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Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.

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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}
This commit is contained in:
Chen Xing
2019-07-01 17:16:32 +02:00
committed by Commit Bot
parent 62eb89d221
commit 3e8ef940fe
23 changed files with 195 additions and 26 deletions
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4
modules/audio_coding/BUILD.gn
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@ -1017,6 +1017,7 @@ rtc_static_library("neteq") {
"..: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_static_library("neteq") {
"../../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 @@ rtc_source_set("neteq_tools_minimal") {
"../../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 @@ if (rtc_include_tests) {
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
"../../test:fileutils",
"../../test:test_support",
"//testing/gtest",