Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.
This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
Bug: webrtc:10668
Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28434}
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 8aad203..9afc0e4 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -1017,6 +1017,7 @@
"..:module_api_public",
"../../api:array_view",
"../../api:rtp_headers",
+ "../../api:rtp_packet_info",
"../../api:scoped_refptr",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
@@ -1029,6 +1030,7 @@
"../../rtc_base:safe_minmax",
"../../rtc_base:sanitizer",
"../../rtc_base/system:fallthrough",
+ "../../system_wrappers",
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/memory",
@@ -1066,6 +1068,7 @@
"../../api/audio_codecs:audio_codecs_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
+ "../../system_wrappers",
"../rtp_rtcp",
"../rtp_rtcp:rtp_rtcp_format",
"//third_party/abseil-cpp/absl/types:optional",
@@ -1591,6 +1594,7 @@
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
+ "../../system_wrappers",
"../../test:fileutils",
"../../test:test_support",
"//testing/gtest",