Merge "Add SpdifStreamIn data path" into main (6490d84c) · Commits · e / os / android_frameworks_av

services/audioflinger/AudioFlinger.cpp

+11 −33

Original line number	Diff line number	Diff line
		@@ -3196,41 +3196,19 @@ sp<IAfThreadBase> AudioFlinger::openInput_l(audio_module_handle_t module,
		return 0;
		}

		audio_config_t halconfig = *config;
		sp<DeviceHalInterface> inHwHal = inHwDev->hwDevice();
		sp<StreamInHalInterface> inStream;
		status_t status = inHwHal->openInputStream(
		*input, devices, &halconfig, flags, address, source,
		outputDevice, outputDeviceAddress, &inStream);
		ALOGV("openInput_l() openInputStream returned input %p, devices %#x, SamplingRate %d"
		", Format %#x, Channels %#x, flags %#x, status %d addr %s",
		inStream.get(),
		AudioStreamIn *inputStream = nullptr;
		status_t status = inHwDev->openInputStream(
		&inputStream,
		*input,
		devices,
		halconfig.sample_rate,
		halconfig.format,
		halconfig.channel_mask,
		flags,
		status, address);

		// 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(config->format) &&
		audio_is_linear_pcm(halconfig.format) &&
		(halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
		(audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_LIMIT) &&
		(audio_channel_count_from_in_mask(config->channel_mask) <= FCC_LIMIT)) {
		// FIXME describe the change proposed by HAL (save old values so we can log them here)
		ALOGV("openInput_l() reopening with proposed sampling rate and channel mask");
		inStream.clear();
		status = inHwHal->openInputStream(
		*input, devices, &halconfig, flags, address, source,
		outputDevice, outputDeviceAddress, &inStream);
		// FIXME log this new status; HAL should not propose any further changes
		}

		if (status == NO_ERROR && inStream != 0) {
		AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream, flags);
		config,
		address,
		source,
		outputDevice,
		outputDeviceAddress.c_str());

		if (status == NO_ERROR) {
		if ((flags & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0) {
		const sp<IAfMmapCaptureThread> thread =
		IAfMmapCaptureThread::create(this, *input, inHwDev, inputStream, mSystemReady);

services/audioflinger/Threads.cpp

+18 −5

Original line number	Diff line number	Diff line
		@@ -9544,10 +9544,24 @@ void RecordThread::ioConfigChanged_l(audio_io_config_event_t event, pid_t pid,

		void RecordThread::readInputParameters_l()
		{
		status_t result = mInput->stream->getAudioProperties(&mSampleRate, &mChannelMask, &mHALFormat);
		LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving audio properties from HAL: %d", result);
		mFormat = mHALFormat;
		const audio_config_base_t audioConfig = mInput->getAudioProperties();
		mSampleRate = audioConfig.sample_rate;
		mChannelMask = audioConfig.channel_mask;
		if (!audio_is_input_channel(mChannelMask)) {
		LOG_ALWAYS_FATAL("Channel mask %#x not valid for input", mChannelMask);
		}

		mChannelCount = audio_channel_count_from_in_mask(mChannelMask);

		// Get actual HAL format.
		status_t result = mInput->stream->getAudioProperties(nullptr, nullptr, &mHALFormat);
		LOG_ALWAYS_FATAL_IF(result != OK, "Error when retrieving input stream format: %d", result);
		// Get format from the shim, which will be different than the HAL format
		// if recording compressed audio from IEC61937 wrapped sources.
		mFormat = audioConfig.format;
		if (!audio_is_valid_format(mFormat)) {
		LOG_ALWAYS_FATAL("Format %#x not valid for input", mFormat);
		}
		if (audio_is_linear_pcm(mFormat)) {
		LOG_ALWAYS_FATAL_IF(mChannelCount > FCC_LIMIT, "HAL channel count %d > %d",
		mChannelCount, FCC_LIMIT);
		@@ -9555,8 +9569,7 @@ void RecordThread::readInputParameters_l()
		// Can have more that FCC_LIMIT channels in encoded streams.
		ALOGI("HAL format %#x is not linear pcm", mFormat);
		}
		result = mInput->stream->getFrameSize(&mFrameSize);
		LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving frame size from HAL: %d", result);
		mFrameSize = mInput->getFrameSize();
		LOG_ALWAYS_FATAL_IF(mFrameSize <= 0, "Error frame size was %zu but must be greater than zero",
		mFrameSize);
		result = mInput->stream->getBufferSize(&mBufferSize);

services/audioflinger/datapath/Android.bp

+3 −0

Original line number	Diff line number	Diff line
		@@ -43,11 +43,14 @@ cc_library {

		srcs: [
		"AudioHwDevice.cpp",
		"AudioStreamIn.cpp",
		"AudioStreamOut.cpp",
		"SpdifStreamIn.cpp",
		"SpdifStreamOut.cpp",
		],

		header_libs: [
		"libaudioclient_headers",
		"libaudiohal_headers",
		"liberror_headers",
		],

services/audioflinger/datapath/AudioHwDevice.cpp

+95 −28

Original line number	Diff line number	Diff line
		/*
		**
		** 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.
		*
		* 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"
		@@ -21,10 +21,13 @@
		#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 {
		@@ -47,12 +50,8 @@ status_t AudioHwDevice::openOutputStream(
		auto outputStream = new AudioStreamOut(this, flags);

		// 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);
		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, address);

		if (status != NO_ERROR) {
		@@ -62,13 +61,8 @@ status_t AudioHwDevice::openOutputStream(
		// 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);
		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)
		@@ -96,6 +90,79 @@ status_t AudioHwDevice::openOutputStream(
		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;

services/audioflinger/datapath/AudioHwDevice.h

+29 −20

Original line number	Diff line number	Diff line
		/*
		**
		** 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.
		*
		* 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.
		*/

		#ifndef ANDROID_AUDIO_HW_DEVICE_H
		#define ANDROID_AUDIO_HW_DEVICE_H
		#pragma once

		#include <stdint.h>
		#include <stdlib.h>
		@@ -30,6 +29,7 @@

		namespace android {

		class AudioStreamIn;
		class AudioStreamOut;

		class AudioHwDevice {
		@@ -89,6 +89,17 @@ public:
		struct audio_config *config,
		const char *address);

		status_t 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);

		[[nodiscard]] bool supportsAudioPatches() const;

		[[nodiscard]] status_t getAudioPort(struct audio_port_v7 *port) const;
		@@ -112,5 +123,3 @@ private:
		};

		} // namespace android

		#endif // ANDROID_AUDIO_HW_DEVICE_H