2009-03-04 03:31:44 +00:00
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/* //device/servers/AudioFlinger/AudioHardwareStub.cpp
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**
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** Copyright 2007, The Android Open Source Project
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**
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** Licensed under the Apache License, Version 2.0 (the "License");
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** you may not use this file except in compliance with the License.
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** You may obtain a copy of the License at
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**
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** http://www.apache.org/licenses/LICENSE-2.0
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**
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** Unless required by applicable law or agreed to in writing, software
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** distributed under the License is distributed on an "AS IS" BASIS,
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** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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** See the License for the specific language governing permissions and
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** limitations under the License.
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*/
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#include <stdint.h>
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#include <sys/types.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <utils/String8.h>
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#include "AudioHardwareStub.h"
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2009-07-21 18:16:54 +00:00
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#include <media/AudioRecord.h>
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2009-03-04 03:31:44 +00:00
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namespace android {
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// ----------------------------------------------------------------------------
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AudioHardwareStub::AudioHardwareStub() : mMicMute(false)
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{
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}
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AudioHardwareStub::~AudioHardwareStub()
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{
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}
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status_t AudioHardwareStub::initCheck()
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{
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return NO_ERROR;
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}
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AudioStreamOut* AudioHardwareStub::openOutputStream(
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int format, int channelCount, uint32_t sampleRate, status_t *status)
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{
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AudioStreamOutStub* out = new AudioStreamOutStub();
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status_t lStatus = out->set(format, channelCount, sampleRate);
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if (status) {
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*status = lStatus;
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}
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if (lStatus == NO_ERROR)
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return out;
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delete out;
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return 0;
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}
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AudioStreamIn* AudioHardwareStub::openInputStream(
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2009-07-21 18:16:54 +00:00
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int inputSource, int format, int channelCount, uint32_t sampleRate,
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2009-03-04 03:31:44 +00:00
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status_t *status, AudioSystem::audio_in_acoustics acoustics)
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{
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2009-07-21 18:16:54 +00:00
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// check for valid input source
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if ((inputSource < AudioRecord::DEFAULT_INPUT) ||
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(inputSource >= AudioRecord::NUM_INPUT_SOURCES)) {
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return 0;
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}
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2009-03-04 03:31:44 +00:00
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AudioStreamInStub* in = new AudioStreamInStub();
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status_t lStatus = in->set(format, channelCount, sampleRate, acoustics);
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if (status) {
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*status = lStatus;
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}
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if (lStatus == NO_ERROR)
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return in;
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delete in;
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return 0;
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}
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status_t AudioHardwareStub::setVoiceVolume(float volume)
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{
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return NO_ERROR;
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}
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status_t AudioHardwareStub::setMasterVolume(float volume)
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{
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return NO_ERROR;
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}
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status_t AudioHardwareStub::dumpInternals(int fd, const Vector<String16>& args)
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{
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const size_t SIZE = 256;
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char buffer[SIZE];
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String8 result;
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result.append("AudioHardwareStub::dumpInternals\n");
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snprintf(buffer, SIZE, "\tmMicMute: %s\n", mMicMute? "true": "false");
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result.append(buffer);
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::write(fd, result.string(), result.size());
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return NO_ERROR;
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}
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status_t AudioHardwareStub::dump(int fd, const Vector<String16>& args)
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{
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dumpInternals(fd, args);
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return NO_ERROR;
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}
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// ----------------------------------------------------------------------------
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status_t AudioStreamOutStub::set(int format, int channels, uint32_t rate)
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{
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// fix up defaults
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if (format == 0) format = AudioSystem::PCM_16_BIT;
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if (channels == 0) channels = channelCount();
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if (rate == 0) rate = sampleRate();
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if ((format == AudioSystem::PCM_16_BIT) &&
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(channels == channelCount()) &&
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(rate == sampleRate()))
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return NO_ERROR;
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return BAD_VALUE;
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}
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ssize_t AudioStreamOutStub::write(const void* buffer, size_t bytes)
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{
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// fake timing for audio output
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usleep(bytes * 1000000 / sizeof(int16_t) / channelCount() / sampleRate());
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return bytes;
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}
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status_t AudioStreamOutStub::standby()
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{
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return NO_ERROR;
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}
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status_t AudioStreamOutStub::dump(int fd, const Vector<String16>& args)
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{
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const size_t SIZE = 256;
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char buffer[SIZE];
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String8 result;
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snprintf(buffer, SIZE, "AudioStreamOutStub::dump\n");
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snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
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snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
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snprintf(buffer, SIZE, "\tchannel count: %d\n", channelCount());
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snprintf(buffer, SIZE, "\tformat: %d\n", format());
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result.append(buffer);
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::write(fd, result.string(), result.size());
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return NO_ERROR;
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}
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// ----------------------------------------------------------------------------
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status_t AudioStreamInStub::set(int format, int channels, uint32_t rate,
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AudioSystem::audio_in_acoustics acoustics)
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{
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if ((format == AudioSystem::PCM_16_BIT) &&
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(channels == channelCount()) &&
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(rate == sampleRate()))
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return NO_ERROR;
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return BAD_VALUE;
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}
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ssize_t AudioStreamInStub::read(void* buffer, ssize_t bytes)
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{
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// fake timing for audio input
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usleep(bytes * 1000000 / sizeof(int16_t) / channelCount() / sampleRate());
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memset(buffer, 0, bytes);
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return bytes;
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}
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status_t AudioStreamInStub::dump(int fd, const Vector<String16>& args)
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{
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const size_t SIZE = 256;
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char buffer[SIZE];
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String8 result;
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snprintf(buffer, SIZE, "AudioStreamInStub::dump\n");
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result.append(buffer);
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snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
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result.append(buffer);
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snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
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result.append(buffer);
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snprintf(buffer, SIZE, "\tchannel count: %d\n", channelCount());
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result.append(buffer);
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snprintf(buffer, SIZE, "\tformat: %d\n", format());
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result.append(buffer);
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::write(fd, result.string(), result.size());
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return NO_ERROR;
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}
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// ----------------------------------------------------------------------------
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}; // namespace android
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