Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

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

1 files changed, 115 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/test/TestVADDTX.h b/third_party/libwebrtc/modules/audio_coding/test/TestVADDTX.h
new file mode 100644
index 0000000000..d81ae28beb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/TestVADDTX.h
@@ -0,0 +1,115 @@
+/*
+ *  Copyright (c) 2011 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_TEST_TESTVADDTX_H_
+#define MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "common_audio/vad/include/vad.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/test/Channel.h"
+
+namespace webrtc {
+
+// This class records the frame type, and delegates actual sending to the
+// `next_` AudioPacketizationCallback.
+class MonitoringAudioPacketizationCallback : public AudioPacketizationCallback {
+ public:
+  explicit MonitoringAudioPacketizationCallback(
+      AudioPacketizationCallback* next);
+
+  int32_t SendData(AudioFrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   size_t payload_len_bytes,
+                   int64_t absolute_capture_timestamp_ms) override;
+
+  void PrintStatistics();
+  void ResetStatistics();
+  void GetStatistics(uint32_t* stats);
+
+ private:
+  // 0 - kEmptyFrame
+  // 1 - kAudioFrameSpeech
+  // 2 - kAudioFrameCN
+  uint32_t counter_[3];
+  AudioPacketizationCallback* const next_;
+};
+
+// TestVadDtx is to verify that VAD/DTX perform as they should. It runs through
+// an audio file and check if the occurrence of various packet types follows
+// expectation. TestVadDtx needs its derived class to implement the Perform()
+// to put the test together.
+class TestVadDtx {
+ public:
+  static const int kOutputFreqHz = 16000;
+
+  TestVadDtx();
+
+ protected:
+  // Returns true iff CN was added.
+  bool RegisterCodec(const SdpAudioFormat& codec_format,
+                     absl::optional<Vad::Aggressiveness> vad_mode);
+
+  // Encoding a file and see if the numbers that various packets occur follow
+  // the expectation. Saves result to a file.
+  // expects[x] means
+  // -1 : do not care,
+  // 0  : there have been no packets of type `x`,
+  // 1  : there have been packets of type `x`,
+  // with `x` indicates the following packet types
+  // 0 - kEmptyFrame
+  // 1 - kAudioFrameSpeech
+  // 2 - kAudioFrameCN
+  void Run(absl::string_view in_filename,
+           int frequency,
+           int channels,
+           absl::string_view out_filename,
+           bool append,
+           const int* expects);
+
+  const rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
+  const rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+  std::unique_ptr<AudioCodingModule> acm_send_;
+  std::unique_ptr<AudioCodingModule> acm_receive_;
+  std::unique_ptr<Channel> channel_;
+  std::unique_ptr<MonitoringAudioPacketizationCallback> packetization_callback_;
+  uint32_t time_stamp_ = 0x12345678;
+};
+
+// TestWebRtcVadDtx is to verify that the WebRTC VAD/DTX perform as they should.
+class TestWebRtcVadDtx final : public TestVadDtx {
+ public:
+  TestWebRtcVadDtx();
+
+  void Perform();
+
+ private:
+  void RunTestCases(const SdpAudioFormat& codec_format);
+  void Test(bool new_outfile, bool expect_dtx_enabled);
+
+  int output_file_num_;
+};
+
+// TestOpusDtx is to verify that the Opus DTX performs as it should.
+class TestOpusDtx final : public TestVadDtx {
+ public:
+  void Perform();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_