Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 93 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h b/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
new file mode 100644
index 0000000000..c5f1d7c259
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
@@ -0,0 +1,93 @@
+/*
+ *  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 MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
+#define MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
+
+#include <memory>
+#include <string>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// Define coding parameter as
+// <channels, bit_rate, file_name, extension, if_save_output>.
+typedef std::tuple<size_t, int, std::string, std::string, bool> coding_param;
+
+class AudioCodecSpeedTest : public ::testing::TestWithParam<coding_param> {
+ protected:
+  AudioCodecSpeedTest(int block_duration_ms,
+                      int input_sampling_khz,
+                      int output_sampling_khz);
+  virtual void SetUp();
+  virtual void TearDown();
+
+  // EncodeABlock(...) does the following:
+  // 1. encodes a block of audio, saved in `in_data`,
+  // 2. save the bit stream to `bit_stream` of `max_bytes` bytes in size,
+  // 3. assign `encoded_bytes` with the length of the bit stream (in bytes),
+  // 4. return the cost of time (in millisecond) spent on actual encoding.
+  virtual float EncodeABlock(int16_t* in_data,
+                             uint8_t* bit_stream,
+                             size_t max_bytes,
+                             size_t* encoded_bytes) = 0;
+
+  // DecodeABlock(...) does the following:
+  // 1. decodes the bit stream in `bit_stream` with a length of `encoded_bytes`
+  // (in bytes),
+  // 2. save the decoded audio in `out_data`,
+  // 3. return the cost of time (in millisecond) spent on actual decoding.
+  virtual float DecodeABlock(const uint8_t* bit_stream,
+                             size_t encoded_bytes,
+                             int16_t* out_data) = 0;
+
+  // Encoding and decode an audio of `audio_duration` (in seconds) and
+  // record the runtime for encoding and decoding separately.
+  void EncodeDecode(size_t audio_duration);
+
+  int block_duration_ms_;
+  int input_sampling_khz_;
+  int output_sampling_khz_;
+
+  // Number of samples-per-channel in a frame.
+  size_t input_length_sample_;
+
+  // Expected output number of samples-per-channel in a frame.
+  size_t output_length_sample_;
+
+  std::unique_ptr<int16_t[]> in_data_;
+  std::unique_ptr<int16_t[]> out_data_;
+  size_t data_pointer_;
+  size_t loop_length_samples_;
+  std::unique_ptr<uint8_t[]> bit_stream_;
+
+  // Maximum number of bytes in output bitstream for a frame of audio.
+  size_t max_bytes_;
+
+  size_t encoded_bytes_;
+  float encoding_time_ms_;
+  float decoding_time_ms_;
+  FILE* out_file_;
+
+  size_t channels_;
+
+  // Bit rate is in bit-per-second.
+  int bit_rate_;
+
+  std::string in_filename_;
+
+  // Determines whether to save the output to file.
+  bool save_out_data_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_