summaryrefslogtreecommitdiffstats
path: root/third_party/libwebrtc/webrtc/video/stream_synchronization.h
diff options
context:
space:
mode:
Diffstat (limited to 'third_party/libwebrtc/webrtc/video/stream_synchronization.h')
-rw-r--r--third_party/libwebrtc/webrtc/video/stream_synchronization.h63
1 files changed, 63 insertions, 0 deletions
diff --git a/third_party/libwebrtc/webrtc/video/stream_synchronization.h b/third_party/libwebrtc/webrtc/video/stream_synchronization.h
new file mode 100644
index 0000000000..52b8bde21d
--- /dev/null
+++ b/third_party/libwebrtc/webrtc/video/stream_synchronization.h
@@ -0,0 +1,63 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef VIDEO_STREAM_SYNCHRONIZATION_H_
+#define VIDEO_STREAM_SYNCHRONIZATION_H_
+
+#include <list>
+
+#include "system_wrappers/include/rtp_to_ntp_estimator.h"
+#include "typedefs.h" // NOLINT(build/include)
+
+namespace webrtc {
+
+class StreamSynchronization {
+ public:
+ struct Measurements {
+ Measurements() : latest_receive_time_ms(0), latest_timestamp(0) {}
+ RtpToNtpEstimator rtp_to_ntp;
+ int64_t latest_receive_time_ms;
+ uint32_t latest_timestamp;
+ };
+
+ StreamSynchronization(int video_stream_id, int audio_stream_id);
+
+ bool ComputeDelays(int relative_delay_ms,
+ int current_audio_delay_ms,
+ int* extra_audio_delay_ms,
+ int* total_video_delay_target_ms);
+
+ // On success |relative_delay| contains the number of milliseconds later video
+ // is rendered relative audio. If audio is played back later than video a
+ // |relative_delay| will be negative.
+ static bool ComputeRelativeDelay(const Measurements& audio_measurement,
+ const Measurements& video_measurement,
+ int* relative_delay_ms);
+ // Set target buffering delay - All audio and video will be delayed by at
+ // least target_delay_ms.
+ void SetTargetBufferingDelay(int target_delay_ms);
+
+ private:
+ struct SynchronizationDelays {
+ int extra_video_delay_ms = 0;
+ int last_video_delay_ms = 0;
+ int extra_audio_delay_ms = 0;
+ int last_audio_delay_ms = 0;
+ };
+
+ SynchronizationDelays channel_delay_;
+ const int video_stream_id_;
+ const int audio_stream_id_;
+ int base_target_delay_ms_;
+ int avg_diff_ms_;
+};
+} // namespace webrtc
+
+#endif // VIDEO_STREAM_SYNCHRONIZATION_H_