36b1a5fcec
const int16_t* data() const; int16_t* mutable_data(); - data() returns a zeroed static buffer on muted frames (to avoid unnecessary zeroing of the member buffer) and directly returns AudioFrame::data_ on unmuted frames. - mutable_data(), lazily zeroes AudioFrame::data_ if the frame is currently muted, sets muted=false, and returns AudioFrame::data_. These accessors serve to "force" callers to be aware of the mute state field, i.e. lazy zeroing is not the primary motivation. This change only optimizes handling of muted frames where it is somewhat trivial to do so. Other improvements requiring more significant structural changes will come later. BUG=webrtc:7343 TBR=henrika Review-Url: https://codereview.webrtc.org/2750783004 Cr-Commit-Position: refs/heads/master@{#18543}
77 lines
2.3 KiB
C++
77 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/base/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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