c4fe8256f3
using find modules/audio_coding -name "*.h" -o -name "*.cc" | grep -v mock_debug_dump_writer.h | xargs tools_webrtc/iwyu/apply-include-cleaner followed by tools_webrtc/gn_check_autofix.py -C out/Default/ and git cl format Manual changes: * controller_manager.h: add api/array_view.h include * RTPFile.cc / dtmf_buffer_unittest.cc: use rtc_base/ip_address.h instea of netinet/in.h * neteq_delay_analyzer.cc: add TODO for stringstream to ostream include * opus: fixup paths and make third_party/opus includes go via opus_interface.h by using IWYU export pragmas The mock_debug_dump_writer has issues with protobuf includes so was ignored for this round. BUG=webrtc:42226242 Change-Id: I5b8613053da6cc0a2a44d1d59bd2efe11e501681 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387640 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/main@{#44546}
79 lines
2.3 KiB
C++
79 lines
2.3 KiB
C++
/*
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* Copyright (c) 2012 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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#ifndef MODULES_AUDIO_CODING_TEST_PCMFILE_H_
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#define MODULES_AUDIO_CODING_TEST_PCMFILE_H_
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#include <stdio.h>
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#include <stdlib.h>
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#include <cstdint>
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#include <optional>
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#include <string>
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#include "absl/strings/string_view.h"
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#include "api/audio/audio_frame.h"
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namespace webrtc {
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class PCMFile {
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public:
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PCMFile();
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PCMFile(uint32_t timestamp);
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~PCMFile();
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void Open(absl::string_view filename,
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uint16_t frequency,
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absl::string_view mode,
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bool auto_rewind = false);
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int32_t Read10MsData(AudioFrame& audio_frame);
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void Write10MsData(const int16_t* playout_buffer, size_t length_smpls);
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void Write10MsData(const AudioFrame& audio_frame);
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uint16_t PayloadLength10Ms() const;
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int32_t SamplingFrequency() const;
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void Close();
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bool EndOfFile() const { return end_of_file_; }
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// Moves forward the specified number of 10 ms blocks. If a limit has been set
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// with SetNum10MsBlocksToRead, fast-forwarding does not count towards this
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// limit.
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void FastForward(int num_10ms_blocks);
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void Rewind();
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static int16_t ChooseFile(std::string* file_name,
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int16_t max_len,
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uint16_t* frequency_hz);
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bool Rewinded();
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void SaveStereo(bool is_stereo = true);
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void ReadStereo(bool is_stereo = true);
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// If set, the reading will stop after the specified number of blocks have
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// been read. When that has happened, EndOfFile() will return true. Calling
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// Rewind() will reset the counter and start over.
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void SetNum10MsBlocksToRead(int value);
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private:
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FILE* pcm_file_;
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uint16_t samples_10ms_;
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int32_t frequency_;
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bool end_of_file_;
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bool auto_rewind_;
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bool rewinded_;
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uint32_t timestamp_;
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bool read_stereo_;
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bool save_stereo_;
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std::optional<int> num_10ms_blocks_to_read_;
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int blocks_read_ = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_TEST_PCMFILE_H_
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