8b3db59b6e
This reverts commit 5341aaccdb64e3336abf5875e8828222446adffa. Reason for revert: Order of initialization of global static strings. Original change's description: > Reland of https://webrtc-review.googlesource.com/c/src/+/114883 > > The difference to the original is new bitexactness strings AND > global static file string constants. The reason for reland is breaking > downstream projects. > > Original CL description: > > Tests for multi-stream Opus. > > This CL (mainly) adds bit-exactness tests for multi-stream Opus. The > tests are in audio_coding_unittest.cc. Some refactoring of > AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it > possible. A few checks for "channels \in {1, 2}" are replaced with > "channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few > other changes are made to be able to write and read multi-channel WAV > files. > > The SDP changes are NOT included; as of this CL there is no way to set > up a multi-stream opus en/de-coder from SDP strings. > > Bug: webrtc:8649 > Change-Id: I9fd47c790c241c1876c4a731b0840bec30b4f1b2 > Reviewed-on: https://webrtc-review.googlesource.com/c/123387 > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26774} TBR=aleloi@webrtc.org,ossu@webrtc.org Change-Id: I88060f2050ccee83d6091b042a10f79b3c4edc47 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8649 Reviewed-on: https://webrtc-review.googlesource.com/c/123580 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26777}
94 lines
3.1 KiB
C++
94 lines
3.1 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_coding/neteq/tools/input_audio_file.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace test {
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InputAudioFile::InputAudioFile(const std::string file_name, bool loop_at_end)
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: loop_at_end_(loop_at_end) {
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fp_ = fopen(file_name.c_str(), "rb");
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}
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InputAudioFile::~InputAudioFile() {
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fclose(fp_);
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}
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bool InputAudioFile::Read(size_t samples, int16_t* destination) {
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if (!fp_) {
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return false;
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}
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size_t samples_read = fread(destination, sizeof(int16_t), samples, fp_);
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if (samples_read < samples) {
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if (!loop_at_end_) {
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return false;
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}
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// Rewind and read the missing samples.
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rewind(fp_);
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size_t missing_samples = samples - samples_read;
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if (fread(destination + samples_read, sizeof(int16_t), missing_samples,
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fp_) < missing_samples) {
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// Could not read enough even after rewinding the file.
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return false;
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}
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}
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return true;
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}
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bool InputAudioFile::Seek(int samples) {
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if (!fp_) {
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return false;
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}
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// Find file boundaries.
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const long current_pos = ftell(fp_);
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RTC_CHECK_NE(EOF, current_pos)
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<< "Error returned when getting file position.";
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RTC_CHECK_EQ(0, fseek(fp_, 0, SEEK_END)); // Move to end of file.
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const long file_size = ftell(fp_);
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RTC_CHECK_NE(EOF, file_size) << "Error returned when getting file position.";
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// Find new position.
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long new_pos = current_pos + sizeof(int16_t) * samples; // Samples to bytes.
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if (loop_at_end_) {
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new_pos = new_pos % file_size; // Wrap around the end of the file.
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if (new_pos < 0) {
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// For negative values of new_pos, newpos % file_size will also be
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// negative. To get the correct result it's needed to add file_size.
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new_pos += file_size;
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}
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} else {
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new_pos = new_pos > file_size ? file_size : new_pos; // Don't loop.
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}
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RTC_CHECK_GE(new_pos, 0)
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<< "Trying to move to before the beginning of the file";
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// Move to new position relative to the beginning of the file.
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RTC_CHECK_EQ(0, fseek(fp_, new_pos, SEEK_SET));
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return true;
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}
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void InputAudioFile::DuplicateInterleaved(const int16_t* source,
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size_t samples,
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size_t channels,
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int16_t* destination) {
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// Start from the end of |source| and |destination|, and work towards the
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// beginning. This is to allow in-place interleaving of the same array (i.e.,
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// |source| and |destination| are the same array).
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for (int i = static_cast<int>(samples - 1); i >= 0; --i) {
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for (int j = static_cast<int>(channels - 1); j >= 0; --j) {
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destination[i * channels + j] = source[i];
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}
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}
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}
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} // namespace test
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} // namespace webrtc
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