AudioFlinger: call SPDIF wrapper from AudioFlinger

Create an interface layer between the AudioFlinger and the HAL
that manages the wrapping and format conversion.

Removed unnecessary includes.
Handle rate conversion in getRenderPosition().
Try to open HAL with encoded format before wrapping with SPDIF.

Bug: 17566660
Change-Id: I00ad888ca15ff0f85b85efb8167c7f5ea761a244
Signed-off-by: Phil Burk <philburk@google.com>

1 files changed, 166 insertions, 0 deletions

diff --git a/services/audioflinger/SpdifStreamOut.cpp b/services/audioflinger/SpdifStreamOut.cpp
new file mode 100644
index 0000000..d23588e
--- /dev/null
+++ b/services/audioflinger/SpdifStreamOut.cpp
@@ -0,0 +1,166 @@
+/*
+**
+** 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)
+        : AudioStreamOut(dev,flags)
+        , mRateMultiplier(1)
+        , mSpdifEncoder(this)
+        , 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;
+
+    customConfig.format = AUDIO_FORMAT_PCM_16_BIT;
+    customConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+
+    // Some data bursts run at a higher sample rate.
+    switch(config->format) {
+        case AUDIO_FORMAT_E_AC3:
+            mRateMultiplier = 4;
+            break;
+        case AUDIO_FORMAT_AC3:
+            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;
+
+    // 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