zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Add absolute capture time property to rtp sources.

Browse Source 
This part of the effort to implement A/V sync metric.

Bug: webrtc:10739
Change-Id: I4adba1b99b37b31868168e37d9aa8e03f8ea6d4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159886
Commit-Queue: Ruslan Burakov <kuddai@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ruslan Burakov <kuddai@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29849}
This commit is contained in:
Ruslan Burakov
2019-11-20 16:48:34 +01:00
committed by Commit Bot
parent bb55e0bc72
commit d51cc7bd71
5 changed files with 86 additions and 28 deletions
Download Patch File Download Diff File Expand all files Collapse all files

39
api/transport/rtp/rtp_source.h
View File

@ -14,6 +14,7 @@
#include <stdint.h>
#include "absl/types/optional.h"
#include "api/rtp_headers.h"
#include "rtc_base/checks.h"
namespace webrtc {
@ -25,17 +26,35 @@ enum class RtpSourceType {
class RtpSource {
public:
struct Extensions {
absl::optional<uint8_t> audio_level;
absl::optional<AbsoluteCaptureTime> absolute_capture_time;
};
RtpSource() = delete;
// TODO(bugs.webrtc.org/10739): Remove this constructor once all clients
// migrate to the version with absolute capture time.
RtpSource(int64_t timestamp_ms,
uint32_t source_id,
RtpSourceType source_type,
absl::optional<uint8_t> audio_level,
uint32_t rtp_timestamp)
: RtpSource(timestamp_ms,
source_id,
source_type,
rtp_timestamp,
{audio_level, absl::nullopt}) {}
RtpSource(int64_t timestamp_ms,
uint32_t source_id,
RtpSourceType source_type,
uint32_t rtp_timestamp,
const RtpSource::Extensions& extensions)
: timestamp_ms_(timestamp_ms),
source_id_(source_id),
source_type_(source_type),
audio_level_(audio_level),
extensions_(extensions),
rtp_timestamp_(rtp_timestamp) {}
RtpSource(const RtpSource&) = default;
@ -54,16 +73,26 @@ class RtpSource {
// The source can be either a contributing source or a synchronization source.
RtpSourceType source_type() const { return source_type_; }
absl::optional<uint8_t> audio_level() const { return audio_level_; }
absl::optional<uint8_t> audio_level() const {
return extensions_.audio_level;
}
void set_audio_level(const absl::optional<uint8_t>& level) {
audio_level_ = level;
extensions_.audio_level = level;
}
uint32_t rtp_timestamp() const { return rtp_timestamp_; }
absl::optional<AbsoluteCaptureTime> absolute_capture_time() const {
return extensions_.absolute_capture_time;
}
bool operator==(const RtpSource& o) const {
return timestamp_ms_ == o.timestamp_ms() && source_id_ == o.source_id() &&
source_type_ == o.source_type() && audio_level_ == o.audio_level_ &&
source_type_ == o.source_type() &&
extensions_.audio_level == o.extensions_.audio_level &&
extensions_.absolute_capture_time ==
o.extensions_.absolute_capture_time &&
rtp_timestamp_ == o.rtp_timestamp();
}
@ -71,7 +100,7 @@ class RtpSource {
int64_t timestamp_ms_;
uint32_t source_id_;
RtpSourceType source_type_;
absl::optional<uint8_t> audio_level_;
RtpSource::Extensions extensions_;
uint32_t rtp_timestamp_;
};