Remove SetLatency/GetLatency from MediaSourceInterface API level
Bug: webrtc:10287
Change-Id: I74fad31db98b75791085688438064f9510b0b6fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133165
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27692}diff --git a/pc/jitter_buffer_delay_interface.h b/pc/jitter_buffer_delay_interface.h
new file mode 100644
index 0000000..f2132d3
--- /dev/null
+++ b/pc/jitter_buffer_delay_interface.h
@@ -0,0 +1,39 @@
+/*
+ * Copyright 2019 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 PC_JITTER_BUFFER_DELAY_INTERFACE_H_
+#define PC_JITTER_BUFFER_DELAY_INTERFACE_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "media/base/delayable.h"
+#include "rtc_base/ref_count.h"
+
+namespace webrtc {
+
+// JitterBufferDelay delivers user's queries to the underlying media channel. It
+// can describe either video or audio delay for receiving stream. "Interface"
+// suffix in the interface name is required to be compatible with api/proxy.cc
+class JitterBufferDelayInterface : public rtc::RefCountInterface {
+ public:
+ // OnStart allows to uniqely identify to which receiving stream playout
+ // delay must correpond through |media_channel| and |ssrc| pair.
+ virtual void OnStart(cricket::Delayable* media_channel, uint32_t ssrc) = 0;
+
+ // Indicates that underlying receiving stream is stopped.
+ virtual void OnStop() = 0;
+
+ virtual void Set(absl::optional<double> delay_seconds) = 0;
+};
+
+} // namespace webrtc
+
+#endif // PC_JITTER_BUFFER_DELAY_INTERFACE_H_