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Add RTP timestamp to contributing sources

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RTP timestamp was recently added to contributing sources in the WebRTC
specification. This CL implements that change in WebRTC.

Bug: webrtc:10650
Change-Id: Ic0ccfbea7049a5b66063fa6cf60d01d5bd713132
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137515
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28020}
This commit is contained in:
Johannes Kron
2019-05-21 13:19:22 +02:00
committed by Commit Bot
parent afb8d5cdae
commit b5d918324c
7 changed files with 125 additions and 61 deletions
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11
modules/rtp_rtcp/source/contributing_sources.h
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@ -32,8 +32,10 @@ class ContributingSources {
ContributingSources();
~ContributingSources();
void Update(int64_t now_ms, rtc::ArrayView<const uint32_t> csrcs,
absl::optional<uint8_t> audio_level);
void Update(int64_t now_ms,
rtc::ArrayView<const uint32_t> csrcs,
absl::optional<uint8_t> audio_level,
uint32_t rtp_timestamp);
// Returns contributing sources seen the last 10 s.
std::vector<RtpSource> GetSources(int64_t now_ms) const;
@ -41,10 +43,13 @@ class ContributingSources {
private:
struct Entry {
Entry();
Entry(int64_t timestamp_ms, absl::optional<uint8_t> audio_level);
Entry(int64_t timestamp_ms,
absl::optional<uint8_t> audio_level,
uint32_t rtp_timestamp);
int64_t last_seen_ms;
absl::optional<uint8_t> audio_level;
uint32_t rtp_timestamp;
};
void DeleteOldEntries(int64_t now_ms);