diff options
author | Niels Möller <nisse@webrtc.org> | 2022-07-05 08:55:19 +0200 |
---|---|---|
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | 2022-07-05 09:59:33 +0000 |
commit | 6939f63ca149b78b2229eea7e3a69f143b3e261b (patch) | |
tree | 4a0934ba4319e8882a85a0e3762e86b4fbfddfb0 | |
parent | c8152fe4a8d62323d01924679dca8acc602ccf48 (diff) | |
download | webrtc-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.gn | 1 | ||||
-rw-r--r-- | api/video/encoded_image.h | 5 | ||||
-rw-r--r-- | api/video/i420_buffer.h | 4 | ||||
-rw-r--r-- | api/video/i422_buffer.h | 4 | ||||
-rw-r--r-- | api/video/i444_buffer.h | 4 | ||||
-rw-r--r-- | api/video/nv12_buffer.h | 4 | ||||
-rw-r--r-- | api/video/video_frame.h | 12 | ||||
-rw-r--r-- | api/video/video_stream_encoder_interface.h | 4 | ||||
-rw-r--r-- | api/video/video_stream_encoder_observer.h | 11 | ||||
-rw-r--r-- | api/video_codecs/video_encoder.h | 12 | ||||
-rw-r--r-- | api/video_codecs/video_encoder_config.h | 2 | ||||
-rw-r--r-- | audio/channel_receive.cc | 3 | ||||
-rw-r--r-- | audio/channel_send.cc | 2 | ||||
-rw-r--r-- | call/call.cc | 10 | ||||
-rw-r--r-- | call/rtp_config.h | 2 | ||||
-rw-r--r-- | modules/video_coding/codecs/h264/h264_decoder_impl.cc | 4 |
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; |