ad5037b4a8
This reverts commit 793bac569fdf1be16cbf24d7871d20d00bbec81b. Reason for revert: rare compilation error fixed Original change's description: > Revert "Refactor the PlatformThread API." > > This reverts commit c89fdd716c4c8af608017c76f75bf27e4c3d602e. > > Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot. > See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745? > > Original change's description: > > Refactor the PlatformThread API. > > > > PlatformThread's API is using old style function pointers, causes > > casting, is unintuitive and forces artificial call sequences, and > > is additionally possible to misuse in release mode. > > > > Fix this by an API face lift: > > 1. The class is turned into a handle, which can be empty. > > 2. The only way of getting a non-empty PlatformThread is by calling > > SpawnJoinable or SpawnDetached, clearly conveying the semantics to the > > code reader. > > 3. Handles can be Finalized, which works differently for joinable and > > detached threads: > > a) Handles for detached threads are simply closed where applicable. > > b) Joinable threads are joined before handles are closed. > > 4. The destructor finalizes handles. No explicit call is needed. > > > > Fixed: webrtc:12727 > > Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075 > > Commit-Queue: Markus Handell <handellm@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Tommi <tommi@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33923} > > # Not skipping CQ checks because original CL landed > 1 day ago. > > TBR=handellm@webrtc.org > > Bug: webrtc:12727 > Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583 > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Reviewed-by: Markus Handell <handellm@webrtc.org> > Commit-Queue: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33936} # Not skipping CQ checks because this is a reland. Bug: webrtc:12727 Change-Id: Ifd6f44eac72fed84474277a1be03eb84d2f4376e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217881 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33950}
162 lines
5.5 KiB
C++
162 lines
5.5 KiB
C++
/*
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* Copyright (c) 2014 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 AUDIO_DEVICE_FILE_AUDIO_DEVICE_H_
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#define AUDIO_DEVICE_FILE_AUDIO_DEVICE_H_
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#include <stdio.h>
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#include <memory>
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#include <string>
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#include "modules/audio_device/audio_device_generic.h"
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#include "rtc_base/platform_thread.h"
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#include "rtc_base/synchronization/mutex.h"
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#include "rtc_base/system/file_wrapper.h"
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#include "rtc_base/time_utils.h"
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namespace webrtc {
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// This is a fake audio device which plays audio from a file as its microphone
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// and plays out into a file.
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class FileAudioDevice : public AudioDeviceGeneric {
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public:
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// Constructs a file audio device with |id|. It will read audio from
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// |inputFilename| and record output audio to |outputFilename|.
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//
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// The input file should be a readable 48k stereo raw file, and the output
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// file should point to a writable location. The output format will also be
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// 48k stereo raw audio.
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FileAudioDevice(const char* inputFilename, const char* outputFilename);
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virtual ~FileAudioDevice();
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// Retrieve the currently utilized audio layer
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int32_t ActiveAudioLayer(
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AudioDeviceModule::AudioLayer& audioLayer) const override;
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// Main initializaton and termination
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InitStatus Init() override;
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int32_t Terminate() override;
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bool Initialized() const override;
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// Device enumeration
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int16_t PlayoutDevices() override;
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int16_t RecordingDevices() override;
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int32_t PlayoutDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]) override;
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int32_t RecordingDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]) override;
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// Device selection
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int32_t SetPlayoutDevice(uint16_t index) override;
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int32_t SetPlayoutDevice(
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AudioDeviceModule::WindowsDeviceType device) override;
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int32_t SetRecordingDevice(uint16_t index) override;
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int32_t SetRecordingDevice(
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AudioDeviceModule::WindowsDeviceType device) override;
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// Audio transport initialization
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int32_t PlayoutIsAvailable(bool& available) override;
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int32_t InitPlayout() override;
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bool PlayoutIsInitialized() const override;
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int32_t RecordingIsAvailable(bool& available) override;
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int32_t InitRecording() override;
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bool RecordingIsInitialized() const override;
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// Audio transport control
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int32_t StartPlayout() override;
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int32_t StopPlayout() override;
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bool Playing() const override;
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int32_t StartRecording() override;
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int32_t StopRecording() override;
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bool Recording() const override;
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// Audio mixer initialization
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int32_t InitSpeaker() override;
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bool SpeakerIsInitialized() const override;
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int32_t InitMicrophone() override;
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bool MicrophoneIsInitialized() const override;
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// Speaker volume controls
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int32_t SpeakerVolumeIsAvailable(bool& available) override;
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int32_t SetSpeakerVolume(uint32_t volume) override;
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int32_t SpeakerVolume(uint32_t& volume) const override;
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int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
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int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
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// Microphone volume controls
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int32_t MicrophoneVolumeIsAvailable(bool& available) override;
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int32_t SetMicrophoneVolume(uint32_t volume) override;
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int32_t MicrophoneVolume(uint32_t& volume) const override;
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int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
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int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
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// Speaker mute control
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int32_t SpeakerMuteIsAvailable(bool& available) override;
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int32_t SetSpeakerMute(bool enable) override;
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int32_t SpeakerMute(bool& enabled) const override;
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// Microphone mute control
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int32_t MicrophoneMuteIsAvailable(bool& available) override;
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int32_t SetMicrophoneMute(bool enable) override;
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int32_t MicrophoneMute(bool& enabled) const override;
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// Stereo support
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int32_t StereoPlayoutIsAvailable(bool& available) override;
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int32_t SetStereoPlayout(bool enable) override;
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int32_t StereoPlayout(bool& enabled) const override;
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int32_t StereoRecordingIsAvailable(bool& available) override;
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int32_t SetStereoRecording(bool enable) override;
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int32_t StereoRecording(bool& enabled) const override;
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// Delay information and control
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int32_t PlayoutDelay(uint16_t& delayMS) const override;
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void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
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private:
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static void RecThreadFunc(void*);
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static void PlayThreadFunc(void*);
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bool RecThreadProcess();
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bool PlayThreadProcess();
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int32_t _playout_index;
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int32_t _record_index;
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AudioDeviceBuffer* _ptrAudioBuffer;
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int8_t* _recordingBuffer; // In bytes.
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int8_t* _playoutBuffer; // In bytes.
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uint32_t _recordingFramesLeft;
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uint32_t _playoutFramesLeft;
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Mutex mutex_;
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size_t _recordingBufferSizeIn10MS;
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size_t _recordingFramesIn10MS;
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size_t _playoutFramesIn10MS;
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rtc::PlatformThread _ptrThreadRec;
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rtc::PlatformThread _ptrThreadPlay;
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bool _playing;
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bool _recording;
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int64_t _lastCallPlayoutMillis;
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int64_t _lastCallRecordMillis;
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FileWrapper _outputFile;
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FileWrapper _inputFile;
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std::string _outputFilename;
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std::string _inputFilename;
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};
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} // namespace webrtc
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#endif // AUDIO_DEVICE_FILE_AUDIO_DEVICE_H_
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