, absl::optional is not trivially copy
- // constructible, and hence a constant ScalingSettings needs a static
- // initializer, which is strongly discouraged in Chrome. We can hopefully
- // fix this when we switch to absl::optional or std::optional.
+ // TODO(bugs.webrtc.org/9078): Since absl::optional should be trivially copy
+ // constructible, this magic value can likely be replaced by a constexpr
+ // ScalingSettings value.
static constexpr KOff kOff = {};
ScalingSettings(int low, int high);
diff --git a/api/video_codecs/video_encoder_config.h b/api/video_codecs/video_encoder_config.h
index 4076208b56..86d89d53da 100644
--- a/api/video_codecs/video_encoder_config.h
+++ b/api/video_codecs/video_encoder_config.h
@@ -141,7 +141,7 @@ class VideoEncoderConfig {
~VideoEncoderConfig();
std::string ToString() const;
- // TODO(nisse): Consolidate on one of these.
+ // TODO(bugs.webrtc.org/6883): Consolidate on one of these.
VideoCodecType codec_type;
SdpVideoFormat video_format;
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 1573765a98..6954447256 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -644,7 +644,8 @@ void ChannelReceive::OnRtpPacket(const RtpPacketReceived& packet) {
const auto& it = payload_type_frequencies_.find(packet.PayloadType());
if (it == payload_type_frequencies_.end())
return;
- // TODO(nisse): Set payload_type_frequency earlier, when packet is parsed.
+ // TODO(bugs.webrtc.org/7135): Set payload_type_frequency earlier, when packet
+ // is parsed.
RtpPacketReceived packet_copy(packet);
packet_copy.set_payload_type_frequency(it->second);
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 05341b9f29..a1b1a63150 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -62,7 +62,7 @@ class ChannelSend : public ChannelSendInterface,
public RtcpPacketTypeCounterObserver {
public:
// TODO(nisse): Make OnUplinkPacketLossRate public, and delete friend
- // declaration.
+ // declaration. Or delete indirection via VoERtcpObserver.
friend class VoERtcpObserver;
ChannelSend(Clock* clock,
diff --git a/call/call.cc b/call/call.cc
index 1bb107c9b9..ad08b8a6f3 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -400,8 +400,8 @@ class Call final : public webrtc::Call,
RTC_GUARDED_BY(worker_thread_);
std::set<VideoReceiveStream2*> video_receive_streams_
RTC_GUARDED_BY(worker_thread_);
- // TODO(nisse): Should eventually be injected at creation,
- // with a single object in the bundled case.
+ // TODO(bugs.webrtc.org/7135, bugs.webrtc.org/9719): Should eventually be
+ // injected at creation, with a single object in the bundled case.
RtpStreamReceiverController audio_receiver_controller_
RTC_GUARDED_BY(worker_thread_);
RtpStreamReceiverController video_receiver_controller_
@@ -1550,12 +1550,6 @@ void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
if (!use_send_side_bwe && header.extension.hasTransportSequenceNumber) {
// Inconsistent configuration of send side BWE. Do nothing.
- // TODO(nisse): Without this check, we may produce RTCP feedback
- // packets even when not negotiated. But it would be cleaner to
- // move the check down to RTCPSender::SendFeedbackPacket, which
- // would also help the PacketRouter to select an appropriate rtp
- // module in the case that some, but not all, have RTCP feedback
- // enabled.
return;
}
// For audio, we only support send side BWE.
diff --git a/call/rtp_config.h b/call/rtp_config.h
index c3b5b4a255..0cc9466a9f 100644
--- a/call/rtp_config.h
+++ b/call/rtp_config.h
@@ -104,7 +104,7 @@ struct RtpConfig {
// changing codec without recreating the VideoSendStream. Then these
// fields must be removed, and association between payload type and codec
// must move above the per-stream level. Ownership could be with
- // RtpTransportControllerSend, with a reference from PayloadRouter, where
+ // RtpTransportControllerSend, with a reference from RtpVideoSender, where
// the latter would be responsible for mapping the codec type of encoded
// images to the right payload type.
std::string payload_name;
diff --git a/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/modules/video_coding/codecs/h264/h264_decoder_impl.cc
index cb83a93829..e654e1835b 100644
--- a/modules/video_coding/codecs/h264/h264_decoder_impl.cc
+++ b/modules/video_coding/codecs/h264/h264_decoder_impl.cc
@@ -114,8 +114,8 @@ int H264DecoderImpl::AVGetBuffer2(AVCodecContext* context,
// FFmpeg expects the initial allocation to be zero-initialized according to
// http://crbug.com/390941. Our pool is set up to zero-initialize new buffers.
- // TODO(nisse): Delete that feature from the video pool, instead add
- // an explicit call to InitializeData here.
+ // TODO(https://crbug.com/390941): Delete that feature from the video pool,
+ // instead add an explicit call to InitializeData here.
rtc::scoped_refptr<PlanarYuvBuffer> frame_buffer;
rtc::scoped_refptr<I444Buffer> i444_buffer;
rtc::scoped_refptr<I420Buffer> i420_buffer;