| /* |
| ** |
| ** 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 |