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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}
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c8152fe4a8
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16 changed files with 36 additions and 48 deletions
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@ -58,7 +58,6 @@ rtc_library("field_trial_based_config") {
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absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
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}
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# TODO(nisse): Rename?
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rtc_source_set("datagram_transport_interface") {
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visibility = [ "*" ]
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sources = [ "data_channel_transport_interface.h" ]
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@ -78,9 +78,8 @@ class RTC_EXPORT EncodedImage {
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EncodedImage& operator=(EncodedImage&&);
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EncodedImage& operator=(const EncodedImage&);
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// TODO(nisse): Change style to timestamp(), set_timestamp(), for consistency
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// with the VideoFrame class.
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// Set frame timestamp (90kHz).
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// TODO(bugs.webrtc.org/9378): Change style to timestamp(), set_timestamp(),
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// for consistency with the VideoFrame class. Set frame timestamp (90kHz).
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void SetTimestamp(uint32_t timestamp) { timestamp_rtp_ = timestamp; }
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// Get frame timestamp (90kHz).
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@ -65,8 +65,8 @@ class RTC_EXPORT I420Buffer : public I420BufferInterface {
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// quirks in memory checkers
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// (https://bugs.chromium.org/p/libyuv/issues/detail?id=377) and
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// ffmpeg (http://crbug.com/390941).
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// TODO(nisse): Deprecated. Should be deleted if/when those issues
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// are resolved in a better way. Or in the mean time, use SetBlack.
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// TODO(https://crbug.com/390941): Deprecated. Should be deleted if/when those
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// issues are resolved in a better way. Or in the mean time, use SetBlack.
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void InitializeData();
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int width() const override;
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@ -61,8 +61,8 @@ class RTC_EXPORT I422Buffer : public I422BufferInterface {
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// quirks in memory checkers
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// (https://bugs.chromium.org/p/libyuv/issues/detail?id=377) and
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// ffmpeg (http://crbug.com/390941).
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// TODO(nisse): Deprecated. Should be deleted if/when those issues
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// are resolved in a better way. Or in the mean time, use SetBlack.
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// TODO(https://crbug.com/390941): Deprecated. Should be deleted if/when those
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// issues are resolved in a better way. Or in the mean time, use SetBlack.
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void InitializeData();
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int width() const override;
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@ -58,8 +58,8 @@ class RTC_EXPORT I444Buffer : public I444BufferInterface {
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// quirks in memory checkers
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// (https://bugs.chromium.org/p/libyuv/issues/detail?id=377) and
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// ffmpeg (http://crbug.com/390941).
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// TODO(nisse): Deprecated. Should be deleted if/when those issues
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// are resolved in a better way. Or in the mean time, use SetBlack.
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// TODO(https://crbug.com/390941): Deprecated. Should be deleted if/when those
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// issues are resolved in a better way. Or in the mean time, use SetBlack.
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void InitializeData();
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int width() const override;
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@ -52,8 +52,8 @@ class RTC_EXPORT NV12Buffer : public NV12BufferInterface {
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// quirks in memory checkers
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// (https://bugs.chromium.org/p/libyuv/issues/detail?id=377) and
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// ffmpeg (http://crbug.com/390941).
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// TODO(nisse): Deprecated. Should be deleted if/when those issues
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// are resolved in a better way. Or in the mean time, use SetBlack.
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// TODO(https://crbug.com/390941): Deprecated. Should be deleted if/when those
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// issues are resolved in a better way. Or in the mean time, use SetBlack.
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void InitializeData();
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// Scale the cropped area of `src` to the size of `this` buffer, and
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@ -166,19 +166,15 @@ class RTC_EXPORT VideoFrame {
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int64_t timestamp_us() const { return timestamp_us_; }
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void set_timestamp_us(int64_t timestamp_us) { timestamp_us_ = timestamp_us; }
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// TODO(nisse): After the cricket::VideoFrame and webrtc::VideoFrame
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// merge, timestamps other than timestamp_us will likely be
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// deprecated.
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// Set frame timestamp (90kHz).
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void set_timestamp(uint32_t timestamp) { timestamp_rtp_ = timestamp; }
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// Get frame timestamp (90kHz).
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uint32_t timestamp() const { return timestamp_rtp_; }
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// For now, transport_frame_id and rtp timestamp are the same.
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// TODO(nisse): Must be handled differently for QUIC.
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uint32_t transport_frame_id() const { return timestamp(); }
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[[deprecated("Use timestamp()")]] uint32_t transport_frame_id() const {
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return timestamp();
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}
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// Set capture ntp time in milliseconds.
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void set_ntp_time_ms(int64_t ntp_time_ms) { ntp_time_ms_ = ntp_time_ms; }
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@ -219,7 +215,6 @@ class RTC_EXPORT VideoFrame {
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}
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// Get render time in milliseconds.
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// TODO(nisse): Deprecated. Migrate all users to timestamp_us().
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int64_t render_time_ms() const;
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// Return the underlying buffer. Never nullptr for a properly
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@ -229,7 +224,6 @@ class RTC_EXPORT VideoFrame {
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void set_video_frame_buffer(
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const rtc::scoped_refptr<VideoFrameBuffer>& buffer);
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// TODO(nisse): Deprecated.
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// Return true if the frame is stored in a texture.
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bool is_texture() const {
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return video_frame_buffer()->type() == VideoFrameBuffer::Type::kNative;
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@ -74,8 +74,8 @@ class VideoStreamEncoderInterface {
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// or frame rate may be reduced. The VideoStreamEncoder registers itself with
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// `source`, and signals adaptation decisions to the source in the form of
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// VideoSinkWants.
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// TODO(nisse): When adaptation logic is extracted from this class,
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// it no longer needs to know the source.
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// TODO(bugs.webrtc.org/14246): When adaptation logic is extracted from this
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// class, it no longer needs to know the source.
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virtual void SetSource(
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rtc::VideoSourceInterface<VideoFrame>* source,
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const DegradationPreference& degradation_preference) = 0;
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@ -53,7 +53,6 @@ class VideoStreamEncoderObserver : public CpuOveruseMetricsObserver {
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bool framerate_scaling_enabled;
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};
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// TODO(nisse): Duplicates enum EncodedImageCallback::DropReason.
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enum class DropReason {
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kSource,
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kEncoderQueue,
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@ -66,7 +65,7 @@ class VideoStreamEncoderObserver : public CpuOveruseMetricsObserver {
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virtual void OnIncomingFrame(int width, int height) = 0;
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// TODO(nisse): Merge into one callback per encoded frame.
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// TODO(bugs.webrtc.org/8504): Merge into one callback per encoded frame.
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using CpuOveruseMetricsObserver::OnEncodedFrameTimeMeasured;
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virtual void OnSendEncodedImage(const EncodedImage& encoded_image,
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const CodecSpecificInfo* codec_info) = 0;
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@ -105,10 +104,10 @@ class VideoStreamEncoderObserver : public CpuOveruseMetricsObserver {
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// down.
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virtual void OnEncoderInternalScalerUpdate(bool is_scaled) {}
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// TODO(nisse): VideoStreamEncoder wants to query the stats, which makes this
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// not a pure observer. GetInputFrameRate is needed for the cpu adaptation, so
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// can be deleted if that responsibility is moved out to a VideoStreamAdaptor
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// class.
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// TODO(bugs.webrtc.org/14246): VideoStreamEncoder wants to query the stats,
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// which makes this not a pure observer. GetInputFrameRate is needed for the
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// cpu adaptation, so can be deleted if that responsibility is moved out to a
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// VideoStreamAdaptor class.
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virtual int GetInputFrameRate() const = 0;
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};
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@ -66,6 +66,10 @@ class RTC_EXPORT EncodedImageCallback {
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// kDroppedByMediaOptimizations - dropped by MediaOptimizations (for rate
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// limiting purposes).
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// kDroppedByEncoder - dropped by encoder's internal rate limiter.
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// TODO(bugs.webrtc.org/10164): Delete this enum? It duplicates the more
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// general VideoStreamEncoderObserver::DropReason. Also,
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// kDroppedByMediaOptimizations is not produced by any encoder, but by
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// VideoStreamEncoder.
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enum class DropReason : uint8_t {
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kDroppedByMediaOptimizations,
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kDroppedByEncoder
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@ -96,11 +100,9 @@ class RTC_EXPORT VideoEncoder {
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struct KOff {};
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public:
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// TODO(nisse): Would be nicer if kOff were a constant ScalingSettings
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// rather than a magic value. However, absl::optional is not trivially copy
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// constructible, and hence a constant ScalingSettings needs a static
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// initializer, which is strongly discouraged in Chrome. We can hopefully
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// fix this when we switch to absl::optional or std::optional.
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// TODO(bugs.webrtc.org/9078): Since absl::optional should be trivially copy
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// constructible, this magic value can likely be replaced by a constexpr
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// ScalingSettings value.
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static constexpr KOff kOff = {};
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ScalingSettings(int low, int high);
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@ -141,7 +141,7 @@ class VideoEncoderConfig {
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~VideoEncoderConfig();
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std::string ToString() const;
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// TODO(nisse): Consolidate on one of these.
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// TODO(bugs.webrtc.org/6883): Consolidate on one of these.
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VideoCodecType codec_type;
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SdpVideoFormat video_format;
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@ -644,7 +644,8 @@ void ChannelReceive::OnRtpPacket(const RtpPacketReceived& packet) {
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const auto& it = payload_type_frequencies_.find(packet.PayloadType());
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if (it == payload_type_frequencies_.end())
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return;
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// TODO(nisse): Set payload_type_frequency earlier, when packet is parsed.
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// TODO(bugs.webrtc.org/7135): Set payload_type_frequency earlier, when packet
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// is parsed.
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RtpPacketReceived packet_copy(packet);
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packet_copy.set_payload_type_frequency(it->second);
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@ -62,7 +62,7 @@ class ChannelSend : public ChannelSendInterface,
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public RtcpPacketTypeCounterObserver {
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public:
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// TODO(nisse): Make OnUplinkPacketLossRate public, and delete friend
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// declaration.
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// declaration. Or delete indirection via VoERtcpObserver.
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friend class VoERtcpObserver;
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ChannelSend(Clock* clock,
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10
call/call.cc
10
call/call.cc
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@ -400,8 +400,8 @@ class Call final : public webrtc::Call,
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RTC_GUARDED_BY(worker_thread_);
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std::set<VideoReceiveStream2*> video_receive_streams_
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RTC_GUARDED_BY(worker_thread_);
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// TODO(nisse): Should eventually be injected at creation,
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// with a single object in the bundled case.
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// TODO(bugs.webrtc.org/7135, bugs.webrtc.org/9719): Should eventually be
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// injected at creation, with a single object in the bundled case.
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RtpStreamReceiverController audio_receiver_controller_
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RTC_GUARDED_BY(worker_thread_);
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RtpStreamReceiverController video_receiver_controller_
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@ -1550,12 +1550,6 @@ void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
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if (!use_send_side_bwe && header.extension.hasTransportSequenceNumber) {
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// Inconsistent configuration of send side BWE. Do nothing.
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// TODO(nisse): Without this check, we may produce RTCP feedback
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// packets even when not negotiated. But it would be cleaner to
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// move the check down to RTCPSender::SendFeedbackPacket, which
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// would also help the PacketRouter to select an appropriate rtp
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// module in the case that some, but not all, have RTCP feedback
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// enabled.
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return;
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}
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// For audio, we only support send side BWE.
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@ -104,7 +104,7 @@ struct RtpConfig {
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// changing codec without recreating the VideoSendStream. Then these
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// fields must be removed, and association between payload type and codec
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// must move above the per-stream level. Ownership could be with
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// RtpTransportControllerSend, with a reference from PayloadRouter, where
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// RtpTransportControllerSend, with a reference from RtpVideoSender, where
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// the latter would be responsible for mapping the codec type of encoded
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// images to the right payload type.
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std::string payload_name;
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@ -114,8 +114,8 @@ int H264DecoderImpl::AVGetBuffer2(AVCodecContext* context,
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// FFmpeg expects the initial allocation to be zero-initialized according to
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// http://crbug.com/390941. Our pool is set up to zero-initialize new buffers.
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// TODO(nisse): Delete that feature from the video pool, instead add
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// an explicit call to InitializeData here.
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// TODO(https://crbug.com/390941): Delete that feature from the video pool,
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// instead add an explicit call to InitializeData here.
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rtc::scoped_refptr<PlanarYuvBuffer> frame_buffer;
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rtc::scoped_refptr<I444Buffer> i444_buffer;
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rtc::scoped_refptr<I420Buffer> i420_buffer;
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