1 /* 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_ 12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_ 13 14 #include <string> 15 #include "testing/gtest/include/gtest/gtest.h" 16 #include "webrtc/system_wrappers/interface/scoped_ptr.h" 17 #include "webrtc/typedefs.h" 18 19 namespace webrtc { 20 21 // Define coding parameter as 22 // <channels, bit_rate, file_name, extension, if_save_output>. 23 typedef std::tr1::tuple<int, int, std::string, std::string, bool> coding_param; 24 25 class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> { 26 protected: 27 AudioCodecSpeedTest(int block_duration_ms, 28 int input_sampling_khz, 29 int output_sampling_khz); 30 virtual void SetUp(); 31 virtual void TearDown(); 32 33 // EncodeABlock(...) does the following: 34 // 1. encodes a block of audio, saved in |in_data|, 35 // 2. save the bit stream to |bit_stream| of |max_bytes| bytes in size, 36 // 3. assign |encoded_bytes| with the length of the bit stream (in bytes), 37 // 4. return the cost of time (in millisecond) spent on actual encoding. 38 virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream, 39 int max_bytes, int* encoded_bytes) = 0; 40 41 // DecodeABlock(...) does the following: 42 // 1. decodes the bit stream in |bit_stream| with a length of |encoded_bytes| 43 // (in bytes), 44 // 2. save the decoded audio in |out_data|, 45 // 3. return the cost of time (in millisecond) spent on actual decoding. 46 virtual float DecodeABlock(const uint8_t* bit_stream, int encoded_bytes, 47 int16_t* out_data) = 0; 48 49 // Encoding and decode an audio of |audio_duration| (in seconds) and 50 // record the runtime for encoding and decoding separately. 51 void EncodeDecode(size_t audio_duration); 52 53 int block_duration_ms_; 54 int input_sampling_khz_; 55 int output_sampling_khz_; 56 57 // Number of samples-per-channel in a frame. 58 int input_length_sample_; 59 60 // Expected output number of samples-per-channel in a frame. 61 int output_length_sample_; 62 63 scoped_ptr<int16_t[]> in_data_; 64 scoped_ptr<int16_t[]> out_data_; 65 size_t data_pointer_; 66 size_t loop_length_samples_; 67 scoped_ptr<uint8_t[]> bit_stream_; 68 69 // Maximum number of bytes in output bitstream for a frame of audio. 70 int max_bytes_; 71 72 int encoded_bytes_; 73 float encoding_time_ms_; 74 float decoding_time_ms_; 75 FILE* out_file_; 76 77 int channels_; 78 79 // Bit rate is in bit-per-second. 80 int bit_rate_; 81 82 std::string in_filename_; 83 84 // Determines whether to save the output to file. 85 bool save_out_data_; 86 }; 87 88 } // namespace webrtc 89 90 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_ 91