Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 135 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_input.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_input.h
new file mode 100644
index 0000000000..56b0212add
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_input.h
@@ -0,0 +1,135 @@
+/*
+ *  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 MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
+
+#include <algorithm>
+#include <memory>
+#include <string>
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+namespace test {
+
+// Interface class for input to the NetEqTest class.
+class NetEqInput {
+ public:
+  struct PacketData {
+    PacketData();
+    ~PacketData();
+    std::string ToString() const;
+
+    RTPHeader header;
+    rtc::Buffer payload;
+    int64_t time_ms;
+  };
+
+  struct SetMinimumDelayInfo {
+    SetMinimumDelayInfo(int64_t timestamp_ms_in, int delay_ms_in)
+        : timestamp_ms(timestamp_ms_in), delay_ms(delay_ms_in) {}
+    int64_t timestamp_ms;
+    int delay_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 absl::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 absl::optional<int64_t> NextOutputEventTime() const = 0;
+
+  // Returns the information related to the next NetEq set minimum delay event
+  // if available.
+  virtual absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo()
+      const = 0;
+
+  // Returns the time (in ms) for the next event (packet, output or set minimum
+  // delay event) or empty if there are no more events.
+  absl::optional<int64_t> NextEventTime() const {
+    absl::optional<int64_t> next_event_time = NextPacketTime();
+    const auto next_output_time = NextOutputEventTime();
+    // Return the minimum of non-empty `a` and `b`, or empty if both are empty.
+    if (next_output_time) {
+      next_event_time = next_event_time ? std::min(next_event_time.value(),
+                                                   next_output_time.value())
+                                        : next_output_time;
+    }
+    const auto next_neteq_minimum_delay = NextSetMinimumDelayInfo();
+    if (next_neteq_minimum_delay) {
+      next_event_time =
+          next_event_time
+              ? std::min(next_event_time.value(),
+                         next_neteq_minimum_delay.value().timestamp_ms)
+              : next_neteq_minimum_delay.value().timestamp_ms;
+    }
+    return next_event_time;
+  }
+
+  // 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;
+
+  // Move to the next NetEq set minimum delay. This will make
+  // `NextSetMinimumDelayInfo` return a new value.
+  virtual void AdvanceSetMinimumDelay() = 0;
+
+  // Returns true if the source has come to an end. An implementation must
+  // eventually return true from this method, or the test will end up in an
+  // infinite loop.
+  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 absl::optional<RTPHeader> NextHeader() const = 0;
+};
+
+// Wrapper class to impose a time limit on a NetEqInput object, typically
+// another time limit than what the object itself provides. For example, an
+// input taken from a file can be cut shorter by wrapping it in this class.
+class TimeLimitedNetEqInput : public NetEqInput {
+ public:
+  TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input, int64_t duration_ms);
+  ~TimeLimitedNetEqInput() override;
+  absl::optional<int64_t> NextPacketTime() const override;
+  absl::optional<int64_t> NextOutputEventTime() const override;
+  absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override;
+  std::unique_ptr<PacketData> PopPacket() override;
+  void AdvanceOutputEvent() override;
+  void AdvanceSetMinimumDelay() override;
+  bool ended() const override;
+  absl::optional<RTPHeader> NextHeader() const override;
+
+ private:
+  void MaybeSetEnded();
+
+  std::unique_ptr<NetEqInput> input_;
+  const absl::optional<int64_t> start_time_ms_;
+  const int64_t duration_ms_;
+  bool ended_ = false;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_