Index: webrtc/modules/audio_coding/neteq/tools/neteq_input.h |
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_input.h b/webrtc/modules/audio_coding/neteq/tools/neteq_input.h |
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index 0000000000000000000000000000000000000000..8abec6303fc70c4e01fd24dfc2cfac97bfef896c |
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+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_input.h |
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+/* |
+ * Copyright (c) 2016 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_ |
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_ |
+ |
+#include <algorithm> |
+#include <memory> |
+ |
+#include "webrtc/base/buffer.h" |
+#include "webrtc/base/optional.h" |
+#include "webrtc/modules/audio_coding/neteq/tools/packet.h" |
+#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" |
+#include "webrtc/modules/include/module_common_types.h" |
+ |
+namespace webrtc { |
+namespace test { |
+ |
+// Interface class for input to the NetEqTest class. |
+class NetEqInput { |
+ public: |
+ struct PacketData { |
+ WebRtcRTPHeader header; |
+ rtc::Buffer payload; |
+ double time_ms; |
+ }; |
+ |
+ virtual ~NetEqInput() = default; |
+ |
+ // Returns at what time (in ms) NetEq::InsertPacket should be called next, or |
+ // empty if the source is out of packets. |
+ virtual rtc::Optional<int64_t> NextPacketTime() const = 0; |
+ |
+ // Returns at what time (in ms) NetEq::GetAudio should be called next, or |
+ // empty if no more output events are available. |
+ virtual rtc::Optional<int64_t> NextOutputEventTime() const = 0; |
+ |
+ // Returns the time (in ms) for the next event from either NextPacketTime() |
+ // or NextOutputEventTime(), or empty if both are out of events. |
+ rtc::Optional<int64_t> NextEventTime() const { |
+ const auto a = NextPacketTime(); |
+ const auto b = NextOutputEventTime(); |
+ // Return the minimum of non-empty |a| and |b|, or empty if both are empty. |
+ if (a) { |
+ return b ? rtc::Optional<int64_t>(std::min(*a, *b)) : a; |
+ } |
+ return b ? b : rtc::Optional<int64_t>(); |
+ } |
+ |
+ // Returns the next packet to be inserted into NetEq. The packet following the |
+ // returned one is pre-fetched in the NetEqInput object, such that future |
+ // calls to NextPacketTime() or NextHeader() will return information from that |
+ // packet. |
+ virtual std::unique_ptr<PacketData> PopPacket() = 0; |
+ |
+ // Move to the next output event. This will make NextOutputEventTime() return |
+ // a new value (potentially the same if several output events share the same |
+ // time). |
+ virtual void AdvanceOutputEvent() = 0; |
+ |
+ // Returns true if the source has come to an end. |
+ virtual bool ended() const = 0; |
+ |
+ // Returns the RTP header for the next packet, i.e., the packet that will be |
+ // delivered next by PopPacket(). |
+ virtual rtc::Optional<RTPHeader> NextHeader() const = 0; |
+}; |
+ |
+} // namespace test |
+} // namespace webrtc |
+#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_ |