|  | /* | 
|  | *  Copyright (c) 2013 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. | 
|  | */ | 
|  |  | 
|  | #include "modules/audio_coding/neteq/tools/input_audio_file.h" | 
|  |  | 
|  | #include "rtc_base/checks.h" | 
|  |  | 
|  | namespace webrtc { | 
|  | namespace test { | 
|  |  | 
|  | InputAudioFile::InputAudioFile(const std::string file_name, bool loop_at_end) | 
|  | : loop_at_end_(loop_at_end) { | 
|  | fp_ = fopen(file_name.c_str(), "rb"); | 
|  | } | 
|  |  | 
|  | InputAudioFile::~InputAudioFile() { | 
|  | fclose(fp_); | 
|  | } | 
|  |  | 
|  | bool InputAudioFile::Read(size_t samples, int16_t* destination) { | 
|  | if (!fp_) { | 
|  | return false; | 
|  | } | 
|  | size_t samples_read = fread(destination, sizeof(int16_t), samples, fp_); | 
|  | if (samples_read < samples) { | 
|  | if (!loop_at_end_) { | 
|  | return false; | 
|  | } | 
|  | // Rewind and read the missing samples. | 
|  | rewind(fp_); | 
|  | size_t missing_samples = samples - samples_read; | 
|  | if (fread(destination + samples_read, sizeof(int16_t), missing_samples, | 
|  | fp_) < missing_samples) { | 
|  | // Could not read enough even after rewinding the file. | 
|  | return false; | 
|  | } | 
|  | } | 
|  | return true; | 
|  | } | 
|  |  | 
|  | bool InputAudioFile::Seek(int samples) { | 
|  | if (!fp_) { | 
|  | return false; | 
|  | } | 
|  | // Find file boundaries. | 
|  | const long current_pos = ftell(fp_); | 
|  | RTC_CHECK_NE(EOF, current_pos) | 
|  | << "Error returned when getting file position."; | 
|  | RTC_CHECK_EQ(0, fseek(fp_, 0, SEEK_END));  // Move to end of file. | 
|  | const long file_size = ftell(fp_); | 
|  | RTC_CHECK_NE(EOF, file_size) << "Error returned when getting file position."; | 
|  | // Find new position. | 
|  | long new_pos = current_pos + sizeof(int16_t) * samples;  // Samples to bytes. | 
|  | if (loop_at_end_) { | 
|  | new_pos = new_pos % file_size;  // Wrap around the end of the file. | 
|  | if (new_pos < 0) { | 
|  | // For negative values of new_pos, newpos % file_size will also be | 
|  | // negative. To get the correct result it's needed to add file_size. | 
|  | new_pos += file_size; | 
|  | } | 
|  | } else { | 
|  | new_pos = new_pos > file_size ? file_size : new_pos;  // Don't loop. | 
|  | } | 
|  | RTC_CHECK_GE(new_pos, 0) | 
|  | << "Trying to move to before the beginning of the file"; | 
|  | // Move to new position relative to the beginning of the file. | 
|  | RTC_CHECK_EQ(0, fseek(fp_, new_pos, SEEK_SET)); | 
|  | return true; | 
|  | } | 
|  |  | 
|  | void InputAudioFile::DuplicateInterleaved(const int16_t* source, | 
|  | size_t samples, | 
|  | size_t channels, | 
|  | int16_t* destination) { | 
|  | // Start from the end of |source| and |destination|, and work towards the | 
|  | // beginning. This is to allow in-place interleaving of the same array (i.e., | 
|  | // |source| and |destination| are the same array). | 
|  | for (int i = static_cast<int>(samples - 1); i >= 0; --i) { | 
|  | for (int j = static_cast<int>(channels - 1); j >= 0; --j) { | 
|  | destination[i * channels + j] = source[i]; | 
|  | } | 
|  | } | 
|  | } | 
|  |  | 
|  | }  // namespace test | 
|  | }  // namespace webrtc |