blob: 45b541a4a67d3a4cbc14d3ee8622557485af6097 [file] [log] [blame]
/*
**
** Copyright 2015, 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
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** 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
** limitations under the License.
*/
#define LOG_TAG "AudioFlinger"
//#define LOG_NDEBUG 0
#include <hardware/audio.h>
#include <utils/Log.h>
#include <audio_utils/spdif/SPDIFEncoder.h>
#include "AudioHwDevice.h"
#include "AudioStreamOut.h"
#include "SpdifStreamOut.h"
namespace android {
/**
* If the AudioFlinger is processing encoded data and the HAL expects
* PCM then we need to wrap the data in an SPDIF wrapper.
*/
SpdifStreamOut::SpdifStreamOut(AudioHwDevice *dev,
audio_output_flags_t flags,
audio_format_t format)
: AudioStreamOut(dev,flags)
, mRateMultiplier(1)
, mSpdifEncoder(this, format)
, mRenderPositionHal(0)
, mPreviousHalPosition32(0)
{
}
status_t SpdifStreamOut::open(
audio_io_handle_t handle,
audio_devices_t devices,
struct audio_config *config,
const char *address)
{
struct audio_config customConfig = *config;
// Some data bursts run at a higher sample rate.
// TODO Move this into the audio_utils as a static method.
switch(config->format) {
case AUDIO_FORMAT_E_AC3:
mRateMultiplier = 4;
break;
case AUDIO_FORMAT_AC3:
case AUDIO_FORMAT_DTS:
case AUDIO_FORMAT_DTS_HD:
mRateMultiplier = 1;
break;
default:
ALOGE("ERROR SpdifStreamOut::open() unrecognized format 0x%08X\n",
config->format);
return BAD_VALUE;
}
customConfig.sample_rate = config->sample_rate * mRateMultiplier;
customConfig.format = AUDIO_FORMAT_PCM_16_BIT;
customConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
// Always print this because otherwise it could be very confusing if the
// HAL and AudioFlinger are using different formats.
// Print before open() because HAL may modify customConfig.
ALOGI("SpdifStreamOut::open() AudioFlinger requested"
" sampleRate %d, format %#x, channelMask %#x",
config->sample_rate,
config->format,
config->channel_mask);
ALOGI("SpdifStreamOut::open() HAL configured for"
" sampleRate %d, format %#x, channelMask %#x",
customConfig.sample_rate,
customConfig.format,
customConfig.channel_mask);
status_t status = AudioStreamOut::open(
handle,
devices,
&customConfig,
address);
ALOGI("SpdifStreamOut::open() status = %d", status);
return status;
}
// Account for possibly higher sample rate.
status_t SpdifStreamOut::getRenderPosition(uint32_t *frames)
{
uint32_t halPosition = 0;
status_t status = AudioStreamOut::getRenderPosition(&halPosition);
if (status != NO_ERROR) {
return status;
}
// Accumulate a 64-bit position so that we wrap at the right place.
if (mRateMultiplier != 1) {
// Maintain a 64-bit render position.
int32_t deltaHalPosition = (int32_t)(halPosition - mPreviousHalPosition32);
mPreviousHalPosition32 = halPosition;
mRenderPositionHal += deltaHalPosition;
// Scale from device sample rate to application rate.
uint64_t renderPositionApp = mRenderPositionHal / mRateMultiplier;
ALOGV("SpdifStreamOut::getRenderPosition() "
"renderPositionAppRate = %llu = %llu / %u\n",
renderPositionApp, mRenderPositionHal, mRateMultiplier);
*frames = (uint32_t)renderPositionApp;
} else {
*frames = halPosition;
}
return status;
}
int SpdifStreamOut::flush()
{
// FIXME Is there an issue here with flush being asynchronous?
mRenderPositionHal = 0;
mPreviousHalPosition32 = 0;
return AudioStreamOut::flush();
}
int SpdifStreamOut::standby()
{
mRenderPositionHal = 0;
mPreviousHalPosition32 = 0;
return AudioStreamOut::standby();
}
// Account for possibly higher sample rate.
// This is much easier when all the values are 64-bit.
status_t SpdifStreamOut::getPresentationPosition(uint64_t *frames,
struct timespec *timestamp)
{
uint64_t halFrames = 0;
status_t status = AudioStreamOut::getPresentationPosition(&halFrames, timestamp);
*frames = halFrames / mRateMultiplier;
return status;
}
size_t SpdifStreamOut::getFrameSize()
{
return sizeof(int8_t);
}
ssize_t SpdifStreamOut::writeDataBurst(const void* buffer, size_t bytes)
{
return AudioStreamOut::write(buffer, bytes);
}
ssize_t SpdifStreamOut::write(const void* buffer, size_t bytes)
{
// Write to SPDIF wrapper. It will call back to writeDataBurst().
return mSpdifEncoder.write(buffer, bytes);
}
} // namespace android