services/audioflinger/datapath/AudioHwDevice.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
 *
 *     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 "AudioHwDevice"
//#define LOG_NDEBUG 0

#include <system/audio.h>
#include <utils/Log.h>

#include <audio_utils/spdif/SPDIFDecoder.h>
#include <audio_utils/spdif/SPDIFEncoder.h>
#include <media/AudioResamplerPublic.h>

#include "AudioHwDevice.h"
#include "AudioStreamOut.h"
#include "SpdifStreamIn.h"
#include "SpdifStreamOut.h"

namespace android {

using media::audio::common::AudioMMapPolicyInfo;
using media::audio::common::AudioMMapPolicyType;

// ----------------------------------------------------------------------------

status_t AudioHwDevice::openOutputStream(
        AudioStreamOut **ppStreamOut,
        audio_io_handle_t handle,
        audio_devices_t deviceType,
        audio_output_flags_t *flags,
        struct audio_config *config,
        const char *address,
        const std::vector<playback_track_metadata_v7_t>& sourceMetadata)
{

    struct audio_config originalConfig = *config;
    auto outputStream = new AudioStreamOut(this);

    // Try to open the HAL first using the current format.
    ALOGV("openOutputStream(), try sampleRate %d, format %#x, channelMask %#x", config->sample_rate,
            config->format, config->channel_mask);
    status_t status = outputStream->open(handle, deviceType, config, flags, address,
                                        sourceMetadata);

    if (status != NO_ERROR) {
        delete outputStream;
        outputStream = nullptr;

        // FIXME Look at any modification to the config.
        // The HAL might modify the config to suggest a wrapped format.
        // Log this so we can see what the HALs are doing.
        ALOGI("openOutputStream(), HAL returned sampleRate %d, format %#x, channelMask %#x,"
                " status %d", config->sample_rate, config->format, config->channel_mask, status);

        // If the data is encoded then try again using wrapped PCM.
        const bool wrapperNeeded = !audio_has_proportional_frames(originalConfig.format)
                && ((*flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0)
                && ((*flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0);

        if (wrapperNeeded) {
            if (SPDIFEncoder::isFormatSupported(originalConfig.format)) {
                outputStream = new SpdifStreamOut(this, originalConfig.format);
                status = outputStream->open(handle, deviceType, &originalConfig, flags, address,
                                            sourceMetadata);
                if (status != NO_ERROR) {
                    ALOGE("ERROR - openOutputStream(), SPDIF open returned %d",
                        status);
                    delete outputStream;
                    outputStream = nullptr;
                }
            } else {
                ALOGE("ERROR - openOutputStream(), SPDIFEncoder does not support format 0x%08x",
                    originalConfig.format);
            }
        }
    }

    *ppStreamOut = outputStream;
    return status;
}

status_t AudioHwDevice::openInputStream(
        AudioStreamIn **ppStreamIn,
        audio_io_handle_t handle,
        audio_devices_t deviceType,
        audio_input_flags_t flags,
        struct audio_config *config,
        const char *address,
        audio_source_t source,
        audio_devices_t outputDevice,
        const char *outputDeviceAddress) {

    struct audio_config originalConfig = *config;
    auto inputStream = new AudioStreamIn(this, flags);

    // Try to open the HAL first using the current format.
    ALOGV("openInputStream(), try sampleRate %d, format %#x, channelMask %#x", config->sample_rate,
            config->format, config->channel_mask);
    status_t status = inputStream->open(handle, deviceType, config, address, source, outputDevice,
                                        outputDeviceAddress);

    // If the input could not be opened with the requested parameters and we can handle the
    // conversion internally, try to open again with the proposed parameters.
    if (status == BAD_VALUE &&
        audio_is_linear_pcm(originalConfig.format) &&
        audio_is_linear_pcm(config->format) &&
        (config->sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
        (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_LIMIT) &&
        (audio_channel_count_from_in_mask(originalConfig.channel_mask) <= FCC_LIMIT)) {
        // FIXME describe the change proposed by HAL (save old values so we can log them here)
        ALOGV("openInputStream() reopening with proposed sampling rate and channel mask");
        status = inputStream->open(handle, deviceType, config, address, source,
                outputDevice, outputDeviceAddress);
        // FIXME log this new status; HAL should not propose any further changes
        if (status != NO_ERROR) {
            delete inputStream;
            inputStream = nullptr;
        }
    } else if (status != NO_ERROR) {
        delete inputStream;
        inputStream = nullptr;

        // FIXME Look at any modification to the config.
        // The HAL might modify the config to suggest a wrapped format.
        // Log this so we can see what the HALs are doing.
        ALOGI("openInputStream(), HAL returned sampleRate %d, format %#x, channelMask %#x,"
                " status %d", config->sample_rate, config->format, config->channel_mask, status);

        // If the data is encoded then try again using wrapped PCM.
        const bool unwrapperNeeded = !audio_has_proportional_frames(originalConfig.format)
                && ((flags & AUDIO_INPUT_FLAG_DIRECT) != 0);

        if (unwrapperNeeded) {
            if (SPDIFDecoder::isFormatSupported(originalConfig.format)) {
                inputStream = new SpdifStreamIn(this, flags, originalConfig.format);
                status = inputStream->open(handle, deviceType, &originalConfig, address, source,
                        outputDevice, outputDeviceAddress);
                if (status != NO_ERROR) {
                    ALOGE("ERROR - openInputStream(), SPDIF open returned %d",
                        status);
                    delete inputStream;
                    inputStream = nullptr;
                }
            } else {
                ALOGE("ERROR - openInputStream(), SPDIFDecoder does not support format 0x%08x",
                    originalConfig.format);
            }
        }
    }

    *ppStreamIn = inputStream;
    return status;
}

bool AudioHwDevice::supportsAudioPatches() const {
    bool result;
    return mHwDevice->supportsAudioPatches(&result) == OK ? result : false;
}

status_t AudioHwDevice::getAudioPort(struct audio_port_v7 *port) const {
    return mHwDevice->getAudioPort(port);
}

status_t AudioHwDevice::getMmapPolicyInfos(
            AudioMMapPolicyType policyType, std::vector<AudioMMapPolicyInfo> *policyInfos) const {
    return mHwDevice->getMmapPolicyInfos(policyType, policyInfos);
}

int32_t AudioHwDevice::getAAudioMixerBurstCount() const {
    return mHwDevice->getAAudioMixerBurstCount();
}

int32_t AudioHwDevice::getAAudioHardwareBurstMinUsec() const {
    return mHwDevice->getAAudioHardwareBurstMinUsec();
}

status_t AudioHwDevice::getAudioMixPort(const struct audio_port_v7 *devicePort,
                                        struct audio_port_v7 *mixPort) const {
    return mHwDevice->getAudioMixPort(devicePort, mixPort);
}


}; // namespace android