Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

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

1 files changed, 116 insertions, 0 deletions

diff --git a/third_party/libwebrtc/test/mock_audio_encoder.h b/third_party/libwebrtc/test/mock_audio_encoder.h
new file mode 100644
index 0000000000..1f4510e885
--- /dev/null
+++ b/third_party/libwebrtc/test/mock_audio_encoder.h
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2014 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 TEST_MOCK_AUDIO_ENCODER_H_
+#define TEST_MOCK_AUDIO_ENCODER_H_
+
+#include <string>
+
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockAudioEncoder : public AudioEncoder {
+ public:
+  MockAudioEncoder();
+  ~MockAudioEncoder();
+  MOCK_METHOD(int, SampleRateHz, (), (const, override));
+  MOCK_METHOD(size_t, NumChannels, (), (const, override));
+  MOCK_METHOD(int, RtpTimestampRateHz, (), (const, override));
+  MOCK_METHOD(size_t, Num10MsFramesInNextPacket, (), (const, override));
+  MOCK_METHOD(size_t, Max10MsFramesInAPacket, (), (const, override));
+  MOCK_METHOD(int, GetTargetBitrate, (), (const, override));
+  MOCK_METHOD((absl::optional<std::pair<TimeDelta, TimeDelta>>),
+              GetFrameLengthRange,
+              (),
+              (const, override));
+
+  MOCK_METHOD(void, Reset, (), (override));
+  MOCK_METHOD(bool, SetFec, (bool enable), (override));
+  MOCK_METHOD(bool, SetDtx, (bool enable), (override));
+  MOCK_METHOD(bool, SetApplication, (Application application), (override));
+  MOCK_METHOD(void, SetMaxPlaybackRate, (int frequency_hz), (override));
+  MOCK_METHOD(void,
+              OnReceivedUplinkBandwidth,
+              (int target_audio_bitrate_bps,
+               absl::optional<int64_t> probing_interval_ms),
+              (override));
+  MOCK_METHOD(void,
+              OnReceivedUplinkPacketLossFraction,
+              (float uplink_packet_loss_fraction),
+              (override));
+  MOCK_METHOD(void,
+              OnReceivedOverhead,
+              (size_t overhead_bytes_per_packet),
+              (override));
+
+  MOCK_METHOD(bool,
+              EnableAudioNetworkAdaptor,
+              (const std::string& config_string, RtcEventLog*),
+              (override));
+
+  // Note, we explicitly chose not to create a mock for the Encode method.
+  MOCK_METHOD(EncodedInfo,
+              EncodeImpl,
+              (uint32_t timestamp,
+               rtc::ArrayView<const int16_t> audio,
+               rtc::Buffer*),
+              (override));
+
+  class FakeEncoding {
+   public:
+    // Creates a functor that will return `info` and adjust the rtc::Buffer
+    // given as input to it, so it is info.encoded_bytes larger.
+    explicit FakeEncoding(const AudioEncoder::EncodedInfo& info);
+
+    // Shorthand version of the constructor above, for when only setting
+    // encoded_bytes in the EncodedInfo object matters.
+    explicit FakeEncoding(size_t encoded_bytes);
+
+    AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
+                                         rtc::ArrayView<const int16_t> audio,
+                                         rtc::Buffer* encoded);
+
+   private:
+    AudioEncoder::EncodedInfo info_;
+  };
+
+  class CopyEncoding {
+   public:
+    ~CopyEncoding();
+
+    // Creates a functor that will return `info` and append the data in the
+    // payload to the buffer given as input to it. Up to info.encoded_bytes are
+    // appended - make sure the payload is big enough!  Since it uses an
+    // ArrayView, it _does not_ copy the payload. Make sure it doesn't fall out
+    // of scope!
+    CopyEncoding(AudioEncoder::EncodedInfo info,
+                 rtc::ArrayView<const uint8_t> payload);
+
+    // Shorthand version of the constructor above, for when you wish to append
+    // the whole payload and do not care about any EncodedInfo attribute other
+    // than encoded_bytes.
+    explicit CopyEncoding(rtc::ArrayView<const uint8_t> payload);
+
+    AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
+                                         rtc::ArrayView<const int16_t> audio,
+                                         rtc::Buffer* encoded);
+
+   private:
+    AudioEncoder::EncodedInfo info_;
+    rtc::ArrayView<const uint8_t> payload_;
+  };
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_MOCK_AUDIO_ENCODER_H_