replicant-frameworks_native/libs/audioflinger/AudioHardwareStub.cpp

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/* //device/servers/AudioFlinger/AudioHardwareStub.cpp
**
** Copyright 2007, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
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**
** http://www.apache.org/licenses/LICENSE-2.0
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**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
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** limitations under the License.
*/
#include <stdint.h>
#include <sys/types.h>
#include <stdlib.h>
#include <unistd.h>
#include <utils/String8.h>
#include "AudioHardwareStub.h"
namespace android {
// ----------------------------------------------------------------------------
AudioHardwareStub::AudioHardwareStub() : mMicMute(false)
{
}
AudioHardwareStub::~AudioHardwareStub()
{
}
status_t AudioHardwareStub::initCheck()
{
return NO_ERROR;
}
AudioStreamOut* AudioHardwareStub::openOutputStream(
int format, int channelCount, uint32_t sampleRate, status_t *status)
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{
AudioStreamOutStub* out = new AudioStreamOutStub();
status_t lStatus = out->set(format, channelCount, sampleRate);
if (status) {
*status = lStatus;
}
if (lStatus == NO_ERROR)
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return out;
delete out;
return 0;
}
AudioStreamIn* AudioHardwareStub::openInputStream(
int format, int channelCount, uint32_t sampleRate,
status_t *status, AudioSystem::audio_in_acoustics acoustics)
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{
AudioStreamInStub* in = new AudioStreamInStub();
status_t lStatus = in->set(format, channelCount, sampleRate, acoustics);
if (status) {
*status = lStatus;
}
if (lStatus == NO_ERROR)
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return in;
delete in;
return 0;
}
status_t AudioHardwareStub::setVoiceVolume(float volume)
{
return NO_ERROR;
}
status_t AudioHardwareStub::setMasterVolume(float volume)
{
return NO_ERROR;
}
status_t AudioHardwareStub::dumpInternals(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
result.append("AudioHardwareStub::dumpInternals\n");
snprintf(buffer, SIZE, "\tmMicMute: %s\n", mMicMute? "true": "false");
result.append(buffer);
::write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioHardwareStub::dump(int fd, const Vector<String16>& args)
{
dumpInternals(fd, args);
return NO_ERROR;
}
// ----------------------------------------------------------------------------
status_t AudioStreamOutStub::set(int format, int channels, uint32_t rate)
{
// fix up defaults
if (format == 0) format = AudioSystem::PCM_16_BIT;
if (channels == 0) channels = channelCount();
if (rate == 0) rate = sampleRate();
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if ((format == AudioSystem::PCM_16_BIT) &&
(channels == channelCount()) &&
(rate == sampleRate()))
return NO_ERROR;
return BAD_VALUE;
}
ssize_t AudioStreamOutStub::write(const void* buffer, size_t bytes)
{
// fake timing for audio output
usleep(bytes * 1000000 / sizeof(int16_t) / channelCount() / sampleRate());
return bytes;
}
status_t AudioStreamOutStub::standby()
{
return NO_ERROR;
}
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status_t AudioStreamOutStub::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "AudioStreamOutStub::dump\n");
snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
snprintf(buffer, SIZE, "\tchannel count: %d\n", channelCount());
snprintf(buffer, SIZE, "\tformat: %d\n", format());
result.append(buffer);
::write(fd, result.string(), result.size());
return NO_ERROR;
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}
// ----------------------------------------------------------------------------
status_t AudioStreamInStub::set(int format, int channels, uint32_t rate,
AudioSystem::audio_in_acoustics acoustics)
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{
if ((format == AudioSystem::PCM_16_BIT) &&
(channels == channelCount()) &&
(rate == sampleRate()))
return NO_ERROR;
return BAD_VALUE;
}
ssize_t AudioStreamInStub::read(void* buffer, ssize_t bytes)
{
// fake timing for audio input
usleep(bytes * 1000000 / sizeof(int16_t) / channelCount() / sampleRate());
memset(buffer, 0, bytes);
return bytes;
}
status_t AudioStreamInStub::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "AudioStreamInStub::dump\n");
result.append(buffer);
snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
result.append(buffer);
snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
result.append(buffer);
snprintf(buffer, SIZE, "\tchannel count: %d\n", channelCount());
result.append(buffer);
snprintf(buffer, SIZE, "\tformat: %d\n", format());
result.append(buffer);
::write(fd, result.string(), result.size());
return NO_ERROR;
}
// ----------------------------------------------------------------------------
}; // namespace android