Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h
We can then finally delete the top-level common_types.h, and the
corresponding build target webrtc_common.
Bug: webrtc:7660
Change-Id: I1c1096541477586d90774c7a3405b9d36edec14a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182800
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32044}
diff --git a/api/BUILD.gn b/api/BUILD.gn
index 50a9c5a..c257a7d 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -68,7 +68,6 @@
]
deps = [
":array_view",
- "..:webrtc_common",
"units:timestamp",
"video:video_rtp_headers",
]
diff --git a/api/rtp_headers.h b/api/rtp_headers.h
index 454149c..b9a97c8 100644
--- a/api/rtp_headers.h
+++ b/api/rtp_headers.h
@@ -23,7 +23,6 @@
#include "api/video/video_content_type.h"
#include "api/video/video_rotation.h"
#include "api/video/video_timing.h"
-#include "common_types.h" // NOLINT (build/include)
namespace webrtc {
@@ -142,7 +141,7 @@
bool has_video_timing;
VideoSendTiming video_timing;
- PlayoutDelay playout_delay = {-1, -1};
+ VideoPlayoutDelay playout_delay;
// For identification of a stream when ssrc is not signaled. See
// https://tools.ietf.org/html/draft-ietf-avtext-rid-09
diff --git a/api/video/BUILD.gn b/api/video/BUILD.gn
index e864e03..b85c2b6 100644
--- a/api/video/BUILD.gn
+++ b/api/video/BUILD.gn
@@ -135,7 +135,6 @@
"..:refcountedbase",
"..:rtp_packet_info",
"..:scoped_refptr",
- "../..:webrtc_common",
"../../rtc_base:checks",
"../../rtc_base:deprecation",
"../../rtc_base:rtc_base_approved",
diff --git a/api/video/encoded_image.h b/api/video/encoded_image.h
index 35c2584..cb0f2eb 100644
--- a/api/video/encoded_image.h
+++ b/api/video/encoded_image.h
@@ -25,7 +25,6 @@
#include "api/video/video_frame_type.h"
#include "api/video/video_rotation.h"
#include "api/video/video_timing.h"
-#include "common_types.h" // NOLINT(build/include_directory)
#include "rtc_base/checks.h"
#include "rtc_base/deprecation.h"
#include "rtc_base/ref_count.h"
@@ -183,7 +182,7 @@
// When an application indicates non-zero values here, it is taken as an
// indication that all future frames will be constrained with those limits
// until the application indicates a change again.
- PlayoutDelay playout_delay_ = {-1, -1};
+ VideoPlayoutDelay playout_delay_;
struct Timing {
uint8_t flags = VideoSendTiming::kInvalid;
diff --git a/api/video/video_timing.h b/api/video/video_timing.h
index 4cc75dd..fbd9225 100644
--- a/api/video/video_timing.h
+++ b/api/video/video_timing.h
@@ -100,6 +100,30 @@
uint8_t flags; // Flags indicating validity and/or why tracing was triggered.
};
+// Minimum and maximum playout delay values from capture to render.
+// These are best effort values.
+//
+// A value < 0 indicates no change from previous valid value.
+//
+// min = max = 0 indicates that the receiver should try and render
+// frame as soon as possible.
+//
+// min = x, max = y indicates that the receiver is free to adapt
+// in the range (x, y) based on network jitter.
+struct VideoPlayoutDelay {
+ VideoPlayoutDelay() = default;
+ VideoPlayoutDelay(int min_ms, int max_ms) : min_ms(min_ms), max_ms(max_ms) {}
+ int min_ms = -1;
+ int max_ms = -1;
+
+ bool operator==(const VideoPlayoutDelay& rhs) const {
+ return min_ms == rhs.min_ms && max_ms == rhs.max_ms;
+ }
+};
+
+// TODO(bugs.webrtc.org/7660): Old name, delete after downstream use is updated.
+using PlayoutDelay = VideoPlayoutDelay;
+
} // namespace webrtc
#endif // API_VIDEO_VIDEO_TIMING_H_
