Add support for a new AAC decoder library.

Change-Id: I867bf95f7c20503e55b38d0087ac027647834f37

3 files changed, 770 insertions, 172 deletions

diff --git a/media/libstagefright/codecs/aacdec/Android.mk b/media/libstagefright/codecs/aacdec/Android.mk
index b4445a7..5b3d216 100644
--- a/media/libstagefright/codecs/aacdec/Android.mk
+++ b/media/libstagefright/codecs/aacdec/Android.mk
@@ -1,180 +1,213 @@
 LOCAL_PATH:= $(call my-dir)
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
-	analysis_sub_band.cpp \
- 	apply_ms_synt.cpp \
- 	apply_tns.cpp \
- 	buf_getbits.cpp \
- 	byte_align.cpp \
- 	calc_auto_corr.cpp \
- 	calc_gsfb_table.cpp \
- 	calc_sbr_anafilterbank.cpp \
- 	calc_sbr_envelope.cpp \
- 	calc_sbr_synfilterbank.cpp \
- 	check_crc.cpp \
- 	dct16.cpp \
- 	dct64.cpp \
- 	decode_huff_cw_binary.cpp \
- 	decode_noise_floorlevels.cpp \
- 	deinterleave.cpp \
- 	digit_reversal_tables.cpp \
- 	dst16.cpp \
- 	dst32.cpp \
- 	dst8.cpp \
- 	esc_iquant_scaling.cpp \
- 	extractframeinfo.cpp \
- 	fft_rx4_long.cpp \
- 	fft_rx4_short.cpp \
- 	fft_rx4_tables_fxp.cpp \
- 	find_adts_syncword.cpp \
- 	fwd_long_complex_rot.cpp \
- 	fwd_short_complex_rot.cpp \
- 	gen_rand_vector.cpp \
- 	get_adif_header.cpp \
- 	get_adts_header.cpp \
- 	get_audio_specific_config.cpp \
- 	get_dse.cpp \
- 	get_ele_list.cpp \
- 	get_ga_specific_config.cpp \
- 	get_ics_info.cpp \
- 	get_prog_config.cpp \
- 	get_pulse_data.cpp \
- 	get_sbr_bitstream.cpp \
- 	get_sbr_startfreq.cpp \
- 	get_sbr_stopfreq.cpp \
- 	get_tns.cpp \
- 	getfill.cpp \
- 	getgroup.cpp \
- 	getics.cpp \
- 	getmask.cpp \
- 	hcbtables_binary.cpp \
- 	huffcb.cpp \
- 	huffdecode.cpp \
- 	hufffac.cpp \
- 	huffspec_fxp.cpp \
- 	idct16.cpp \
- 	idct32.cpp \
- 	idct8.cpp \
- 	imdct_fxp.cpp \
- 	infoinit.cpp \
- 	init_sbr_dec.cpp \
- 	intensity_right.cpp \
- 	inv_long_complex_rot.cpp \
- 	inv_short_complex_rot.cpp \
- 	iquant_table.cpp \
- 	long_term_prediction.cpp \
- 	long_term_synthesis.cpp \
- 	lt_decode.cpp \
- 	mdct_fxp.cpp \
- 	mdct_tables_fxp.cpp \
- 	mdst.cpp \
- 	mix_radix_fft.cpp \
- 	ms_synt.cpp \
- 	pns_corr.cpp \
- 	pns_intensity_right.cpp \
- 	pns_left.cpp \
- 	ps_all_pass_filter_coeff.cpp \
- 	ps_all_pass_fract_delay_filter.cpp \
- 	ps_allocate_decoder.cpp \
- 	ps_applied.cpp \
- 	ps_bstr_decoding.cpp \
- 	ps_channel_filtering.cpp \
- 	ps_decode_bs_utils.cpp \
- 	ps_decorrelate.cpp \
- 	ps_fft_rx8.cpp \
- 	ps_hybrid_analysis.cpp \
- 	ps_hybrid_filter_bank_allocation.cpp \
- 	ps_hybrid_synthesis.cpp \
- 	ps_init_stereo_mixing.cpp \
- 	ps_pwr_transient_detection.cpp \
- 	ps_read_data.cpp \
- 	ps_stereo_processing.cpp \
- 	pulse_nc.cpp \
- 	pv_div.cpp \
- 	pv_log2.cpp \
- 	pv_normalize.cpp \
- 	pv_pow2.cpp \
- 	pv_sine.cpp \
- 	pv_sqrt.cpp \
- 	pvmp4audiodecoderconfig.cpp \
- 	pvmp4audiodecoderframe.cpp \
- 	pvmp4audiodecodergetmemrequirements.cpp \
- 	pvmp4audiodecoderinitlibrary.cpp \
- 	pvmp4audiodecoderresetbuffer.cpp \
- 	q_normalize.cpp \
- 	qmf_filterbank_coeff.cpp \
- 	sbr_aliasing_reduction.cpp \
- 	sbr_applied.cpp \
- 	sbr_code_book_envlevel.cpp \
- 	sbr_crc_check.cpp \
- 	sbr_create_limiter_bands.cpp \
- 	sbr_dec.cpp \
- 	sbr_decode_envelope.cpp \
- 	sbr_decode_huff_cw.cpp \
- 	sbr_downsample_lo_res.cpp \
- 	sbr_envelope_calc_tbl.cpp \
- 	sbr_envelope_unmapping.cpp \
- 	sbr_extract_extended_data.cpp \
- 	sbr_find_start_andstop_band.cpp \
- 	sbr_generate_high_freq.cpp \
- 	sbr_get_additional_data.cpp \
- 	sbr_get_cpe.cpp \
- 	sbr_get_dir_control_data.cpp \
- 	sbr_get_envelope.cpp \
- 	sbr_get_header_data.cpp \
- 	sbr_get_noise_floor_data.cpp \
- 	sbr_get_sce.cpp \
- 	sbr_inv_filt_levelemphasis.cpp \
- 	sbr_open.cpp \
- 	sbr_read_data.cpp \
- 	sbr_requantize_envelope_data.cpp \
- 	sbr_reset_dec.cpp \
- 	sbr_update_freq_scale.cpp \
- 	set_mc_info.cpp \
- 	sfb.cpp \
- 	shellsort.cpp \
- 	synthesis_sub_band.cpp \
- 	tns_ar_filter.cpp \
- 	tns_decode_coef.cpp \
- 	tns_inv_filter.cpp \
- 	trans4m_freq_2_time_fxp.cpp \
- 	trans4m_time_2_freq_fxp.cpp \
- 	unpack_idx.cpp \
- 	window_tables_fxp.cpp \
- 	pvmp4setaudioconfig.cpp \
-
-LOCAL_CFLAGS := -DAAC_PLUS -DHQ_SBR -DPARAMETRICSTEREO -DOSCL_IMPORT_REF= -DOSCL_EXPORT_REF= -DOSCL_UNUSED_ARG=
-
-LOCAL_C_INCLUDES := \
-        frameworks/av/media/libstagefright/include \
-
-LOCAL_ARM_MODE := arm
-
-LOCAL_MODULE := libstagefright_aacdec
-
-include $(BUILD_STATIC_LIBRARY)
-
-################################################################################
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
+
+AAC_LIBRARY = pv
+
+ifeq ($(AAC_LIBRARY), fraunhofer)
+  include $(CLEAR_VARS)
+
+  LOCAL_SRC_FILES := \
+          SoftAAC2.cpp
+
+  LOCAL_C_INCLUDES := \
+          frameworks/av/media/libstagefright/include \
+          frameworks/native/include/media/openmax \
+          external/aac/libAACdec/include \
+          external/aac/libCDK/include \
+          external/aac/libMpegTPDec/include \
+          external/aac/libSBRdec/include \
+          external/aac/libSYS/include
+
+  LOCAL_CFLAGS :=
+
+  LOCAL_STATIC_LIBRARIES := \
+          libAACdec libMpegTPDec libSBRdec libCDK libSYS
+
+  LOCAL_SHARED_LIBRARIES := \
+          libstagefright_omx libstagefright_foundation libutils
+
+  LOCAL_MODULE := libstagefright_soft_aacdec
+  LOCAL_MODULE_TAGS := optional
+
+  include $(BUILD_SHARED_LIBRARY)
+
+else # pv
+
+  LOCAL_SRC_FILES := \
+          analysis_sub_band.cpp \
+          apply_ms_synt.cpp \
+          apply_tns.cpp \
+          buf_getbits.cpp \
+          byte_align.cpp \
+          calc_auto_corr.cpp \
+          calc_gsfb_table.cpp \
+          calc_sbr_anafilterbank.cpp \
+          calc_sbr_envelope.cpp \
+          calc_sbr_synfilterbank.cpp \
+          check_crc.cpp \
+          dct16.cpp \
+          dct64.cpp \
+          decode_huff_cw_binary.cpp \
+          decode_noise_floorlevels.cpp \
+          deinterleave.cpp \
+          digit_reversal_tables.cpp \
+          dst16.cpp \
+          dst32.cpp \
+          dst8.cpp \
+          esc_iquant_scaling.cpp \
+          extractframeinfo.cpp \
+          fft_rx4_long.cpp \
+          fft_rx4_short.cpp \
+          fft_rx4_tables_fxp.cpp \
+          find_adts_syncword.cpp \
+          fwd_long_complex_rot.cpp \
+          fwd_short_complex_rot.cpp \
+          gen_rand_vector.cpp \
+          get_adif_header.cpp \
+          get_adts_header.cpp \
+          get_audio_specific_config.cpp \
+          get_dse.cpp \
+          get_ele_list.cpp \
+          get_ga_specific_config.cpp \
+          get_ics_info.cpp \
+          get_prog_config.cpp \
+          get_pulse_data.cpp \
+          get_sbr_bitstream.cpp \
+          get_sbr_startfreq.cpp \
+          get_sbr_stopfreq.cpp \
+          get_tns.cpp \
+          getfill.cpp \
+          getgroup.cpp \
+          getics.cpp \
+          getmask.cpp \
+          hcbtables_binary.cpp \
+          huffcb.cpp \
+          huffdecode.cpp \
+          hufffac.cpp \
+          huffspec_fxp.cpp \
+          idct16.cpp \
+          idct32.cpp \
+          idct8.cpp \
+          imdct_fxp.cpp \
+          infoinit.cpp \
+          init_sbr_dec.cpp \
+          intensity_right.cpp \
+          inv_long_complex_rot.cpp \
+          inv_short_complex_rot.cpp \
+          iquant_table.cpp \
+          long_term_prediction.cpp \
+          long_term_synthesis.cpp \
+          lt_decode.cpp \
+          mdct_fxp.cpp \
+          mdct_tables_fxp.cpp \
+          mdst.cpp \
+          mix_radix_fft.cpp \
+          ms_synt.cpp \
+          pns_corr.cpp \
+          pns_intensity_right.cpp \
+          pns_left.cpp \
+          ps_all_pass_filter_coeff.cpp \
+          ps_all_pass_fract_delay_filter.cpp \
+          ps_allocate_decoder.cpp \
+          ps_applied.cpp \
+          ps_bstr_decoding.cpp \
+          ps_channel_filtering.cpp \
+          ps_decode_bs_utils.cpp \
+          ps_decorrelate.cpp \
+          ps_fft_rx8.cpp \
+          ps_hybrid_analysis.cpp \
+          ps_hybrid_filter_bank_allocation.cpp \
+          ps_hybrid_synthesis.cpp \
+          ps_init_stereo_mixing.cpp \
+          ps_pwr_transient_detection.cpp \
+          ps_read_data.cpp \
+          ps_stereo_processing.cpp \
+          pulse_nc.cpp \
+          pv_div.cpp \
+          pv_log2.cpp \
+          pv_normalize.cpp \
+          pv_pow2.cpp \
+          pv_sine.cpp \
+          pv_sqrt.cpp \
+          pvmp4audiodecoderconfig.cpp \
+          pvmp4audiodecoderframe.cpp \
+          pvmp4audiodecodergetmemrequirements.cpp \
+          pvmp4audiodecoderinitlibrary.cpp \
+          pvmp4audiodecoderresetbuffer.cpp \
+          q_normalize.cpp \
+          qmf_filterbank_coeff.cpp \
+          sbr_aliasing_reduction.cpp \
+          sbr_applied.cpp \
+          sbr_code_book_envlevel.cpp \
+          sbr_crc_check.cpp \
+          sbr_create_limiter_bands.cpp \
+          sbr_dec.cpp \
+          sbr_decode_envelope.cpp \
+          sbr_decode_huff_cw.cpp \
+          sbr_downsample_lo_res.cpp \
+          sbr_envelope_calc_tbl.cpp \
+          sbr_envelope_unmapping.cpp \
+          sbr_extract_extended_data.cpp \
+          sbr_find_start_andstop_band.cpp \
+          sbr_generate_high_freq.cpp \
+          sbr_get_additional_data.cpp \
+          sbr_get_cpe.cpp \
+          sbr_get_dir_control_data.cpp \
+          sbr_get_envelope.cpp \
+          sbr_get_header_data.cpp \
+          sbr_get_noise_floor_data.cpp \
+          sbr_get_sce.cpp \
+          sbr_inv_filt_levelemphasis.cpp \
+          sbr_open.cpp \
+          sbr_read_data.cpp \
+          sbr_requantize_envelope_data.cpp \
+          sbr_reset_dec.cpp \
+          sbr_update_freq_scale.cpp \
+          set_mc_info.cpp \
+          sfb.cpp \
+          shellsort.cpp \
+          synthesis_sub_band.cpp \
+          tns_ar_filter.cpp \
+          tns_decode_coef.cpp \
+          tns_inv_filter.cpp \
+          trans4m_freq_2_time_fxp.cpp \
+          trans4m_time_2_freq_fxp.cpp \
+          unpack_idx.cpp \
+          window_tables_fxp.cpp \
+          pvmp4setaudioconfig.cpp \
+
+  LOCAL_CFLAGS := -DAAC_PLUS -DHQ_SBR -DPARAMETRICSTEREO -DOSCL_IMPORT_REF= -DOSCL_EXPORT_REF= -DOSCL_UNUSED_ARG=
+
+  LOCAL_C_INCLUDES := \
+          frameworks/av/media/libstagefright/include \
+
+  LOCAL_ARM_MODE := arm
+
+  LOCAL_MODULE := libstagefright_aacdec
+
+  include $(BUILD_STATIC_LIBRARY)
+
+  ################################################################################
+
+  include $(CLEAR_VARS)
+
+  LOCAL_SRC_FILES := \
         SoftAAC.cpp
 
-LOCAL_C_INCLUDES := \
-        frameworks/av/media/libstagefright/include \
-        frameworks/native/include/media/openmax
+  LOCAL_C_INCLUDES := \
+          frameworks/av/media/libstagefright/include \
+          frameworks/native/include/media/openmax
+
+  LOCAL_CFLAGS := -DOSCL_IMPORT_REF=
 
-LOCAL_CFLAGS := -DOSCL_IMPORT_REF=
+  LOCAL_STATIC_LIBRARIES := \
+          libstagefright_aacdec
 
-LOCAL_STATIC_LIBRARIES := \
-        libstagefright_aacdec
+  LOCAL_SHARED_LIBRARIES := \
+          libstagefright_omx libstagefright_foundation libutils
 
-LOCAL_SHARED_LIBRARIES := \
-        libstagefright_omx libstagefright_foundation libutils
+  LOCAL_MODULE := libstagefright_soft_aacdec
+  LOCAL_MODULE_TAGS := optional
 
-LOCAL_MODULE := libstagefright_soft_aacdec
-LOCAL_MODULE_TAGS := optional
+  include $(BUILD_SHARED_LIBRARY)
 
-include $(BUILD_SHARED_LIBRARY)
+endif # $(AAC_LIBRARY)
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
new file mode 100644
index 0000000..4589d37
--- /dev/null
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -0,0 +1,492 @@
+/*
+ * Copyright (C) 2012 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 "SoftAAC2"
+#include <utils/Log.h>
+
+#include "SoftAAC2.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/hexdump.h>
+
+#define FILEREAD_MAX_LAYERS 2
+
+namespace android {
+
+static Mutex gAACLibraryLock;
+static int gAACLibraryCount = 0;
+
+void initializeAACLibrary() {
+  Mutex::Autolock autoLock(gAACLibraryLock);
+  if (gAACLibraryCount++ == 0) {
+    CDKprolog();
+  }
+}
+
+void cleanupAACLibrary() {
+  Mutex::Autolock autoLock(gAACLibraryLock);
+  if (--gAACLibraryCount == 0) {
+    CDKepilog();
+  }
+}
+
+template<class T>
+static void InitOMXParams(T *params) {
+    params->nSize = sizeof(T);
+    params->nVersion.s.nVersionMajor = 1;
+    params->nVersion.s.nVersionMinor = 0;
+    params->nVersion.s.nRevision = 0;
+    params->nVersion.s.nStep = 0;
+}
+
+SoftAAC2::SoftAAC2(
+        const char *name,
+        const OMX_CALLBACKTYPE *callbacks,
+        OMX_PTR appData,
+        OMX_COMPONENTTYPE **component)
+    : SimpleSoftOMXComponent(name, callbacks, appData, component),
+      mAACDecoder(NULL),
+      mStreamInfo(NULL),
+      mIsADTS(false),
+      mInputBufferCount(0),
+      mSignalledError(false),
+      mAnchorTimeUs(0),
+      mNumSamplesOutput(0),
+      mOutputPortSettingsChange(NONE) {
+    initializeAACLibrary();
+    initPorts();
+    CHECK_EQ(initDecoder(), (status_t)OK);
+}
+
+SoftAAC2::~SoftAAC2() {
+    aacDecoder_Close(mAACDecoder);
+    cleanupAACLibrary();
+}
+
+void SoftAAC2::initPorts() {
+    OMX_PARAM_PORTDEFINITIONTYPE def;
+    InitOMXParams(&def);
+
+    def.nPortIndex = 0;
+    def.eDir = OMX_DirInput;
+    def.nBufferCountMin = kNumBuffers;
+    def.nBufferCountActual = def.nBufferCountMin;
+    def.nBufferSize = 8192;
+    def.bEnabled = OMX_TRUE;
+    def.bPopulated = OMX_FALSE;
+    def.eDomain = OMX_PortDomainAudio;
+    def.bBuffersContiguous = OMX_FALSE;
+    def.nBufferAlignment = 1;
+
+    def.format.audio.cMIMEType = const_cast<char *>("audio/aac");
+    def.format.audio.pNativeRender = NULL;
+    def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+    def.format.audio.eEncoding = OMX_AUDIO_CodingAAC;
+
+    addPort(def);
+
+    def.nPortIndex = 1;
+    def.eDir = OMX_DirOutput;
+    def.nBufferCountMin = kNumBuffers;
+    def.nBufferCountActual = def.nBufferCountMin;
+    def.nBufferSize = 8192;
+    def.bEnabled = OMX_TRUE;
+    def.bPopulated = OMX_FALSE;
+    def.eDomain = OMX_PortDomainAudio;
+    def.bBuffersContiguous = OMX_FALSE;
+    def.nBufferAlignment = 2;
+
+    def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
+    def.format.audio.pNativeRender = NULL;
+    def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+    def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
+
+    addPort(def);
+}
+
+status_t SoftAAC2::initDecoder() {
+    status_t status = UNKNOWN_ERROR;
+    mAACDecoder = aacDecoder_Open(TT_MP4_RAW, /* num layers */ 1);
+    if (mAACDecoder != NULL) {
+        mStreamInfo = aacDecoder_GetStreamInfo(mAACDecoder);
+        if (mStreamInfo != NULL) {
+            status = OK;
+        }
+    }
+    return status;
+}
+
+OMX_ERRORTYPE SoftAAC2::internalGetParameter(
+        OMX_INDEXTYPE index, OMX_PTR params) {
+    switch (index) {
+        case OMX_IndexParamAudioAac:
+        {
+            OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
+                (OMX_AUDIO_PARAM_AACPROFILETYPE *)params;
+
+            if (aacParams->nPortIndex != 0) {
+                return OMX_ErrorUndefined;
+            }
+
+            aacParams->nBitRate = 0;
+            aacParams->nAudioBandWidth = 0;
+            aacParams->nAACtools = 0;
+            aacParams->nAACERtools = 0;
+            aacParams->eAACProfile = OMX_AUDIO_AACObjectMain;
+
+            aacParams->eAACStreamFormat =
+                mIsADTS
+                    ? OMX_AUDIO_AACStreamFormatMP4ADTS
+                    : OMX_AUDIO_AACStreamFormatMP4FF;
+
+            aacParams->eChannelMode = OMX_AUDIO_ChannelModeStereo;
+
+            if (!isConfigured()) {
+                aacParams->nChannels = 1;
+                aacParams->nSampleRate = 44100;
+                aacParams->nFrameLength = 0;
+            } else {
+                aacParams->nChannels = mStreamInfo->channelConfig;
+                aacParams->nSampleRate = mStreamInfo->aacSampleRate;
+                aacParams->nFrameLength = mStreamInfo->aacSamplesPerFrame;
+            }
+
+            return OMX_ErrorNone;
+        }
+
+        case OMX_IndexParamAudioPcm:
+        {
+            OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+                (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+            if (pcmParams->nPortIndex != 1) {
+                return OMX_ErrorUndefined;
+            }
+
+            pcmParams->eNumData = OMX_NumericalDataSigned;
+            pcmParams->eEndian = OMX_EndianBig;
+            pcmParams->bInterleaved = OMX_TRUE;
+            pcmParams->nBitPerSample = 16;
+            pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
+            pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelLF;
+            pcmParams->eChannelMapping[1] = OMX_AUDIO_ChannelRF;
+
+            if (!isConfigured()) {
+                pcmParams->nChannels = 1;
+                pcmParams->nSamplingRate = 44100;
+            } else {
+                pcmParams->nChannels = mStreamInfo->channelConfig;
+                pcmParams->nSamplingRate = mStreamInfo->sampleRate;
+            }
+
+            return OMX_ErrorNone;
+        }
+
+        default:
+            return SimpleSoftOMXComponent::internalGetParameter(index, params);
+    }
+}
+
+OMX_ERRORTYPE SoftAAC2::internalSetParameter(
+        OMX_INDEXTYPE index, const OMX_PTR params) {
+    switch (index) {
+        case OMX_IndexParamStandardComponentRole:
+        {
+            const OMX_PARAM_COMPONENTROLETYPE *roleParams =
+                (const OMX_PARAM_COMPONENTROLETYPE *)params;
+
+            if (strncmp((const char *)roleParams->cRole,
+                        "audio_decoder.aac",
+                        OMX_MAX_STRINGNAME_SIZE - 1)) {
+                return OMX_ErrorUndefined;
+            }
+
+            return OMX_ErrorNone;
+        }
+
+        case OMX_IndexParamAudioAac:
+        {
+            const OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
+                (const OMX_AUDIO_PARAM_AACPROFILETYPE *)params;
+
+            if (aacParams->nPortIndex != 0) {
+                return OMX_ErrorUndefined;
+            }
+
+            if (aacParams->eAACStreamFormat == OMX_AUDIO_AACStreamFormatMP4FF) {
+                mIsADTS = false;
+            } else if (aacParams->eAACStreamFormat
+                        == OMX_AUDIO_AACStreamFormatMP4ADTS) {
+                mIsADTS = true;
+            } else {
+                return OMX_ErrorUndefined;
+            }
+
+            return OMX_ErrorNone;
+        }
+
+        case OMX_IndexParamAudioPcm:
+        {
+            const OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+                (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+            if (pcmParams->nPortIndex != 1) {
+                return OMX_ErrorUndefined;
+            }
+
+            return OMX_ErrorNone;
+        }
+
+        default:
+            return SimpleSoftOMXComponent::internalSetParameter(index, params);
+    }
+}
+
+bool SoftAAC2::isConfigured() const {
+    return mInputBufferCount > 0;
+}
+
+void SoftAAC2::onQueueFilled(OMX_U32 portIndex) {
+    if (mSignalledError || mOutputPortSettingsChange != NONE) {
+        return;
+    }
+
+    UCHAR* inBuffer[FILEREAD_MAX_LAYERS];
+    UINT inBufferLength[FILEREAD_MAX_LAYERS] = {0};
+    UINT bytesValid[FILEREAD_MAX_LAYERS] = {0};
+    AAC_DECODER_ERROR decoderErr;
+
+    List<BufferInfo *> &inQueue = getPortQueue(0);
+    List<BufferInfo *> &outQueue = getPortQueue(1);
+
+    if (portIndex == 0 && mInputBufferCount == 0) {
+        ++mInputBufferCount;
+        BufferInfo *info = *inQueue.begin();
+        OMX_BUFFERHEADERTYPE *header = info->mHeader;
+
+        inBuffer[0] = header->pBuffer + header->nOffset;
+        inBufferLength[0] = header->nFilledLen;
+
+        AAC_DECODER_ERROR decoderErr =
+            aacDecoder_ConfigRaw(mAACDecoder,
+                                 inBuffer,
+                                 inBufferLength);
+
+        if (decoderErr != AAC_DEC_OK) {
+            mSignalledError = true;
+            notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
+            return;
+        }
+
+        inQueue.erase(inQueue.begin());
+        info->mOwnedByUs = false;
+        notifyEmptyBufferDone(header);
+
+        notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
+        mOutputPortSettingsChange = AWAITING_DISABLED;
+        return;
+    }
+
+    while (!inQueue.empty() && !outQueue.empty()) {
+        BufferInfo *inInfo = *inQueue.begin();
+        OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+
+        BufferInfo *outInfo = *outQueue.begin();
+        OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+
+        if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+            inQueue.erase(inQueue.begin());
+            inInfo->mOwnedByUs = false;
+            notifyEmptyBufferDone(inHeader);
+
+            outHeader->nFilledLen = 0;
+            outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+
+            outQueue.erase(outQueue.begin());
+            outInfo->mOwnedByUs = false;
+            notifyFillBufferDone(outHeader);
+            return;
+        }
+
+        if (inHeader->nOffset == 0) {
+            mAnchorTimeUs = inHeader->nTimeStamp;
+            mNumSamplesOutput = 0;
+        }
+
+        if (mIsADTS) {
+            // skip 30 bits, aac_frame_length follows.
+            // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????
+
+            const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset;
+
+            CHECK_GE(inHeader->nFilledLen, 7);
+
+            bool protectionAbsent = (adtsHeader[1] & 1);
+
+            unsigned aac_frame_length =
+                ((adtsHeader[3] & 3) << 11)
+                | (adtsHeader[4] << 3)
+                | (adtsHeader[5] >> 5);
+
+            CHECK_GE(inHeader->nFilledLen, aac_frame_length);
+
+            size_t headerSize = (protectionAbsent ? 7 : 9);
+
+            inBuffer[0] = (UCHAR *)adtsHeader + headerSize;
+            inBufferLength[0] = aac_frame_length - headerSize;
+
+            inHeader->nOffset += headerSize;
+            inHeader->nFilledLen -= headerSize;
+        } else {
+            inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
+            inBufferLength[0] = inHeader->nFilledLen;
+        }
+
+
+        // Fill and decode
+        INT_PCM *outBuffer = reinterpret_cast<INT_PCM *>(outHeader->pBuffer + outHeader->nOffset);
+        bytesValid[0] = inBufferLength[0];
+
+        int prevSampleRate = mStreamInfo->sampleRate;
+        decoderErr = aacDecoder_Fill(mAACDecoder,
+                                      inBuffer,
+                                      inBufferLength,
+                                      bytesValid);
+        decoderErr = aacDecoder_DecodeFrame(mAACDecoder,
+                                            outBuffer,
+                                            outHeader->nAllocLen,
+                                            /* flags */ 0);
+
+        /*
+         * AAC+/eAAC+ streams can be signalled in two ways: either explicitly
+         * or implicitly, according to MPEG4 spec. AAC+/eAAC+ is a dual
+         * rate system and the sampling rate in the final output is actually
+         * doubled compared with the core AAC decoder sampling rate.
+         *
+         * Explicit signalling is done by explicitly defining SBR audio object
+         * type in the bitstream. Implicit signalling is done by embedding
+         * SBR content in AAC extension payload specific to SBR, and hence
+         * requires an AAC decoder to perform pre-checks on actual audio frames.
+         *
+         * Thus, we could not say for sure whether a stream is
+         * AAC+/eAAC+ until the first data frame is decoded.
+         */
+        if (decoderErr == AAC_DEC_OK && mInputBufferCount <= 2) {
+            if (mStreamInfo->sampleRate != prevSampleRate) {
+                notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
+                mOutputPortSettingsChange = AWAITING_DISABLED;
+                return;
+            }
+        }
+
+        size_t numOutBytes =
+            mStreamInfo->frameSize * sizeof(int16_t) * mStreamInfo->numChannels;
+
+        if (decoderErr == AAC_DEC_OK) {
+            UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
+            inHeader->nFilledLen -= inBufferUsedLength;
+            inHeader->nOffset += inBufferUsedLength;
+        } else {
+            ALOGW("AAC decoder returned error %d, substituting silence",
+                  decoderErr);
+
+            memset(outHeader->pBuffer + outHeader->nOffset, 0, numOutBytes);
+
+            // Discard input buffer.
+            inHeader->nFilledLen = 0;
+
+            // fall through
+        }
+
+        if (decoderErr == AAC_DEC_OK || mNumSamplesOutput > 0) {
+            // We'll only output data if we successfully decoded it or
+            // we've previously decoded valid data, in the latter case
+            // (decode failed) we'll output a silent frame.
+            outHeader->nFilledLen = numOutBytes;
+            outHeader->nFlags = 0;
+
+            outHeader->nTimeStamp =
+                mAnchorTimeUs
+                    + (mNumSamplesOutput * 1000000ll) / mStreamInfo->sampleRate;
+
+            mNumSamplesOutput += mStreamInfo->frameSize;
+
+            outInfo->mOwnedByUs = false;
+            outQueue.erase(outQueue.begin());
+            outInfo = NULL;
+            notifyFillBufferDone(outHeader);
+            outHeader = NULL;
+        }
+
+        if (inHeader->nFilledLen == 0) {
+            inInfo->mOwnedByUs = false;
+            inQueue.erase(inQueue.begin());
+            inInfo = NULL;
+            notifyEmptyBufferDone(inHeader);
+            inHeader = NULL;
+        }
+
+        if (decoderErr == AAC_DEC_OK) {
+            ++mInputBufferCount;
+        }
+    }
+}
+
+void SoftAAC2::onPortFlushCompleted(OMX_U32 portIndex) {
+    if (portIndex == 0) {
+
+        // Make sure that the next buffer output does not still
+        // depend on fragments from the last one decoded.
+        aacDecoder_DecodeFrame(mAACDecoder,
+                               NULL,
+                               0,
+                               AACDEC_FLUSH);
+    }
+}
+
+void SoftAAC2::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) {
+    if (portIndex != 1) {
+        return;
+    }
+
+    switch (mOutputPortSettingsChange) {
+        case NONE:
+            break;
+
+        case AWAITING_DISABLED:
+        {
+            CHECK(!enabled);
+            mOutputPortSettingsChange = AWAITING_ENABLED;
+            break;
+        }
+
+        default:
+        {
+            CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED);
+            CHECK(enabled);
+            mOutputPortSettingsChange = NONE;
+            break;
+        }
+    }
+}
+
+}  // namespace android
+
+android::SoftOMXComponent *createSoftOMXComponent(
+        const char *name, const OMX_CALLBACKTYPE *callbacks,
+        OMX_PTR appData, OMX_COMPONENTTYPE **component) {
+    return new android::SoftAAC2(name, callbacks, appData, component);
+}
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.h b/media/libstagefright/codecs/aacdec/SoftAAC2.h
new file mode 100644
index 0000000..828b34e
--- /dev/null
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.h
@@ -0,0 +1,73 @@
+/*
+ * Copyright (C) 2012 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 SOFT_AAC_2_H_
+#define SOFT_AAC_2_H_
+
+#include "SimpleSoftOMXComponent.h"
+
+#include "aacdecoder_lib.h"
+
+namespace android {
+
+struct SoftAAC2 : public SimpleSoftOMXComponent {
+    SoftAAC2(const char *name,
+            const OMX_CALLBACKTYPE *callbacks,
+            OMX_PTR appData,
+            OMX_COMPONENTTYPE **component);
+
+protected:
+    virtual ~SoftAAC2();
+
+    virtual OMX_ERRORTYPE internalGetParameter(
+            OMX_INDEXTYPE index, OMX_PTR params);
+
+    virtual OMX_ERRORTYPE internalSetParameter(
+            OMX_INDEXTYPE index, const OMX_PTR params);
+
+    virtual void onQueueFilled(OMX_U32 portIndex);
+    virtual void onPortFlushCompleted(OMX_U32 portIndex);
+    virtual void onPortEnableCompleted(OMX_U32 portIndex, bool enabled);
+
+private:
+    enum {
+        kNumBuffers = 4
+    };
+
+    HANDLE_AACDECODER mAACDecoder;
+    CStreamInfo *mStreamInfo;
+    bool mIsADTS;
+    size_t mInputBufferCount;
+    bool mSignalledError;
+    int64_t mAnchorTimeUs;
+    int64_t mNumSamplesOutput;
+
+    enum {
+        NONE,
+        AWAITING_DISABLED,
+        AWAITING_ENABLED
+    } mOutputPortSettingsChange;
+
+    void initPorts();
+    status_t initDecoder();
+    bool isConfigured() const;
+
+    DISALLOW_EVIL_CONSTRUCTORS(SoftAAC2);
+};
+
+}  // namespace android
+
+#endif  // SOFT_AAC_2_H_