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Removes all uses of FileWrapper in audio_device.

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BUG=webrtc:6463

Review-Url: https://codereview.webrtc.org/2386963003
Cr-Commit-Position: refs/heads/master@{#14811}
This commit is contained in:
palmkvist 2016-10-28 02:08:23 -07:00 committed by Commit bot
parent 8b8d3e4c30
commit 04055e95bf
3 changed files with 88 additions and 96 deletions
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3
webrtc/modules/audio_device/audio_device_buffer.h
View File

@ -16,7 +16,6 @@
#include "webrtc/base/task_queue.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -33,7 +32,7 @@ class AudioDeviceBuffer {
AudioDeviceBuffer();
virtual ~AudioDeviceBuffer();
void SetId(uint32_t id) {};
void SetId(uint32_t id) {}
int32_t RegisterAudioCallback(AudioTransport* audio_callback);
int32_t InitPlayout();

168
webrtc/modules/audio_device/dummy/file_audio_device.cc
View File

@ -39,18 +39,11 @@ FileAudioDevice::FileAudioDevice(const int32_t id,
_recording(false),
_lastCallPlayoutMillis(0),
_lastCallRecordMillis(0),
_outputFile(*FileWrapper::Create()),
_inputFile(*FileWrapper::Create()),
_outputFilename(outputFilename),
_inputFilename(inputFilename),
_clock(Clock::GetRealTimeClock()) {
}
FileAudioDevice::~FileAudioDevice() {
delete &_outputFile;
delete &_inputFile;
}
int32_t FileAudioDevice::ActiveAudioLayer(
AudioDeviceModule::AudioLayer& audioLayer) const {
return -1;
@ -197,13 +190,15 @@ int32_t FileAudioDevice::StartPlayout() {
}
// PLAYOUT
if (!_outputFilename.empty() &&
!_outputFile.OpenFile(_outputFilename.c_str(), false)) {
LOG(LS_ERROR) << "Failed to open playout file: " << _outputFilename;
_playing = false;
delete [] _playoutBuffer;
_playoutBuffer = NULL;
return -1;
if (!_outputFilename.empty()) {
_outputFile = rtc::File::Open(_outputFilename);
if (!_outputFile.IsOpen()) {
LOG(LS_ERROR) << "Failed to open playout file: " << _outputFilename;
_playing = false;
delete[] _playoutBuffer;
_playoutBuffer = NULL;
return -1;
}
}
_ptrThreadPlay.reset(new rtc::PlatformThread(
@ -233,7 +228,7 @@ int32_t FileAudioDevice::StopPlayout() {
_playoutFramesLeft = 0;
delete [] _playoutBuffer;
_playoutBuffer = NULL;
_outputFile.CloseFile();
_outputFile.Close();
LOG(LS_INFO) << "Stopped playout capture to output file: "
<< _outputFilename;
@ -255,13 +250,15 @@ int32_t FileAudioDevice::StartRecording() {
_recordingBuffer = new int8_t[_recordingBufferSizeIn10MS];
}
if (!_inputFilename.empty() &&
!_inputFile.OpenFile(_inputFilename.c_str(), true)) {
LOG(LS_ERROR) << "Failed to open audio input file: " << _inputFilename;
_recording = false;
delete[] _recordingBuffer;
_recordingBuffer = NULL;
return -1;
if (!_inputFilename.empty()) {
_inputFile = rtc::File::Open(_inputFilename);
if (!_inputFile.IsOpen()) {
LOG(LS_ERROR) << "Failed to open audio input file: " << _inputFilename;
_recording = false;
delete[] _recordingBuffer;
_recordingBuffer = NULL;
return -1;
}
}
_ptrThreadRec.reset(new rtc::PlatformThread(
@ -294,7 +291,7 @@ int32_t FileAudioDevice::StopRecording() {
delete [] _recordingBuffer;
_recordingBuffer = NULL;
}
_inputFile.CloseFile();
_inputFile.Close();
LOG(LS_INFO) << "Stopped recording from input file: "
<< _inputFilename;
@ -310,12 +307,12 @@ int32_t FileAudioDevice::SetAGC(bool enable) { return -1; }
bool FileAudioDevice::AGC() const { return false; }
int32_t FileAudioDevice::SetWaveOutVolume(uint16_t volumeLeft,
uint16_t volumeRight) {
uint16_t volumeRight) {
return -1;
}
int32_t FileAudioDevice::WaveOutVolume(uint16_t& volumeLeft,
uint16_t& volumeRight) const {
uint16_t& volumeRight) const {
return -1;
}
@ -468,81 +465,78 @@ void FileAudioDevice::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
_ptrAudioBuffer->SetPlayoutChannels(0);
}
bool FileAudioDevice::PlayThreadFunc(void* pThis)
{
return (static_cast<FileAudioDevice*>(pThis)->PlayThreadProcess());
bool FileAudioDevice::PlayThreadFunc(void* pThis) {
return (static_cast<FileAudioDevice*>(pThis)->PlayThreadProcess());
}
bool FileAudioDevice::RecThreadFunc(void* pThis)
{
return (static_cast<FileAudioDevice*>(pThis)->RecThreadProcess());
bool FileAudioDevice::RecThreadFunc(void* pThis) {
return (static_cast<FileAudioDevice*>(pThis)->RecThreadProcess());
}
bool FileAudioDevice::PlayThreadProcess()
{
if(!_playing) {
return false;
}
uint64_t currentTime = _clock->CurrentNtpInMilliseconds();
_critSect.Enter();
bool FileAudioDevice::PlayThreadProcess() {
if (!_playing) {
return false;
}
uint64_t currentTime = _clock->CurrentNtpInMilliseconds();
_critSect.Enter();
if (_lastCallPlayoutMillis == 0 ||
currentTime - _lastCallPlayoutMillis >= 10) {
_critSect.Leave();
_ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
_critSect.Enter();
_playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
assert(_playoutFramesLeft == _playoutFramesIn10MS);
if (_outputFile.is_open()) {
_outputFile.Write(_playoutBuffer, kPlayoutBufferSize);
}
_lastCallPlayoutMillis = currentTime;
}
_playoutFramesLeft = 0;
if (_lastCallPlayoutMillis == 0 ||
currentTime - _lastCallPlayoutMillis >= 10) {
_critSect.Leave();
uint64_t deltaTimeMillis = _clock->CurrentNtpInMilliseconds() - currentTime;
if(deltaTimeMillis < 10) {
SleepMs(10 - deltaTimeMillis);
}
return true;
}
bool FileAudioDevice::RecThreadProcess()
{
if (!_recording) {
return false;
}
uint64_t currentTime = _clock->CurrentNtpInMilliseconds();
_ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
_critSect.Enter();
if (_lastCallRecordMillis == 0 ||
currentTime - _lastCallRecordMillis >= 10) {
if (_inputFile.is_open()) {
if (_inputFile.Read(_recordingBuffer, kRecordingBufferSize) > 0) {
_ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
_recordingFramesIn10MS);
} else {
_inputFile.Rewind();
}
_lastCallRecordMillis = currentTime;
_critSect.Leave();
_ptrAudioBuffer->DeliverRecordedData();
_critSect.Enter();
_playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
assert(_playoutFramesLeft == _playoutFramesIn10MS);
if (_outputFile.IsOpen()) {
_outputFile.Write(reinterpret_cast<uint8_t*>(_playoutBuffer),
kPlayoutBufferSize);
}
_lastCallPlayoutMillis = currentTime;
}
_playoutFramesLeft = 0;
_critSect.Leave();
uint64_t deltaTimeMillis = _clock->CurrentNtpInMilliseconds() - currentTime;
if (deltaTimeMillis < 10) {
SleepMs(10 - deltaTimeMillis);
}
return true;
}
bool FileAudioDevice::RecThreadProcess() {
if (!_recording) {
return false;
}
uint64_t currentTime = _clock->CurrentNtpInMilliseconds();
_critSect.Enter();
if (_lastCallRecordMillis == 0 || currentTime - _lastCallRecordMillis >= 10) {
if (_inputFile.IsOpen()) {
if (_inputFile.Read(reinterpret_cast<uint8_t*>(_recordingBuffer),
kRecordingBufferSize) > 0) {
_ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
_recordingFramesIn10MS);
} else {
_inputFile.Seek(0);
}
_lastCallRecordMillis = currentTime;
_critSect.Leave();
_ptrAudioBuffer->DeliverRecordedData();
_critSect.Enter();
}
}
_critSect.Leave();
_critSect.Leave();
uint64_t deltaTimeMillis = _clock->CurrentNtpInMilliseconds() - currentTime;
if(deltaTimeMillis < 10) {
SleepMs(10 - deltaTimeMillis);
}
uint64_t deltaTimeMillis = _clock->CurrentNtpInMilliseconds() - currentTime;
if (deltaTimeMillis < 10) {
SleepMs(10 - deltaTimeMillis);
}
return true;
return true;
}
} // namespace webrtc

13
webrtc/modules/audio_device/dummy/file_audio_device.h
View File

@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H
#define WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H
#ifndef WEBRTC_MODULES_AUDIO_DEVICE_DUMMY_FILE_AUDIO_DEVICE_H_
#define WEBRTC_MODULES_AUDIO_DEVICE_DUMMY_FILE_AUDIO_DEVICE_H_
#include <stdio.h>
#include <memory>
#include <string>
#include "webrtc/base/file.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/system_wrappers/include/clock.h"
namespace rtc {
@ -41,7 +41,6 @@ class FileAudioDevice : public AudioDeviceGeneric {
FileAudioDevice(const int32_t id,
const char* inputFilename,
const char* outputFilename);
virtual ~FileAudioDevice();
// Retrieve the currently utilized audio layer
int32_t ActiveAudioLayer(
@ -191,8 +190,8 @@ class FileAudioDevice : public AudioDeviceGeneric {
uint64_t _lastCallPlayoutMillis;
uint64_t _lastCallRecordMillis;
FileWrapper& _outputFile;
FileWrapper& _inputFile;
rtc::File _outputFile;
rtc::File _inputFile;
std::string _outputFilename;
std::string _inputFilename;
@ -201,4 +200,4 @@ class FileAudioDevice : public AudioDeviceGeneric {
} // namespace webrtc
#endif // WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H
#endif // WEBRTC_MODULES_AUDIO_DEVICE_DUMMY_FILE_AUDIO_DEVICE_H_