Merge "codec2: add C2SoftMP3" (d54bacf0) · Commits · e / os / android_frameworks_av

media/libstagefright/codec2/vndk/C2Store.cpp

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Original line number	Diff line number	Diff line
		@@ -407,6 +407,7 @@ C2PlatformComponentStore::C2PlatformComponentStore() {
		mComponents.emplace("c2.google.avc.encoder", "libstagefright_soft_c2avcenc.so");
		mComponents.emplace("c2.google.aac.decoder", "libstagefright_soft_c2aacdec.so");
		mComponents.emplace("c2.google.aac.encoder", "libstagefright_soft_c2aacenc.so");
		mComponents.emplace("c2.google.mp3.decoder", "libstagefright_soft_c2mp3dec.so");
		}

		c2_status_t C2PlatformComponentStore::copyBuffer(

media/libstagefright/codecs/mp3dec/Android.bp

+44 −0

Original line number	Diff line number	Diff line
		@@ -119,6 +119,50 @@ cc_library_shared {
		compile_multilib: "32",
		}

		cc_library_shared {
		name: "libstagefright_soft_c2mp3dec",
		// vendor_available: true,
		// vndk: {
		// enabled: true,
		// },

		srcs: ["C2SoftMP3.cpp"],

		cflags: [
		"-Wall",
		"-Werror",
		],

		local_include_dirs: [
		"src",
		"include",
		],

		sanitize: {
		misc_undefined: [
		"signed-integer-overflow",
		"unsigned-integer-overflow",
		],
		cfi: true,
		diag: {
		cfi: true,
		},
		},

		static_libs: [
		"libstagefright_mp3dec"
		],

		shared_libs: [
		"liblog",
		"libutils",
		"libstagefright_codec2",
		"libstagefright_codec2_vndk",
		"libstagefright_foundation",
		"libstagefright_simple_c2component",
		],
		}

		//###############################################################################
		cc_test {
		name: "libstagefright_mp3dec_test",

media/libstagefright/codecs/mp3dec/C2SoftMP3.cpp

0 → 100644

+431 −0

Original line number	Diff line number	Diff line
		/*
		* Copyright (C) 2018 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_NDEBUG 0
		#define LOG_TAG "C2SoftMP3"
		#include <utils/Log.h>

		#include "pvmp3decoder_api.h"

		#include "C2SoftMP3.h"

		#include <C2PlatformSupport.h>
		#include <SimpleC2Interface.h>

		#include <media/stagefright/foundation/ADebug.h>

		#include <numeric>

		namespace android {

		C2SoftMP3::C2SoftMP3(const char *name, c2_node_id_t id)
		: SimpleC2Component(
		SimpleC2Interface::Builder(name, id)
		.inputFormat(C2FormatCompressed)
		.outputFormat(C2FormatAudio)
		.build()),
		mConfig(nullptr),
		mDecoderBuf(nullptr) {
		}

		C2SoftMP3::~C2SoftMP3() {
		onRelease();
		}

		c2_status_t C2SoftMP3::onInit() {
		status_t err = initDecoder();
		return err == OK ? C2_OK : C2_NO_MEMORY;
		}

		c2_status_t C2SoftMP3::onStop() {
		// Make sure that the next buffer output does not still
		// depend on fragments from the last one decoded.
		pvmp3_InitDecoder(mConfig, mDecoderBuf);
		mSignalledError = false;
		mIsFirst = true;
		mSignalledOutputEos = false;

		return C2_OK;
		}

		void C2SoftMP3::onReset() {
		(void)onStop();
		}

		void C2SoftMP3::onRelease() {
		if (mDecoderBuf) {
		free(mDecoderBuf);
		mDecoderBuf = nullptr;
		}

		if (mConfig) {
		delete mConfig;
		mConfig = nullptr;
		}
		}

		status_t C2SoftMP3::initDecoder() {
		mConfig = new tPVMP3DecoderExternal{};
		if (!mConfig) return NO_MEMORY;
		mConfig->equalizerType = flat;
		mConfig->crcEnabled = false;

		size_t memRequirements = pvmp3_decoderMemRequirements();
		mDecoderBuf = malloc(memRequirements);
		if (!mDecoderBuf) return NO_MEMORY;

		pvmp3_InitDecoder(mConfig, mDecoderBuf);

		mNumChannels = 2;
		mSamplingRate = 44100;
		mIsFirst = true;
		mSignalledError = false;
		mSignalledOutputEos = false;

		return OK;
		}

		/* The below code is borrowed from ./test/mp3reader.cpp */
		static bool parseMp3Header(uint32_t header, size_t *frame_size,
		uint32_t *out_sampling_rate = nullptr,
		uint32_t *out_channels = nullptr,
		uint32_t *out_bitrate = nullptr,
		uint32_t *out_num_samples = nullptr) {
		*frame_size = 0;
		if (out_sampling_rate) *out_sampling_rate = 0;
		if (out_channels) *out_channels = 0;
		if (out_bitrate) *out_bitrate = 0;
		if (out_num_samples) *out_num_samples = 1152;

		if ((header & 0xffe00000) != 0xffe00000) return false;

		unsigned version = (header >> 19) & 3;
		if (version == 0x01) return false;

		unsigned layer = (header >> 17) & 3;
		if (layer == 0x00) return false;

		unsigned bitrate_index = (header >> 12) & 0x0f;
		if (bitrate_index == 0 \|\| bitrate_index == 0x0f) return false;

		unsigned sampling_rate_index = (header >> 10) & 3;
		if (sampling_rate_index == 3) return false;

		static const int kSamplingRateV1[] = { 44100, 48000, 32000 };
		int sampling_rate = kSamplingRateV1[sampling_rate_index];
		if (version == 2 /* V2 */) {
		sampling_rate /= 2;
		} else if (version == 0 /* V2.5 */) {
		sampling_rate /= 4;
		}

		unsigned padding = (header >> 9) & 1;

		if (layer == 3) { // layer I
		static const int kBitrateV1[] =
		{
		32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448
		};
		static const int kBitrateV2[] =
		{
		32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256
		};

		int bitrate = (version == 3 /* V1 */) ? kBitrateV1[bitrate_index - 1] :
		kBitrateV2[bitrate_index - 1];

		if (out_bitrate) {
		*out_bitrate = bitrate;
		}
		frame_size = (12000 bitrate / sampling_rate + padding) * 4;
		if (out_num_samples) {
		*out_num_samples = 384;
		}
		} else { // layer II or III
		static const int kBitrateV1L2[] =
		{
		32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384
		};

		static const int kBitrateV1L3[] =
		{
		32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320
		};

		static const int kBitrateV2[] =
		{
		8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160
		};

		int bitrate;
		if (version == 3 /* V1 */) {
		bitrate = (layer == 2 /* L2 */) ? kBitrateV1L2[bitrate_index - 1] :
		kBitrateV1L3[bitrate_index - 1];

		if (out_num_samples) {
		*out_num_samples = 1152;
		}
		} else { // V2 (or 2.5)
		bitrate = kBitrateV2[bitrate_index - 1];
		if (out_num_samples) {
		out_num_samples = (layer == 1 / L3 */) ? 576 : 1152;
		}
		}

		if (out_bitrate) {
		*out_bitrate = bitrate;
		}

		if (version == 3 /* V1 */) {
		frame_size = 144000 bitrate / sampling_rate + padding;
		} else { // V2 or V2.5
		size_t tmp = (layer == 1 /* L3 */) ? 72000 : 144000;
		frame_size = tmp bitrate / sampling_rate + padding;
		}
		}

		if (out_sampling_rate) {
		*out_sampling_rate = sampling_rate;
		}

		if (out_channels) {
		int channel_mode = (header >> 6) & 3;

		*out_channels = (channel_mode == 3) ? 1 : 2;
		}

		return true;
		}

		static uint32_t U32_AT(const uint8_t *ptr) {
		return ptr[0] << 24 \| ptr[1] << 16 \| ptr[2] << 8 \| ptr[3];
		}

		static status_t calculateOutSize(uint8 *header, size_t inSize,
		std::vector<size_t> *decodedSizes) {
		uint32_t channels;
		uint32_t numSamples;
		size_t frameSize;
		size_t totalInSize = 0;

		while (totalInSize + 4 < inSize) {
		if (!parseMp3Header(U32_AT(header + totalInSize), &frameSize,
		nullptr, &channels, nullptr, &numSamples)) {
		ALOGE("Error in parse mp3 header during outSize estimation");
		return UNKNOWN_ERROR;
		}
		totalInSize += frameSize;
		decodedSizes->push_back(numSamples * channels * sizeof(int16_t));
		}

		if (decodedSizes->empty()) return UNKNOWN_ERROR;

		return OK;
		}

		c2_status_t C2SoftMP3::onFlush_sm() {
		return onStop();
		}

		c2_status_t C2SoftMP3::drain(
		uint32_t drainMode,
		const std::shared_ptr<C2BlockPool> &pool) {
		(void) pool;
		if (drainMode == NO_DRAIN) {
		ALOGW("drain with NO_DRAIN: no-op");
		return C2_OK;
		}
		if (drainMode == DRAIN_CHAIN) {
		ALOGW("DRAIN_CHAIN not supported");
		return C2_OMITTED;
		}

		return C2_OK;
		}

		// TODO: Can overall error checking be improved? As in the check for validity of
		// work, pool ptr, work->input.buffers.size() == 1, ...
		// TODO: Gapless playback: decoder has a delay of 529 samples. For the first
		// frame we intend to remove 529 samples worth of data. When this is
		// done it is going to effect the timestamps of future frames. This
		// timestamp correction is handled by the client or plugin? Soft omx mp3
		// plugin also has this problem
		// TODO: Blind removal of 529 samples from the output may not work. Because
		// mpeg layer 1 frame size is 384 samples per frame. This should introduce
		// negative values and can cause SEG faults. Soft omx mp3 plugin can have
		// this problem (CHECK!)
		void C2SoftMP3::process(
		const std::unique_ptr<C2Work> &work,
		const std::shared_ptr<C2BlockPool> &pool) {
		work->result = C2_OK;
		work->workletsProcessed = 0u;
		if (mSignalledError \|\| mSignalledOutputEos) {
		work->result = C2_BAD_VALUE;
		return;
		}

		const C2ConstLinearBlock &inBuffer = work->input.buffers[0]->data().linearBlocks().front();
		bool eos = ((work->input.flags & C2FrameData::FLAG_END_OF_STREAM) != 0);
		size_t inOffset = inBuffer.offset();
		size_t inSize = inBuffer.size();
		C2ReadView rView = inBuffer.map().get();
		if (inSize && rView.error()) {
		ALOGE("read view map failed %d", rView.error());
		work->result = C2_CORRUPTED;
		return;
		}

		if (inSize == 0) {
		work->worklets.front()->output.flags = work->input.flags;
		work->worklets.front()->output.buffers.clear();
		work->worklets.front()->output.ordinal = work->input.ordinal;
		work->workletsProcessed = 1u;
		if (eos) {
		mSignalledOutputEos = true;
		ALOGV("signalled EOS");
		}
		return;
		}
		ALOGV("in buffer attr. size %zu timestamp %d frameindex %d", inSize,
		(int)work->input.ordinal.timestamp.peeku(), (int)work->input.ordinal.frameIndex.peeku());

		size_t calOutSize;
		std::vector<size_t> decodedSizes;
		if (OK != calculateOutSize(const_cast<uint8 *>(rView.data() + inOffset),
		inSize, &decodedSizes)) {
		work->result = C2_CORRUPTED;
		return;
		}
		calOutSize = std::accumulate(decodedSizes.begin(), decodedSizes.end(), 0);
		std::shared_ptr<C2LinearBlock> block;
		C2MemoryUsage usage = { C2MemoryUsage::CPU_READ, C2MemoryUsage::CPU_WRITE };
		c2_status_t err = pool->fetchLinearBlock(calOutSize, usage, &block);
		if (err != C2_OK) {
		ALOGE("fetchLinearBlock for Output failed with status %d", err);
		work->result = C2_NO_MEMORY;
		return;
		}
		C2WriteView wView = block->map().get();
		if (wView.error()) {
		ALOGE("write view map failed %d", wView.error());
		work->result = C2_CORRUPTED;
		return;
		}

		int outSize = 0;
		int outOffset = 0;
		auto it = decodedSizes.begin();
		while (inOffset < inSize) {
		if (it == decodedSizes.end()) {
		ALOGE("unexpected trailing bytes, ignoring them");
		break;
		}

		mConfig->pInputBuffer = const_cast<uint8 *>(rView.data() + inOffset);
		mConfig->inputBufferCurrentLength = (inSize - inOffset);
		mConfig->inputBufferMaxLength = 0;
		mConfig->inputBufferUsedLength = 0;
		mConfig->outputFrameSize = (calOutSize - outSize);
		mConfig->pOutputBuffer = reinterpret_cast<int16_t *> (wView.data() + outSize);

		ERROR_CODE decoderErr;
		if ((decoderErr = pvmp3_framedecoder(mConfig, mDecoderBuf))
		!= NO_DECODING_ERROR) {
		ALOGE("mp3 decoder returned error %d", decoderErr);
		if (decoderErr != NO_ENOUGH_MAIN_DATA_ERROR
		&& decoderErr != SIDE_INFO_ERROR) {
		mSignalledError = true;
		work->result = C2_CORRUPTED;
		return;
		}

		// This is recoverable, just ignore the current frame and
		// play silence instead.
		ALOGV("ignoring error and sending silence");
		if (mConfig->outputFrameSize == 0) {
		mConfig->outputFrameSize = *it / sizeof(int16_t);
		}
		memset(mConfig->pOutputBuffer, 0, mConfig->outputFrameSize * sizeof(int16_t));
		} else if (mConfig->samplingRate != mSamplingRate
		\|\| mConfig->num_channels != mNumChannels) {
		mSamplingRate = mConfig->samplingRate;
		mNumChannels = mConfig->num_channels;
		}
		if (it != mConfig->outputFrameSize sizeof(int16_t)) {
		ALOGE("panic, parsed size does not match decoded size");
		mSignalledError = true;
		work->result = C2_CORRUPTED;
		return;
		}
		outSize += mConfig->outputFrameSize * sizeof(int16_t);
		inOffset += mConfig->inputBufferUsedLength;
		it++;
		}
		if (mIsFirst) {
		mIsFirst = false;
		// The decoder delay is 529 samples, so trim that many samples off
		// the start of the first output buffer. This essentially makes this
		// decoder have zero delay, which the rest of the pipeline assumes.
		outOffset = kPVMP3DecoderDelay * mNumChannels * sizeof(int16_t);
		}
		ALOGV("out buffer attr. offset %d size %d", outOffset, outSize);
		decodedSizes.clear();
		work->worklets.front()->output.flags = work->input.flags;
		work->worklets.front()->output.buffers.clear();
		work->worklets.front()->output.buffers.push_back(createLinearBuffer(block, outOffset, outSize));
		work->worklets.front()->output.ordinal = work->input.ordinal;
		work->workletsProcessed = 1u;
		if (eos) {
		mSignalledOutputEos = true;
		ALOGV("signalled EOS");
		}
		}

		class C2SoftMp3DecFactory : public C2ComponentFactory {
		public:
		virtual c2_status_t createComponent(
		c2_node_id_t id, std::shared_ptr<C2Component>* const component,
		std::function<void(::android::C2Component*)> deleter) override {
		*component = std::shared_ptr<C2Component>(new C2SoftMP3("mp3", id), deleter);
		return C2_OK;
		}

		virtual c2_status_t createInterface(
		c2_node_id_t id, std::shared_ptr<C2ComponentInterface>* const interface,
		std::function<void(::android::C2ComponentInterface*)> deleter) override {
		*interface =
		SimpleC2Interface::Builder("mp3", id, deleter)
		.inputFormat(C2FormatCompressed)
		.outputFormat(C2FormatAudio)
		.build();
		return C2_OK;
		}

		virtual ~C2SoftMp3DecFactory() override = default;
		};

		} // namespace android

		extern "C" ::android::C2ComponentFactory* CreateCodec2Factory() {
		ALOGV("in %s", __func__);
		return new ::android::C2SoftMp3DecFactory();
		}

		extern "C" void DestroyCodec2Factory(::android::C2ComponentFactory* factory) {
		ALOGV("in %s", __func__);
		delete factory;
		}

media/libstagefright/codecs/mp3dec/C2SoftMP3.h

0 → 100644

+72 −0

Original line number	Diff line number	Diff line
		/*
		* Copyright (C) 2018 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 C2_SOFT_MP3_H_
		#define C2_SOFT_MP3_H_

		#include <SimpleC2Component.h>

		#include <media/stagefright/foundation/ABase.h>

		struct tPVMP3DecoderExternal;

		bool parseMp3Header(uint32_t header, size_t *frame_size,
		uint32_t *out_sampling_rate = nullptr,
		uint32_t *out_channels = nullptr,
		uint32_t *out_bitrate = nullptr,
		uint32_t *out_num_samples = nullptr);

		namespace android {

		struct C2SoftMP3 : public SimpleC2Component {
		C2SoftMP3(const char *name, c2_node_id_t id);
		virtual ~C2SoftMP3();

		// From SimpleC2Component
		c2_status_t onInit() override;
		c2_status_t onStop() override;
		void onReset() override;
		void onRelease() override;
		c2_status_t onFlush_sm() override;
		void process(
		const std::unique_ptr<C2Work> &work,
		const std::shared_ptr<C2BlockPool> &pool) override;
		c2_status_t drain(
		uint32_t drainMode,
		const std::shared_ptr<C2BlockPool> &pool) override;

		private:
		enum {
		kPVMP3DecoderDelay = 529 // samples
		};

		tPVMP3DecoderExternal *mConfig;
		void *mDecoderBuf;

		int32_t mNumChannels;
		int32_t mSamplingRate;
		bool mIsFirst;
		bool mSignalledError;
		bool mSignalledOutputEos;

		status_t initDecoder();

		DISALLOW_EVIL_CONSTRUCTORS(C2SoftMP3);
		};

		} // namespace android

		#endif // C2_SOFT_MP3_H_