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Implement PlayoutDelay extension as a trait

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to be used with rtp::Packet class

BUG=webrtc:1994
R=isheriff@chromium.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2224063004 .

Cr-Commit-Position: refs/heads/master@{#14105}
This commit is contained in:
Danil Chapovalov
2016-09-07 15:08:13 +02:00
parent 10e8f8e2a4
commit 08b0351ddd
3 changed files with 67 additions and 1 deletions
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17
webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
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@ -75,5 +75,22 @@ class VideoOrientation {
static bool Write(uint8_t* data, uint8_t value);
};
class PlayoutDelayLimits {
public:
static constexpr RTPExtensionType kId = kRtpExtensionPlayoutDelay;
static constexpr uint8_t kValueSizeBytes = 3;
static const char* kName;
static bool IsSupportedFor(MediaType type);
// Playout delay in milliseconds. A playout delay limit (min or max)
// has 12 bits allocated. This allows a range of 0-4095 values which
// translates to a range of 0-40950 in milliseconds.
static constexpr int kGranularityMs = 10;
// Maximum playout delay value in milliseconds.
static constexpr int kMaxMs = 0xfff * kGranularityMs; // 40950.
static bool Parse(const uint8_t* data, PlayoutDelay* playout_delay);
static bool Write(uint8_t* data, const PlayoutDelay& playout_delay);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_