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In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
76 lines
2.3 KiB
C++
76 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 WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
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#define WEBRTC_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 <string>
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#include "webrtc/api/optional.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/typedefs.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(const std::string& filename, uint16_t frequency, const char* 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 {
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return end_of_file_;
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}
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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, 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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rtc::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 // WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
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