/* ** ** 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 #include #include #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) // Tell the HAL that the data will be compressed audio wrapped in a data burst. : AudioStreamOut(dev, (audio_output_flags_t) (flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO)) , mSpdifEncoder(this, format) , mApplicationFormat(AUDIO_FORMAT_DEFAULT) , mApplicationSampleRate(0) , mApplicationChannelMask(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; mApplicationFormat = config->format; mApplicationSampleRate = config->sample_rate; mApplicationChannelMask = config->channel_mask; // 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; } int SpdifStreamOut::flush() { mSpdifEncoder.reset(); return AudioStreamOut::flush(); } int SpdifStreamOut::standby() { mSpdifEncoder.reset(); return AudioStreamOut::standby(); } ssize_t SpdifStreamOut::writeDataBurst(const void* buffer, size_t bytes) { return AudioStreamOut::write(buffer, bytes); } ssize_t SpdifStreamOut::write(const void* buffer, size_t numBytes) { // Write to SPDIF wrapper. It will call back to writeDataBurst(). return mSpdifEncoder.write(buffer, numBytes); } } // namespace android