diff --git a/media/libeffects/config/src/EffectsConfig.cpp b/media/libeffects/config/src/EffectsConfig.cpp
index 218249d..90a4fbb 100644
--- a/media/libeffects/config/src/EffectsConfig.cpp
+++ b/media/libeffects/config/src/EffectsConfig.cpp
@@ -117,6 +117,8 @@
         {AUDIO_SOURCE_VOICE_COMMUNICATION, "voice_communication"},
         {AUDIO_SOURCE_UNPROCESSED, "unprocessed"},
         {AUDIO_SOURCE_VOICE_PERFORMANCE, "voice_performance"},
+        {AUDIO_SOURCE_ECHO_REFERENCE, "echo_reference"},
+        {AUDIO_SOURCE_FM_TUNER, "fm_tuner"},
 };
 
 /** Find the stream type enum corresponding to the stream type name or return false */
diff --git a/media/libstagefright/codecs/amrnb/enc/test/AmrnbEncTestEnvironment.h b/media/libstagefright/codecs/amrnb/enc/test/AmrnbEncTestEnvironment.h
new file mode 100644
index 0000000..5a2fcd1
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/enc/test/AmrnbEncTestEnvironment.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2019 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 __AMRNBENC_TEST_ENVIRONMENT_H__
+#define __AMRNBENC_TEST_ENVIRONMENT_H__
+
+#include <gtest/gtest.h>
+
+#include <getopt.h>
+
+using namespace std;
+
+class AmrnbEncTestEnvironment : public ::testing::Environment {
+  public:
+    AmrnbEncTestEnvironment() : res("/data/local/tmp/") {}
+
+    // Parses the command line arguments
+    int initFromOptions(int argc, char **argv);
+
+    void setRes(const char *_res) { res = _res; }
+
+    const string getRes() const { return res; }
+
+  private:
+    string res;
+};
+
+int AmrnbEncTestEnvironment::initFromOptions(int argc, char **argv) {
+    static struct option options[] = {{"res", required_argument, 0, 'P'}, {0, 0, 0, 0}};
+
+    while (true) {
+        int index = 0;
+        int c = getopt_long(argc, argv, "P:", options, &index);
+        if (c == -1) {
+            break;
+        }
+
+        switch (c) {
+            case 'P':
+                setRes(optarg);
+                break;
+            default:
+                break;
+        }
+    }
+
+    if (optind < argc) {
+        fprintf(stderr,
+                "unrecognized option: %s\n\n"
+                "usage: %s <gtest options> <test options>\n\n"
+                "test options are:\n\n"
+                "-P, --path: Resource files directory location\n",
+                argv[optind ?: 1], argv[0]);
+        return 2;
+    }
+    return 0;
+}
+
+#endif  // __AMRNBENC_TEST_ENVIRONMENT_H__
diff --git a/media/libstagefright/codecs/amrnb/enc/test/AmrnbEncoderTest.cpp b/media/libstagefright/codecs/amrnb/enc/test/AmrnbEncoderTest.cpp
new file mode 100644
index 0000000..fb72998
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/enc/test/AmrnbEncoderTest.cpp
@@ -0,0 +1,207 @@
+/*
+ * Copyright (C) 2019 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 "AmrnbEncoderTest"
+
+#include <utils/Log.h>
+
+#include <audio_utils/sndfile.h>
+#include <stdio.h>
+
+#include "gsmamr_enc.h"
+
+#include "AmrnbEncTestEnvironment.h"
+
+#define OUTPUT_FILE "/data/local/tmp/amrnbEncode.out"
+
+constexpr int32_t kInputBufferSize = L_FRAME * 2;  // 160 samples * 16-bit per sample.
+constexpr int32_t kOutputBufferSize = 1024;
+constexpr int32_t kNumFrameReset = 200;
+constexpr int32_t kMaxCount = 10;
+struct AmrNbEncState {
+    void *encCtx;
+    void *pidSyncCtx;
+};
+
+static AmrnbEncTestEnvironment *gEnv = nullptr;
+
+class AmrnbEncoderTest : public ::testing::TestWithParam<pair<string, int32_t>> {
+  public:
+    AmrnbEncoderTest() : mAmrEncHandle(nullptr) {}
+
+    ~AmrnbEncoderTest() {
+        if (mAmrEncHandle) {
+            free(mAmrEncHandle);
+            mAmrEncHandle = nullptr;
+        }
+    }
+
+    AmrNbEncState *mAmrEncHandle;
+    int32_t EncodeFrames(int32_t mode, FILE *fpInput, FILE *mFpOutput,
+                         int32_t frameCount = INT32_MAX);
+};
+
+int32_t AmrnbEncoderTest::EncodeFrames(int32_t mode, FILE *fpInput, FILE *mFpOutput,
+                                       int32_t frameCount) {
+    int32_t frameNum = 0;
+    uint16_t inputBuf[kInputBufferSize];
+    uint8_t outputBuf[kOutputBufferSize];
+    while (frameNum < frameCount) {
+        int32_t bytesRead = fread(inputBuf, 1, kInputBufferSize, fpInput);
+        if (bytesRead != kInputBufferSize && !feof(fpInput)) {
+            ALOGE("Unable to read data from input file");
+            return -1;
+        } else if (feof(fpInput) && bytesRead == 0) {
+            break;
+        }
+        Frame_Type_3GPP frame_type = (Frame_Type_3GPP)mode;
+        int32_t bytesGenerated =
+                AMREncode(mAmrEncHandle->encCtx, mAmrEncHandle->pidSyncCtx, (Mode)mode,
+                          (Word16 *)inputBuf, outputBuf, &frame_type, AMR_TX_WMF);
+        frameNum++;
+        if (bytesGenerated < 0) {
+            ALOGE("Error in encoging the file: Invalid output format");
+            return -1;
+        }
+
+        // Convert from WMF to RFC 3267 format.
+        if (bytesGenerated > 0) {
+            outputBuf[0] = ((outputBuf[0] << 3) | 4) & 0x7c;
+        }
+        fwrite(outputBuf, 1, bytesGenerated, mFpOutput);
+    }
+    return 0;
+}
+
+TEST_F(AmrnbEncoderTest, CreateAmrnbEncoderTest) {
+    mAmrEncHandle = (AmrNbEncState *)malloc(sizeof(AmrNbEncState));
+    ASSERT_NE(mAmrEncHandle, nullptr) << "Error in allocating memory to Codec handle";
+    for (int count = 0; count < kMaxCount; count++) {
+        int32_t status = AMREncodeInit(&mAmrEncHandle->encCtx, &mAmrEncHandle->pidSyncCtx, 0);
+        ASSERT_EQ(status, 0) << "Error creating AMR-NB encoder";
+        ALOGV("Successfully created encoder");
+    }
+    if (mAmrEncHandle) {
+        AMREncodeExit(&mAmrEncHandle->encCtx, &mAmrEncHandle->pidSyncCtx);
+        ASSERT_EQ(mAmrEncHandle->encCtx, nullptr) << "Error deleting AMR-NB encoder";
+        ASSERT_EQ(mAmrEncHandle->pidSyncCtx, nullptr) << "Error deleting AMR-NB encoder";
+        free(mAmrEncHandle);
+        mAmrEncHandle = nullptr;
+        ALOGV("Successfully deleted encoder");
+    }
+}
+
+TEST_P(AmrnbEncoderTest, EncodeTest) {
+    mAmrEncHandle = (AmrNbEncState *)malloc(sizeof(AmrNbEncState));
+    ASSERT_NE(mAmrEncHandle, nullptr) << "Error in allocating memory to Codec handle";
+    int32_t status = AMREncodeInit(&mAmrEncHandle->encCtx, &mAmrEncHandle->pidSyncCtx, 0);
+    ASSERT_EQ(status, 0) << "Error creating AMR-NB encoder";
+
+    string inputFile = gEnv->getRes() + GetParam().first;
+    FILE *fpInput = fopen(inputFile.c_str(), "rb");
+    ASSERT_NE(fpInput, nullptr) << "Error opening input file " << inputFile;
+
+    FILE *fpOutput = fopen(OUTPUT_FILE, "wb");
+    ASSERT_NE(fpOutput, nullptr) << "Error opening output file " << OUTPUT_FILE;
+
+    // Write file header.
+    fwrite("#!AMR\n", 1, 6, fpOutput);
+
+    int32_t mode = GetParam().second;
+    int32_t encodeErr = EncodeFrames(mode, fpInput, fpOutput);
+    ASSERT_EQ(encodeErr, 0) << "EncodeFrames returned error for Codec mode: " << mode;
+
+    fclose(fpOutput);
+    fclose(fpInput);
+
+    AMREncodeExit(&mAmrEncHandle->encCtx, &mAmrEncHandle->pidSyncCtx);
+    ASSERT_EQ(mAmrEncHandle->encCtx, nullptr) << "Error deleting AMR-NB encoder";
+    ASSERT_EQ(mAmrEncHandle->pidSyncCtx, nullptr) << "Error deleting AMR-NB encoder";
+    free(mAmrEncHandle);
+    mAmrEncHandle = nullptr;
+    ALOGV("Successfully deleted encoder");
+}
+
+TEST_P(AmrnbEncoderTest, ResetEncoderTest) {
+    mAmrEncHandle = (AmrNbEncState *)malloc(sizeof(AmrNbEncState));
+    ASSERT_NE(mAmrEncHandle, nullptr) << "Error in allocating memory to Codec handle";
+    int32_t status = AMREncodeInit(&mAmrEncHandle->encCtx, &mAmrEncHandle->pidSyncCtx, 0);
+    ASSERT_EQ(status, 0) << "Error creating AMR-NB encoder";
+
+    string inputFile = gEnv->getRes() + GetParam().first;
+    FILE *fpInput = fopen(inputFile.c_str(), "rb");
+    ASSERT_NE(fpInput, nullptr) << "Error opening input file " << inputFile;
+
+    FILE *fpOutput = fopen(OUTPUT_FILE, "wb");
+    ASSERT_NE(fpOutput, nullptr) << "Error opening output file " << OUTPUT_FILE;
+
+    // Write file header.
+    fwrite("#!AMR\n", 1, 6, fpOutput);
+
+    int32_t mode = GetParam().second;
+    // Encode kNumFrameReset first
+    int32_t encodeErr = EncodeFrames(mode, fpInput, fpOutput, kNumFrameReset);
+    ASSERT_EQ(encodeErr, 0) << "EncodeFrames returned error for Codec mode: " << mode;
+
+    status = AMREncodeReset(mAmrEncHandle->encCtx, mAmrEncHandle->pidSyncCtx);
+    ASSERT_EQ(status, 0) << "Error resting AMR-NB encoder";
+
+    // Start encoding again
+    encodeErr = EncodeFrames(mode, fpInput, fpOutput);
+    ASSERT_EQ(encodeErr, 0) << "EncodeFrames returned error for Codec mode: " << mode;
+
+    fclose(fpOutput);
+    fclose(fpInput);
+
+    AMREncodeExit(&mAmrEncHandle->encCtx, &mAmrEncHandle->pidSyncCtx);
+    ASSERT_EQ(mAmrEncHandle->encCtx, nullptr) << "Error deleting AMR-NB encoder";
+    ASSERT_EQ(mAmrEncHandle->pidSyncCtx, nullptr) << "Error deleting AMR-NB encoder";
+    free(mAmrEncHandle);
+    mAmrEncHandle = nullptr;
+    ALOGV("Successfully deleted encoder");
+}
+
+// TODO: Add more test vectors
+INSTANTIATE_TEST_SUITE_P(AmrnbEncoderTestAll, AmrnbEncoderTest,
+                         ::testing::Values(make_pair("bbb_raw_1ch_8khz_s16le.raw", MR475),
+                                           make_pair("bbb_raw_1ch_8khz_s16le.raw", MR515),
+                                           make_pair("bbb_raw_1ch_8khz_s16le.raw", MR59),
+                                           make_pair("bbb_raw_1ch_8khz_s16le.raw", MR67),
+                                           make_pair("bbb_raw_1ch_8khz_s16le.raw", MR74),
+                                           make_pair("bbb_raw_1ch_8khz_s16le.raw", MR795),
+                                           make_pair("bbb_raw_1ch_8khz_s16le.raw", MR102),
+                                           make_pair("bbb_raw_1ch_8khz_s16le.raw", MR122),
+                                           make_pair("sinesweepraw.raw", MR475),
+                                           make_pair("sinesweepraw.raw", MR515),
+                                           make_pair("sinesweepraw.raw", MR59),
+                                           make_pair("sinesweepraw.raw", MR67),
+                                           make_pair("sinesweepraw.raw", MR74),
+                                           make_pair("sinesweepraw.raw", MR795),
+                                           make_pair("sinesweepraw.raw", MR102),
+                                           make_pair("sinesweepraw.raw", MR122)));
+
+int main(int argc, char **argv) {
+    gEnv = new AmrnbEncTestEnvironment();
+    ::testing::AddGlobalTestEnvironment(gEnv);
+    ::testing::InitGoogleTest(&argc, argv);
+    int status = gEnv->initFromOptions(argc, argv);
+    if (status == 0) {
+        status = RUN_ALL_TESTS();
+        ALOGV("Test result = %d\n", status);
+    }
+    return status;
+}
diff --git a/media/libstagefright/codecs/amrnb/enc/test/Android.bp b/media/libstagefright/codecs/amrnb/enc/test/Android.bp
new file mode 100644
index 0000000..e8982fe
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/enc/test/Android.bp
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2019 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.
+ */
+
+cc_test {
+    name: "AmrnbEncoderTest",
+    gtest: true,
+
+    srcs: [
+        "AmrnbEncoderTest.cpp",
+    ],
+
+    static_libs: [
+        "libstagefright_amrnb_common",
+        "libstagefright_amrnbenc",
+        "libaudioutils",
+        "libsndfile",
+    ],
+
+    shared_libs: [
+        "liblog",
+    ],
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+    ],
+
+    sanitize: {
+        cfi: true,
+        misc_undefined: [
+            "unsigned-integer-overflow",
+            "signed-integer-overflow",
+        ],
+    },
+}
diff --git a/media/libstagefright/codecs/amrnb/enc/test/README.md b/media/libstagefright/codecs/amrnb/enc/test/README.md
new file mode 100644
index 0000000..4ddaa57
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/enc/test/README.md
@@ -0,0 +1,34 @@
+## Media Testing ##
+---
+#### AMR-NB Encoder :
+The Amr-Nb Encoder Test Suite validates the amrnb encoder available in libstagefright.
+
+Run the following steps to build the test suite:
+```
+m AmrnbEncoderTest
+```
+
+The 32-bit binaries will be created in the following path : ${OUT}/data/nativetest/
+
+The 64-bit binaries will be created in the following path : ${OUT}/data/nativetest64/
+
+To test 64-bit binary push binaries from nativetest64.
+```
+adb push ${OUT}/data/nativetest64/AmrnbEncoderTest/AmrnbEncoderTest /data/local/tmp/
+```
+
+To test 32-bit binary push binaries from nativetest.
+```
+adb push ${OUT}/data/nativetest/AmrnbEncoderTest/AmrnbEncoderTest /data/local/tmp/
+```
+
+The resource file for the tests is taken from [here](https://drive.google.com/drive/folders/13cM4tAaVFrmr-zGFqaAzFBbKs75pnm9b). Push these files into device for testing.
+Download amr-nb_encoder folder and push all the files in this folder to /data/local/tmp/ on the device.
+```
+adb push amr-nb_encoder/. /data/local/tmp/
+```
+
+usage: AmrnbEncoderTest -P \<path_to_folder\>
+```
+adb shell /data/local/tmp/AmrnbEncoderTest -P /data/local/tmp/
+```
diff --git a/media/libstagefright/codecs/mp3dec/test/Android.bp b/media/libstagefright/codecs/mp3dec/test/Android.bp
new file mode 100644
index 0000000..0ff8b12
--- /dev/null
+++ b/media/libstagefright/codecs/mp3dec/test/Android.bp
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2019 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.
+ */
+
+cc_test {
+    name: "Mp3DecoderTest",
+    gtest: true,
+
+    srcs: [
+        "mp3reader.cpp",
+        "Mp3DecoderTest.cpp",
+    ],
+
+    static_libs: [
+        "libstagefright_mp3dec",
+        "libsndfile",
+        "libaudioutils",
+    ],
+
+    shared_libs: [
+        "liblog",
+    ],
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+    ],
+
+    sanitize: {
+        cfi: true,
+        misc_undefined: [
+            "unsigned-integer-overflow",
+            "signed-integer-overflow",
+        ],
+    },
+}
diff --git a/media/libstagefright/codecs/mp3dec/test/Mp3DecoderTest.cpp b/media/libstagefright/codecs/mp3dec/test/Mp3DecoderTest.cpp
new file mode 100644
index 0000000..99553ec
--- /dev/null
+++ b/media/libstagefright/codecs/mp3dec/test/Mp3DecoderTest.cpp
@@ -0,0 +1,200 @@
+/*
+ * Copyright (C) 2019 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 "Mp3DecoderTest"
+
+#include <utils/Log.h>
+
+#include <audio_utils/sndfile.h>
+#include <stdio.h>
+
+#include "mp3reader.h"
+#include "pvmp3decoder_api.h"
+
+#include "Mp3DecoderTestEnvironment.h"
+
+#define OUTPUT_FILE "/data/local/tmp/mp3Decode.out"
+
+constexpr int32_t kInputBufferSize = 1024 * 10;
+constexpr int32_t kOutputBufferSize = 4608 * 2;
+constexpr int32_t kMaxCount = 10;
+constexpr int32_t kNumFrameReset = 150;
+
+static Mp3DecoderTestEnvironment *gEnv = nullptr;
+
+class Mp3DecoderTest : public ::testing::TestWithParam<string> {
+  public:
+    Mp3DecoderTest() : mConfig(nullptr) {}
+
+    ~Mp3DecoderTest() {
+        if (mConfig) {
+            delete mConfig;
+            mConfig = nullptr;
+        }
+    }
+
+    virtual void SetUp() override {
+        mConfig = new tPVMP3DecoderExternal{};
+        ASSERT_NE(mConfig, nullptr) << "Failed to initialize config. No Memory available";
+        mConfig->equalizerType = flat;
+        mConfig->crcEnabled = false;
+    }
+
+    tPVMP3DecoderExternal *mConfig;
+    Mp3Reader mMp3Reader;
+
+    ERROR_CODE DecodeFrames(void *decoderbuf, SNDFILE *outFileHandle, SF_INFO sfInfo,
+                            int32_t frameCount = INT32_MAX);
+    SNDFILE *openOutputFile(SF_INFO *sfInfo);
+};
+
+ERROR_CODE Mp3DecoderTest::DecodeFrames(void *decoderBuf, SNDFILE *outFileHandle, SF_INFO sfInfo,
+                                        int32_t frameCount) {
+    uint8_t inputBuf[kInputBufferSize];
+    int16_t outputBuf[kOutputBufferSize];
+    uint32_t bytesRead;
+    ERROR_CODE decoderErr;
+    while (frameCount > 0) {
+        bool success = mMp3Reader.getFrame(inputBuf, &bytesRead);
+        if (!success) {
+            break;
+        }
+        mConfig->inputBufferCurrentLength = bytesRead;
+        mConfig->inputBufferMaxLength = 0;
+        mConfig->inputBufferUsedLength = 0;
+        mConfig->pInputBuffer = inputBuf;
+        mConfig->pOutputBuffer = outputBuf;
+        mConfig->outputFrameSize = kOutputBufferSize / sizeof(int16_t);
+        decoderErr = pvmp3_framedecoder(mConfig, decoderBuf);
+        if (decoderErr != NO_DECODING_ERROR) break;
+        sf_writef_short(outFileHandle, outputBuf, mConfig->outputFrameSize / sfInfo.channels);
+        frameCount--;
+    }
+    return decoderErr;
+}
+
+SNDFILE *Mp3DecoderTest::openOutputFile(SF_INFO *sfInfo) {
+    memset(sfInfo, 0, sizeof(SF_INFO));
+    sfInfo->channels = mMp3Reader.getNumChannels();
+    sfInfo->format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
+    sfInfo->samplerate = mMp3Reader.getSampleRate();
+    SNDFILE *outFileHandle = sf_open(OUTPUT_FILE, SFM_WRITE, sfInfo);
+    return outFileHandle;
+}
+
+TEST_F(Mp3DecoderTest, MultiCreateMp3DecoderTest) {
+    size_t memRequirements = pvmp3_decoderMemRequirements();
+    ASSERT_NE(memRequirements, 0) << "Failed to get the memory requirement size";
+    void *decoderBuf = malloc(memRequirements);
+    ASSERT_NE(decoderBuf, nullptr)
+            << "Failed to allocate decoder memory of size " << memRequirements;
+    for (int count = 0; count < kMaxCount; count++) {
+        pvmp3_InitDecoder(mConfig, decoderBuf);
+        ALOGV("Decoder created successfully");
+    }
+    if (decoderBuf) {
+        free(decoderBuf);
+        decoderBuf = nullptr;
+    }
+}
+
+TEST_P(Mp3DecoderTest, DecodeTest) {
+    size_t memRequirements = pvmp3_decoderMemRequirements();
+    ASSERT_NE(memRequirements, 0) << "Failed to get the memory requirement size";
+    void *decoderBuf = malloc(memRequirements);
+    ASSERT_NE(decoderBuf, nullptr)
+            << "Failed to allocate decoder memory of size " << memRequirements;
+
+    pvmp3_InitDecoder(mConfig, decoderBuf);
+    ALOGV("Decoder created successfully");
+    string inputFile = gEnv->getRes() + GetParam();
+    bool status = mMp3Reader.init(inputFile.c_str());
+    ASSERT_TRUE(status) << "Unable to initialize the mp3Reader";
+
+    // Open the output file.
+    SF_INFO sfInfo;
+    SNDFILE *outFileHandle = openOutputFile(&sfInfo);
+    ASSERT_NE(outFileHandle, nullptr) << "Error opening output file for writing decoded output";
+
+    ERROR_CODE decoderErr = DecodeFrames(decoderBuf, outFileHandle, sfInfo);
+    ASSERT_EQ(decoderErr, NO_DECODING_ERROR) << "Failed to decode the frames";
+    ASSERT_EQ(sfInfo.channels, mConfig->num_channels) << "Number of channels does not match";
+    ASSERT_EQ(sfInfo.samplerate, mConfig->samplingRate) << "Sample rate does not match";
+
+    mMp3Reader.close();
+    sf_close(outFileHandle);
+    if (decoderBuf) {
+        free(decoderBuf);
+        decoderBuf = nullptr;
+    }
+}
+
+TEST_P(Mp3DecoderTest, ResetDecoderTest) {
+    size_t memRequirements = pvmp3_decoderMemRequirements();
+    ASSERT_NE(memRequirements, 0) << "Failed to get the memory requirement size";
+    void *decoderBuf = malloc(memRequirements);
+    ASSERT_NE(decoderBuf, nullptr)
+            << "Failed to allocate decoder memory of size " << memRequirements;
+
+    pvmp3_InitDecoder(mConfig, decoderBuf);
+    ALOGV("Decoder created successfully.");
+    string inputFile = gEnv->getRes() + GetParam();
+    bool status = mMp3Reader.init(inputFile.c_str());
+    ASSERT_TRUE(status) << "Unable to initialize the mp3Reader";
+
+    // Open the output file.
+    SF_INFO sfInfo;
+    SNDFILE *outFileHandle = openOutputFile(&sfInfo);
+    ASSERT_NE(outFileHandle, nullptr) << "Error opening output file for writing decoded output";
+
+    ERROR_CODE decoderErr;
+    decoderErr = DecodeFrames(decoderBuf, outFileHandle, sfInfo, kNumFrameReset);
+    ASSERT_EQ(decoderErr, NO_DECODING_ERROR) << "Failed to decode the frames";
+    ASSERT_EQ(sfInfo.channels, mConfig->num_channels) << "Number of channels does not match";
+    ASSERT_EQ(sfInfo.samplerate, mConfig->samplingRate) << "Sample rate does not match";
+
+    pvmp3_resetDecoder(decoderBuf);
+    // Decode the same file.
+    decoderErr = DecodeFrames(decoderBuf, outFileHandle, sfInfo);
+    ASSERT_EQ(decoderErr, NO_DECODING_ERROR) << "Failed to decode the frames";
+    ASSERT_EQ(sfInfo.channels, mConfig->num_channels) << "Number of channels does not match";
+    ASSERT_EQ(sfInfo.samplerate, mConfig->samplingRate) << "Sample rate does not match";
+
+    mMp3Reader.close();
+    sf_close(outFileHandle);
+    if (decoderBuf) {
+        free(decoderBuf);
+        decoderBuf = nullptr;
+    }
+}
+
+INSTANTIATE_TEST_SUITE_P(Mp3DecoderTestAll, Mp3DecoderTest,
+                         ::testing::Values(("bbb_44100hz_2ch_128kbps_mp3_30sec.mp3"),
+                                           ("bbb_44100hz_2ch_128kbps_mp3_5mins.mp3"),
+                                           ("bbb_mp3_stereo_192kbps_48000hz.mp3")));
+
+int main(int argc, char **argv) {
+    gEnv = new Mp3DecoderTestEnvironment();
+    ::testing::AddGlobalTestEnvironment(gEnv);
+    ::testing::InitGoogleTest(&argc, argv);
+    int status = gEnv->initFromOptions(argc, argv);
+    if (status == 0) {
+        status = RUN_ALL_TESTS();
+        ALOGV("Test result = %d\n", status);
+    }
+    return status;
+}
diff --git a/media/libstagefright/codecs/mp3dec/test/Mp3DecoderTestEnvironment.h b/media/libstagefright/codecs/mp3dec/test/Mp3DecoderTestEnvironment.h
new file mode 100644
index 0000000..a54b34c
--- /dev/null
+++ b/media/libstagefright/codecs/mp3dec/test/Mp3DecoderTestEnvironment.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2019 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 __MP3DECODER_TEST_ENVIRONMENT_H__
+#define __MP3DECODER_TEST_ENVIRONMENT_H__
+
+#include <gtest/gtest.h>
+
+#include <getopt.h>
+
+using namespace std;
+
+class Mp3DecoderTestEnvironment : public ::testing::Environment {
+  public:
+    Mp3DecoderTestEnvironment() : res("/data/local/tmp/") {}
+
+    // Parses the command line arguments
+    int initFromOptions(int argc, char **argv);
+
+    void setRes(const char *_res) { res = _res; }
+
+    const string getRes() const { return res; }
+
+  private:
+    string res;
+};
+
+int Mp3DecoderTestEnvironment::initFromOptions(int argc, char **argv) {
+    static struct option options[] = {{"res", required_argument, 0, 'P'}, {0, 0, 0, 0}};
+
+    while (true) {
+        int index = 0;
+        int c = getopt_long(argc, argv, "P:", options, &index);
+        if (c == -1) {
+            break;
+        }
+
+        switch (c) {
+            case 'P':
+                setRes(optarg);
+                break;
+            default:
+                break;
+        }
+    }
+
+    if (optind < argc) {
+        fprintf(stderr,
+                "unrecognized option: %s\n\n"
+                "usage: %s <gtest options> <test options>\n\n"
+                "test options are:\n\n"
+                "-P, --path: Resource files directory location\n",
+                argv[optind ?: 1], argv[0]);
+        return 2;
+    }
+    return 0;
+}
+
+#endif  // __MP3DECODER_TEST_ENVIRONMENT_H__
diff --git a/media/libstagefright/codecs/mp3dec/test/README.md b/media/libstagefright/codecs/mp3dec/test/README.md
new file mode 100644
index 0000000..019239e
--- /dev/null
+++ b/media/libstagefright/codecs/mp3dec/test/README.md
@@ -0,0 +1,34 @@
+## Media Testing ##
+---
+#### Mp3Decoder :
+The Mp3Decoder Test Suite validates the mp3decoder available in libstagefright.
+
+Run the following steps to build the test suite:
+```
+m Mp3DecoderTest
+```
+
+The 32-bit binaries will be created in the following path : ${OUT}/data/nativetest/
+
+The 64-bit binaries will be created in the following path : ${OUT}/data/nativetest64/
+
+To test 64-bit binary push binaries from nativetest64.
+```
+adb push ${OUT}/data/nativetest64/Mp3DecoderTest/Mp3DecoderTest /data/local/tmp/
+```
+
+To test 32-bit binary push binaries from nativetest.
+```
+adb push ${OUT}/data/nativetest/Mp3DecoderTest/Mp3DecoderTest /data/local/tmp/
+```
+
+The resource file for the tests is taken from [here](https://drive.google.com/drive/folders/13cM4tAaVFrmr-zGFqaAzFBbKs75pnm9b). Push these files into device for testing.
+Download mp3 folder and push all the files in this folder to /data/local/tmp/ on the device.
+```
+adb push mp3/. /data/local/tmp/
+```
+
+usage: Mp3DecoderTest -P \<path_to_folder\>
+```
+adb shell /data/local/tmp/Mp3DecoderTest -P /data/local/tmp/
+```
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h b/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h
index 0843fea..a5de655 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h
@@ -31,12 +31,24 @@
 public:
     AudioPatch(const struct audio_patch *patch, uid_t uid);
 
+    audio_patch_handle_t getHandle() const { return mHandle; }
+
+    audio_patch_handle_t getAfHandle() const { return mAfPatchHandle; }
+
+    void setAfHandle(audio_patch_handle_t afHandle) { mAfPatchHandle = afHandle; }
+
+    uid_t getUid() const { return mUid; }
+
+    void setUid(uid_t uid) { mUid = uid; }
+
     void dump(String8 *dst, int spaces, int index) const;
 
-    audio_patch_handle_t mHandle;
     struct audio_patch mPatch;
+
+private:
+    const audio_patch_handle_t mHandle;
     uid_t mUid;
-    audio_patch_handle_t mAfPatchHandle;
+    audio_patch_handle_t mAfPatchHandle = AUDIO_PATCH_HANDLE_NONE;
 };
 
 class AudioPatchCollection : public DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> >
diff --git a/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
index 0d05a63..0c5d1d0 100644
--- a/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
@@ -183,13 +183,17 @@
 {
 public:
     SourceClientDescriptor(audio_port_handle_t portId, uid_t uid, audio_attributes_t attributes,
-                           const sp<AudioPatch>& patchDesc, const sp<DeviceDescriptor>& srcDevice,
+                           const struct audio_port_config &config,
+                           const sp<DeviceDescriptor>& srcDevice,
                            audio_stream_type_t stream, product_strategy_t strategy,
                            VolumeSource volumeSource);
+
     ~SourceClientDescriptor() override = default;
 
-    sp<AudioPatch> patchDesc() const { return mPatchDesc; }
-    sp<DeviceDescriptor> srcDevice() const { return mSrcDevice; };
+    audio_patch_handle_t getPatchHandle() const { return mPatchHandle; }
+    void setPatchHandle(audio_patch_handle_t patchHandle) { mPatchHandle = patchHandle; }
+
+    sp<DeviceDescriptor> srcDevice() const { return mSrcDevice; }
     wp<SwAudioOutputDescriptor> swOutput() const { return mSwOutput; }
     void setSwOutput(const sp<SwAudioOutputDescriptor>& swOutput);
     wp<HwAudioOutputDescriptor> hwOutput() const { return mHwOutput; }
@@ -199,7 +203,7 @@
     void dump(String8 *dst, int spaces, int index) const override;
 
  private:
-    const sp<AudioPatch> mPatchDesc;
+    audio_patch_handle_t mPatchHandle = AUDIO_PATCH_HANDLE_NONE;
     const sp<DeviceDescriptor> mSrcDevice;
     wp<SwAudioOutputDescriptor> mSwOutput;
     wp<HwAudioOutputDescriptor> mHwOutput;
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
index bf0cc94..d79110a 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
@@ -26,10 +26,9 @@
 namespace android {
 
 AudioPatch::AudioPatch(const struct audio_patch *patch, uid_t uid) :
-    mHandle(HandleGenerator<audio_patch_handle_t>::getNextHandle()),
     mPatch(*patch),
-    mUid(uid),
-    mAfPatchHandle(AUDIO_PATCH_HANDLE_NONE)
+    mHandle(HandleGenerator<audio_patch_handle_t>::getNextHandle()),
+    mUid(uid)
 {
 }
 
@@ -68,7 +67,7 @@
     add(handle, patch);
     ALOGV("addAudioPatch() handle %d af handle %d num_sources %d num_sinks %d source handle %d"
             "sink handle %d",
-          handle, patch->mAfPatchHandle, patch->mPatch.num_sources, patch->mPatch.num_sinks,
+          handle, patch->getAfHandle(), patch->mPatch.num_sources, patch->mPatch.num_sinks,
           patch->mPatch.sources[0].id, patch->mPatch.sinks[0].id);
     return NO_ERROR;
 }
@@ -81,7 +80,7 @@
         ALOGW("removeAudioPatch() patch %d not in", handle);
         return ALREADY_EXISTS;
     }
-    ALOGV("removeAudioPatch() handle %d af handle %d", handle, valueAt(index)->mAfPatchHandle);
+    ALOGV("removeAudioPatch() handle %d af handle %d", handle, valueAt(index)->getAfHandle());
     removeItemsAt(index);
     return NO_ERROR;
 }
@@ -123,7 +122,7 @@
         }
         if (patchesWritten < patchesMax) {
             patches[patchesWritten] = patch->mPatch;
-            patches[patchesWritten++].id = patch->mHandle;
+            patches[patchesWritten++].id = patch->getHandle();
         }
         (*num_patches)++;
         ALOGV("listAudioPatches() patch %zu num_sources %d num_sinks %d",
diff --git a/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
index 1dc7020..95822b9 100644
--- a/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
@@ -82,14 +82,13 @@
 }
 
 SourceClientDescriptor::SourceClientDescriptor(audio_port_handle_t portId, uid_t uid,
-         audio_attributes_t attributes, const sp<AudioPatch>& patchDesc,
+         audio_attributes_t attributes, const struct audio_port_config &config,
          const sp<DeviceDescriptor>& srcDevice, audio_stream_type_t stream,
          product_strategy_t strategy, VolumeSource volumeSource) :
     TrackClientDescriptor::TrackClientDescriptor(portId, uid, AUDIO_SESSION_NONE, attributes,
-        AUDIO_CONFIG_BASE_INITIALIZER, AUDIO_PORT_HANDLE_NONE,
+        {config.sample_rate, config.channel_mask, config.format}, AUDIO_PORT_HANDLE_NONE,
         stream, strategy, volumeSource, AUDIO_OUTPUT_FLAG_NONE, false,
-        {} /* Sources do not support secondary outputs*/),
-        mPatchDesc(patchDesc), mSrcDevice(srcDevice)
+        {} /* Sources do not support secondary outputs*/), mSrcDevice(srcDevice)
 {
 }
 
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index fbe5d0d..632c356 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -526,12 +526,12 @@
 
     // release existing RX patch if any
     if (mCallRxPatch != 0) {
-        mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
+        releaseAudioPatchInternal(mCallRxPatch->getHandle());
         mCallRxPatch.clear();
     }
     // release TX patch if any
     if (mCallTxPatch != 0) {
-        mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
+        releaseAudioPatchInternal(mCallTxPatch->getHandle());
         mCallTxPatch.clear();
     }
 
@@ -557,11 +557,9 @@
             ALOGE("updateCallRouting() no telephony Tx and/or RX device");
             return muteWaitMs;
         }
-        // do not create a patch (aka Sw Bridging) if Primary HW module has declared supporting a
-        // route between telephony RX to Sink device and Source device to telephony TX
-        const auto &primaryModule = telephonyRxModule;
-        createRxPatch = !primaryModule->supportsPatch(rxSourceDevice, rxDevices.itemAt(0));
-        createTxPatch = !primaryModule->supportsPatch(txSourceDevice, txSinkDevice);
+        // createAudioPatchInternal now supports both HW / SW bridging
+        createRxPatch = true;
+        createTxPatch = true;
     } else {
         // If the RX device is on the primary HW module, then use legacy routing method for
         // voice calls via setOutputDevice() on primary output.
@@ -585,6 +583,15 @@
         // assuming the device uses audio HAL V5.0 and above
     }
     if (createTxPatch) { // create TX path audio patch
+        // terminate active capture if on the same HW module as the call TX source device
+        // FIXME: would be better to refine to only inputs whose profile connects to the
+        // call TX device but this information is not in the audio patch and logic here must be
+        // symmetric to the one in startInput()
+        for (const auto& activeDesc : mInputs.getActiveInputs()) {
+            if (activeDesc->hasSameHwModuleAs(txSourceDevice)) {
+                closeActiveClients(activeDesc);
+            }
+        }
         mCallTxPatch = createTelephonyPatch(false /*isRx*/, txSourceDevice, delayMs);
     }
 
@@ -598,6 +605,8 @@
     if (device == nullptr) {
         return nullptr;
     }
+
+    // @TODO: still ignoring the address, or not dealing platform with mutliple telephony devices
     if (isRx) {
         patchBuilder.addSink(device).
                 addSource(mAvailableInputDevices.getDevice(
@@ -608,45 +617,15 @@
                     AUDIO_DEVICE_OUT_TELEPHONY_TX, String8(), AUDIO_FORMAT_DEFAULT));
     }
 
-    // @TODO: still ignoring the address, or not dealing platform with mutliple telephonydevices
-    const sp<DeviceDescriptor> outputDevice = isRx ?
-                device : mAvailableOutputDevices.getDevice(
-                    AUDIO_DEVICE_OUT_TELEPHONY_TX, String8(), AUDIO_FORMAT_DEFAULT);
-    SortedVector<audio_io_handle_t> outputs =
-            getOutputsForDevices(DeviceVector(outputDevice), mOutputs);
-    const audio_io_handle_t output = selectOutput(outputs);
-    // request to reuse existing output stream if one is already opened to reach the target device
-    if (output != AUDIO_IO_HANDLE_NONE) {
-        sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
-        ALOG_ASSERT(!outputDesc->isDuplicated(), "%s() %s device output %d is duplicated", __func__,
-                    outputDevice->toString().c_str(), output);
-        patchBuilder.addSource(outputDesc, { .stream = AUDIO_STREAM_PATCH });
+    audio_patch_handle_t patchHandle = AUDIO_PATCH_HANDLE_NONE;
+    status_t status =
+            createAudioPatchInternal(patchBuilder.patch(), &patchHandle, mUidCached, delayMs);
+    ssize_t index = mAudioPatches.indexOfKey(patchHandle);
+    if (status != NO_ERROR || index < 0) {
+        ALOGW("%s() error %d creating %s audio patch", __func__, status, isRx ? "RX" : "TX");
+        return nullptr;
     }
-
-    if (!isRx) {
-        // terminate active capture if on the same HW module as the call TX source device
-        // FIXME: would be better to refine to only inputs whose profile connects to the
-        // call TX device but this information is not in the audio patch and logic here must be
-        // symmetric to the one in startInput()
-        for (const auto& activeDesc : mInputs.getActiveInputs()) {
-            if (activeDesc->hasSameHwModuleAs(device)) {
-                closeActiveClients(activeDesc);
-            }
-        }
-    }
-
-    audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
-    status_t status = mpClientInterface->createAudioPatch(
-            patchBuilder.patch(), &afPatchHandle, delayMs);
-    ALOGW_IF(status != NO_ERROR,
-            "%s() error %d creating %s audio patch", __func__, status, isRx ? "RX" : "TX");
-    sp<AudioPatch> audioPatch;
-    if (status == NO_ERROR) {
-        audioPatch = new AudioPatch(patchBuilder.patch(), mUidCached);
-        audioPatch->mAfPatchHandle = afPatchHandle;
-        audioPatch->mUid = mUidCached;
-    }
-    return audioPatch;
+    return mAudioPatches.valueAt(index);
 }
 
 bool AudioPolicyManager::isDeviceOfModule(
@@ -729,11 +708,11 @@
             updateCallRouting(rxDevices, delayMs);
         } else if (oldState == AUDIO_MODE_IN_CALL) {
             if (mCallRxPatch != 0) {
-                mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
+                releaseAudioPatchInternal(mCallRxPatch->getHandle());
                 mCallRxPatch.clear();
             }
             if (mCallTxPatch != 0) {
-                mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
+                releaseAudioPatchInternal(mCallTxPatch->getHandle());
                 mCallTxPatch.clear();
             }
             setOutputDevices(mPrimaryOutput, rxDevices, force, 0);
@@ -1209,7 +1188,7 @@
                         devices.containsDeviceWithType(sink->ext.device.type) &&
                         (address.isEmpty() || strncmp(sink->ext.device.address, address.string(),
                                 AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0)) {
-                    releaseAudioPatch(patch->mHandle, mUidCached);
+                    releaseAudioPatch(patch->getHandle(), mUidCached);
                     break;
                 }
             }
@@ -1296,7 +1275,7 @@
                 const struct audio_port_config *source = &patch->mPatch.sources[j];
                 if (source->type == AUDIO_PORT_TYPE_DEVICE &&
                         source->ext.device.hw_module == msdModule->getHandle()) {
-                    msdPatches.addAudioPatch(patch->mHandle, patch);
+                    msdPatches.addAudioPatch(patch->getHandle(), patch);
                 }
             }
         }
@@ -1426,7 +1405,7 @@
         if (audio_patches_are_equal(&currentPatch->mPatch, patch)) {
             return NO_ERROR;
         }
-        releaseAudioPatch(currentPatch->mHandle, mUidCached);
+        releaseAudioPatch(currentPatch->getHandle(), mUidCached);
     }
     status_t status = installPatch(__func__, -1 /*index*/, nullptr /*patchHandle*/,
             patch, 0 /*delayMs*/, mUidCached, nullptr /*patchDescPtr*/);
@@ -3391,16 +3370,16 @@
     return BAD_VALUE;
 }
 
-status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch,
-                                               audio_patch_handle_t *handle,
-                                               uid_t uid)
+status_t AudioPolicyManager::createAudioPatchInternal(const struct audio_patch *patch,
+                                                      audio_patch_handle_t *handle,
+                                                      uid_t uid, uint32_t delayMs,
+                                                      const sp<SourceClientDescriptor>& sourceDesc)
 {
-    ALOGV("createAudioPatch()");
-
+    ALOGV("%s", __func__);
     if (handle == NULL || patch == NULL) {
         return BAD_VALUE;
     }
-    ALOGV("createAudioPatch() num sources %d num sinks %d", patch->num_sources, patch->num_sinks);
+    ALOGV("%s num sources %d num sinks %d", __func__, patch->num_sources, patch->num_sinks);
 
     if (!audio_patch_is_valid(patch)) {
         return BAD_VALUE;
@@ -3422,22 +3401,22 @@
     sp<AudioPatch> patchDesc;
     ssize_t index = mAudioPatches.indexOfKey(*handle);
 
-    ALOGV("createAudioPatch source id %d role %d type %d", patch->sources[0].id,
-                                                           patch->sources[0].role,
-                                                           patch->sources[0].type);
+    ALOGV("%s source id %d role %d type %d", __func__, patch->sources[0].id,
+                                                       patch->sources[0].role,
+                                                       patch->sources[0].type);
 #if LOG_NDEBUG == 0
     for (size_t i = 0; i < patch->num_sinks; i++) {
-        ALOGV("createAudioPatch sink %zu: id %d role %d type %d", i, patch->sinks[i].id,
-                                                             patch->sinks[i].role,
-                                                             patch->sinks[i].type);
+        ALOGV("%s sink %zu: id %d role %d type %d", __func__ ,i, patch->sinks[i].id,
+                                                                 patch->sinks[i].role,
+                                                                 patch->sinks[i].type);
     }
 #endif
 
     if (index >= 0) {
         patchDesc = mAudioPatches.valueAt(index);
-        ALOGV("createAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
-                                                                  mUidCached, patchDesc->mUid, uid);
-        if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
+        ALOGV("%s mUidCached %d patchDesc->mUid %d uid %d",
+              __func__, mUidCached, patchDesc->getUid(), uid);
+        if (patchDesc->getUid() != mUidCached && uid != patchDesc->getUid()) {
             return INVALID_OPERATION;
         }
     } else {
@@ -3447,15 +3426,15 @@
     if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
         sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id);
         if (outputDesc == NULL) {
-            ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id);
+            ALOGV("%s output not found for id %d", __func__, patch->sources[0].id);
             return BAD_VALUE;
         }
         ALOG_ASSERT(!outputDesc->isDuplicated(),"duplicated output %d in source in ports",
                                                 outputDesc->mIoHandle);
         if (patchDesc != 0) {
             if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) {
-                ALOGV("createAudioPatch() source id differs for patch current id %d new id %d",
-                                          patchDesc->mPatch.sources[0].id, patch->sources[0].id);
+                ALOGV("%s source id differs for patch current id %d new id %d",
+                      __func__, patchDesc->mPatch.sources[0].id, patch->sources[0].id);
                 return BAD_VALUE;
             }
         }
@@ -3464,13 +3443,13 @@
             // Only support mix to devices connection
             // TODO add support for mix to mix connection
             if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
-                ALOGV("createAudioPatch() source mix but sink is not a device");
+                ALOGV("%s source mix but sink is not a device", __func__);
                 return INVALID_OPERATION;
             }
             sp<DeviceDescriptor> devDesc =
                     mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
             if (devDesc == 0) {
-                ALOGV("createAudioPatch() out device not found for id %d", patch->sinks[i].id);
+                ALOGV("%s out device not found for id %d", __func__, patch->sinks[i].id);
                 return BAD_VALUE;
             }
 
@@ -3482,8 +3461,7 @@
                                                            patch->sources[0].channel_mask,
                                                            NULL,  // updatedChannelMask
                                                            AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) {
-                ALOGV("createAudioPatch() profile not supported for device %08x",
-                        devDesc->type());
+                ALOGV("%s profile not supported for device %08x", __func__, devDesc->type());
                 return INVALID_OPERATION;
             }
             devices.add(devDesc);
@@ -3493,19 +3471,19 @@
         }
 
         // TODO: reconfigure output format and channels here
-        ALOGV("createAudioPatch() setting device %s on output %d",
-              dumpDeviceTypes(devices.types()).c_str(), outputDesc->mIoHandle);
+        ALOGV("%s setting device %s on output %d",
+              __func__, dumpDeviceTypes(devices.types()).c_str(), outputDesc->mIoHandle);
         setOutputDevices(outputDesc, devices, true, 0, handle);
         index = mAudioPatches.indexOfKey(*handle);
         if (index >= 0) {
             if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
-                ALOGW("createAudioPatch() setOutputDevice() did not reuse the patch provided");
+                ALOGW("%s setOutputDevice() did not reuse the patch provided", __func__);
             }
             patchDesc = mAudioPatches.valueAt(index);
-            patchDesc->mUid = uid;
-            ALOGV("createAudioPatch() success");
+            patchDesc->setUid(uid);
+            ALOGV("%s success", __func__);
         } else {
-            ALOGW("createAudioPatch() setOutputDevice() failed to create a patch");
+            ALOGW("%s setOutputDevice() failed to create a patch", __func__);
             return INVALID_OPERATION;
         }
     } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
@@ -3544,19 +3522,19 @@
                 return INVALID_OPERATION;
             }
             // TODO: reconfigure output format and channels here
-            ALOGV("%s() setting device %s on output %d", __func__,
+            ALOGV("%s setting device %s on output %d", __func__,
                   device->toString().c_str(), inputDesc->mIoHandle);
             setInputDevice(inputDesc->mIoHandle, device, true, handle);
             index = mAudioPatches.indexOfKey(*handle);
             if (index >= 0) {
                 if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
-                    ALOGW("createAudioPatch() setInputDevice() did not reuse the patch provided");
+                    ALOGW("%s setInputDevice() did not reuse the patch provided", __func__);
                 }
                 patchDesc = mAudioPatches.valueAt(index);
-                patchDesc->mUid = uid;
-                ALOGV("createAudioPatch() success");
+                patchDesc->setUid(uid);
+                ALOGV("%s success", __func__);
             } else {
-                ALOGW("createAudioPatch() setInputDevice() failed to create a patch");
+                ALOGW("%s setInputDevice() failed to create a patch", __func__);
                 return INVALID_OPERATION;
             }
         } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
@@ -3574,55 +3552,96 @@
 
             //update source and sink with our own data as the data passed in the patch may
             // be incomplete.
-            struct audio_patch newPatch = *patch;
-            srcDevice->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]);
+            PatchBuilder patchBuilder;
+            audio_port_config sourcePortConfig = {};
+            srcDevice->toAudioPortConfig(&sourcePortConfig, &patch->sources[0]);
+            patchBuilder.addSource(sourcePortConfig);
 
             for (size_t i = 0; i < patch->num_sinks; i++) {
                 if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
-                    ALOGV("createAudioPatch() source device but one sink is not a device");
+                    ALOGV("%s source device but one sink is not a device", __func__);
                     return INVALID_OPERATION;
                 }
-
                 sp<DeviceDescriptor> sinkDevice =
                         mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
                 if (sinkDevice == 0) {
                     return BAD_VALUE;
                 }
-                sinkDevice->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]);
+                audio_port_config sinkPortConfig = {};
+                sinkDevice->toAudioPortConfig(&sinkPortConfig, &patch->sinks[i]);
+                patchBuilder.addSink(sinkPortConfig);
 
                 // create a software bridge in PatchPanel if:
                 // - source and sink devices are on different HW modules OR
                 // - audio HAL version is < 3.0
                 // - audio HAL version is >= 3.0 but no route has been declared between devices
+                // - called from startAudioSource (aka sourceDesc != nullptr) and source device does
+                //   not have a gain controller
                 if (!srcDevice->hasSameHwModuleAs(sinkDevice) ||
                         (srcDevice->getModuleVersionMajor() < 3) ||
-                        !srcDevice->getModule()->supportsPatch(srcDevice, sinkDevice)) {
+                        !srcDevice->getModule()->supportsPatch(srcDevice, sinkDevice) ||
+                        (sourceDesc != nullptr &&
+                         srcDevice->getAudioPort()->getGains().size() == 0)) {
                     // support only one sink device for now to simplify output selection logic
                     if (patch->num_sinks > 1) {
                         return INVALID_OPERATION;
                     }
-                    SortedVector<audio_io_handle_t> outputs =
-                            getOutputsForDevices(DeviceVector(sinkDevice), mOutputs);
-                    // if the sink device is reachable via an opened output stream, request to go via
-                    // this output stream by adding a second source to the patch description
-                    const audio_io_handle_t output = selectOutput(outputs);
-                    if (output != AUDIO_IO_HANDLE_NONE) {
-                        sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
-                        if (outputDesc->isDuplicated()) {
+                    audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+                    if (sourceDesc != nullptr) {
+                        // take care of dynamic routing for SwOutput selection,
+                        audio_attributes_t attributes = sourceDesc->attributes();
+                        audio_stream_type_t stream = sourceDesc->stream();
+                        audio_attributes_t resultAttr;
+                        audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+                        config.sample_rate = sourceDesc->config().sample_rate;
+                        config.channel_mask = sourceDesc->config().channel_mask;
+                        config.format = sourceDesc->config().format;
+                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
+                        audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+                        bool isRequestedDeviceForExclusiveUse = false;
+                        std::vector<sp<SwAudioOutputDescriptor>> secondaryOutputs;
+                        getOutputForAttrInt(&resultAttr, &output, AUDIO_SESSION_NONE, &attributes,
+                                            &stream, sourceDesc->uid(), &config, &flags,
+                                            &selectedDeviceId, &isRequestedDeviceForExclusiveUse,
+                                            &secondaryOutputs);
+                        if (output == AUDIO_IO_HANDLE_NONE) {
+                            ALOGV("%s no output for device %s",
+                                  __FUNCTION__, sinkDevice->toString().c_str());
                             return INVALID_OPERATION;
                         }
-                        outputDesc->toAudioPortConfig(&newPatch.sources[1], &patch->sources[0]);
-                        newPatch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH;
-                        newPatch.num_sources = 2;
+                    } else {
+                        SortedVector<audio_io_handle_t> outputs =
+                                getOutputsForDevices(DeviceVector(sinkDevice), mOutputs);
+                        // if the sink device is reachable via an opened output stream, request to
+                        // go via this output stream by adding a second source to the patch
+                        // description
+                        output = selectOutput(outputs);
+                    }
+                    if (output != AUDIO_IO_HANDLE_NONE) {
+                        sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+                        if (outputDesc->isDuplicated()) {
+                            ALOGV("%s output for device %s is duplicated",
+                                  __FUNCTION__, sinkDevice->toString().c_str());
+                            return INVALID_OPERATION;
+                        }
+                        audio_port_config srcMixPortConfig = {};
+                        outputDesc->toAudioPortConfig(&srcMixPortConfig, &patch->sources[0]);
+                        if (sourceDesc != nullptr) {
+                            sourceDesc->setSwOutput(outputDesc);
+                        }
+                        // for volume control, we may need a valid stream
+                        srcMixPortConfig.ext.mix.usecase.stream = sourceDesc != nullptr ?
+                                    sourceDesc->stream() : AUDIO_STREAM_PATCH;
+                        patchBuilder.addSource(srcMixPortConfig);
                     }
                 }
             }
             // TODO: check from routing capabilities in config file and other conflicting patches
 
-            status_t status = installPatch(__func__, index, handle, &newPatch, 0, uid, &patchDesc);
+            status_t status = installPatch(
+                        __func__, index, handle, patchBuilder.patch(), delayMs, uid, &patchDesc);
             if (status != NO_ERROR) {
-                ALOGW("createAudioPatch() patch panel could not connect device patch, error %d",
-                status);
+                ALOGW("%s patch panel could not connect device patch, error %d", __func__, status);
                 return INVALID_OPERATION;
             }
         } else {
@@ -3645,18 +3664,29 @@
         return BAD_VALUE;
     }
     sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
-    ALOGV("releaseAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
-          mUidCached, patchDesc->mUid, uid);
-    if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
+    ALOGV("%s() mUidCached %d patchDesc->mUid %d uid %d",
+          __func__, mUidCached, patchDesc->getUid(), uid);
+    if (patchDesc->getUid() != mUidCached && uid != patchDesc->getUid()) {
         return INVALID_OPERATION;
     }
+    return releaseAudioPatchInternal(handle);
+}
 
+status_t AudioPolicyManager::releaseAudioPatchInternal(audio_patch_handle_t handle,
+                                                       uint32_t delayMs)
+{
+    ALOGV("%s patch %d", __func__, handle);
+    if (mAudioPatches.indexOfKey(handle) < 0) {
+        ALOGE("%s: no patch found with handle=%d", __func__, handle);
+        return BAD_VALUE;
+    }
+    sp<AudioPatch> patchDesc = mAudioPatches.valueFor(handle);
     struct audio_patch *patch = &patchDesc->mPatch;
-    patchDesc->mUid = mUidCached;
+    patchDesc->setUid(mUidCached);
     if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
         sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id);
         if (outputDesc == NULL) {
-            ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id);
+            ALOGV("%s output not found for id %d", __func__, patch->sources[0].id);
             return BAD_VALUE;
         }
 
@@ -3669,7 +3699,7 @@
         if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
             sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(patch->sinks[0].id);
             if (inputDesc == NULL) {
-                ALOGV("releaseAudioPatch() input not found for id %d", patch->sinks[0].id);
+                ALOGV("%s input not found for id %d", __func__, patch->sinks[0].id);
                 return BAD_VALUE;
             }
             setInputDevice(inputDesc->mIoHandle,
@@ -3677,10 +3707,11 @@
                            true,
                            NULL);
         } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
-            status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
-            ALOGV("releaseAudioPatch() patch panel returned %d patchHandle %d",
-                                                              status, patchDesc->mAfPatchHandle);
-            removeAudioPatch(patchDesc->mHandle);
+            status_t status =
+                    mpClientInterface->releaseAudioPatch(patchDesc->getAfHandle(), delayMs);
+            ALOGV("%s patch panel returned %d patchHandle %d",
+                  __func__, status, patchDesc->getAfHandle());
+            removeAudioPatch(patchDesc->getHandle());
             nextAudioPortGeneration();
             mpClientInterface->onAudioPatchListUpdate();
         } else {
@@ -3778,7 +3809,7 @@
 {
     for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--)  {
         sp<AudioPatch> patchDesc = mAudioPatches.valueAt(i);
-        if (patchDesc->mUid == uid) {
+        if (patchDesc->getUid() == uid) {
             releaseAudioPatch(mAudioPatches.keyAt(i), uid);
         }
     }
@@ -3914,11 +3945,8 @@
 
     *portId = PolicyAudioPort::getNextUniqueId();
 
-    struct audio_patch dummyPatch = {};
-    sp<AudioPatch> patchDesc = new AudioPatch(&dummyPatch, uid);
-
     sp<SourceClientDescriptor> sourceDesc =
-        new SourceClientDescriptor(*portId, uid, *attributes, patchDesc, srcDevice,
+        new SourceClientDescriptor(*portId, uid, *attributes, *source, srcDevice,
                                    mEngine->getStreamTypeForAttributes(*attributes),
                                    mEngine->getProductStrategyForAttributes(*attributes),
                                    toVolumeSource(*attributes));
@@ -3938,7 +3966,6 @@
     disconnectAudioSource(sourceDesc);
 
     audio_attributes_t attributes = sourceDesc->attributes();
-    audio_stream_type_t stream = sourceDesc->stream();
     sp<DeviceDescriptor> srcDevice = sourceDesc->srcDevice();
 
     DeviceVector sinkDevices =
@@ -3947,92 +3974,55 @@
     sp<DeviceDescriptor> sinkDevice = sinkDevices.itemAt(0);
     ALOG_ASSERT(mAvailableOutputDevices.contains(sinkDevice), "%s: Device %s not available",
                 __FUNCTION__, sinkDevice->toString().c_str());
-
-    audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
-
-    if (srcDevice->hasSameHwModuleAs(sinkDevice) &&
-            srcDevice->getModuleVersionMajor() >= 3 &&
-            sinkDevice->getModule()->supportsPatch(srcDevice, sinkDevice) &&
-            srcDevice->getAudioPort()->getGains().size() > 0) {
-        ALOGV("%s Device to Device route supported by >=3.0 HAL", __FUNCTION__);
-        // TODO: may explicitly specify whether we should use HW or SW patch
-        //   create patch between src device and output device
-        //   create Hwoutput and add to mHwOutputs
-    } else {
-        audio_attributes_t resultAttr;
-        audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
-        audio_config_t config = AUDIO_CONFIG_INITIALIZER;
-        config.sample_rate = sourceDesc->config().sample_rate;
-        config.channel_mask = sourceDesc->config().channel_mask;
-        config.format = sourceDesc->config().format;
-        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
-        audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
-        bool isRequestedDeviceForExclusiveUse = false;
-        std::vector<sp<SwAudioOutputDescriptor>> secondaryOutputs;
-        getOutputForAttrInt(&resultAttr, &output, AUDIO_SESSION_NONE,
-                &attributes, &stream, sourceDesc->uid(), &config, &flags,
-                &selectedDeviceId, &isRequestedDeviceForExclusiveUse,
-                &secondaryOutputs);
-        if (output == AUDIO_IO_HANDLE_NONE) {
-            ALOGV("%s no output for device %s",
-                  __FUNCTION__, dumpDeviceTypes(sinkDevices.types()).c_str());
-            return INVALID_OPERATION;
-        }
-        sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
-        if (outputDesc->isDuplicated()) {
-            ALOGV("%s output for device %s is duplicated",
-                  __FUNCTION__, dumpDeviceTypes(sinkDevices.types()).c_str());
-            return INVALID_OPERATION;
-        }
-        status_t status = outputDesc->start();
+    PatchBuilder patchBuilder;
+    patchBuilder.addSink(sinkDevice).addSource(srcDevice);
+    audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE;
+    status_t status =
+            createAudioPatchInternal(patchBuilder.patch(), &handle, mUidCached, 0, sourceDesc);
+    if (status != NO_ERROR || mAudioPatches.indexOfKey(handle) < 0) {
+        ALOGW("%s patch panel could not connect device patch, error %d", __func__, status);
+        return INVALID_OPERATION;
+    }
+    sourceDesc->setPatchHandle(handle);
+    // SW Bridge? (@todo: HW bridge, keep track of HwOutput for device selection "reconsideration")
+    sp<SwAudioOutputDescriptor> swOutput = sourceDesc->swOutput().promote();
+    if (swOutput != 0) {
+        status = swOutput->start();
         if (status != NO_ERROR) {
-            return status;
+            goto FailureSourceAdded;
         }
-
-        // create a special patch with no sink and two sources:
-        // - the second source indicates to PatchPanel through which output mix this patch should
-        // be connected as well as the stream type for volume control
-        // - the sink is defined by whatever output device is currently selected for the output
-        // though which this patch is routed.
-        PatchBuilder patchBuilder;
-        patchBuilder.addSource(srcDevice).addSource(outputDesc, { .stream = stream });
-        status = mpClientInterface->createAudioPatch(patchBuilder.patch(),
-                                                              &afPatchHandle,
-                                                              0);
-        ALOGV("%s patch panel returned %d patchHandle %d", __FUNCTION__,
-                                                              status, afPatchHandle);
-        sourceDesc->patchDesc()->mPatch = *patchBuilder.patch();
-        if (status != NO_ERROR) {
-            ALOGW("%s patch panel could not connect device patch, error %d",
-                  __FUNCTION__, status);
-            return INVALID_OPERATION;
-        }
-
-        if (outputDesc->getClient(sourceDesc->portId()) != nullptr) {
+        if (swOutput->getClient(sourceDesc->portId()) != nullptr) {
             ALOGW("%s source portId has already been attached to outputDesc", __func__);
-            return INVALID_OPERATION;
+            goto FailureReleasePatch;
         }
-        outputDesc->addClient(sourceDesc);
-
+        swOutput->addClient(sourceDesc);
         uint32_t delayMs = 0;
-        status = startSource(outputDesc, sourceDesc, &delayMs);
-
+        status = startSource(swOutput, sourceDesc, &delayMs);
         if (status != NO_ERROR) {
-            mpClientInterface->releaseAudioPatch(sourceDesc->patchDesc()->mAfPatchHandle, 0);
-            outputDesc->removeClient(sourceDesc->portId());
-            outputDesc->stop();
-            return status;
+            ALOGW("%s failed to start source, error %d", __FUNCTION__, status);
+            goto FailureSourceActive;
         }
-        sourceDesc->setSwOutput(outputDesc);
         if (delayMs != 0) {
             usleep(delayMs * 1000);
         }
+    } else {
+        sp<HwAudioOutputDescriptor> hwOutputDesc = sourceDesc->hwOutput().promote();
+        if (hwOutputDesc != 0) {
+          //   create Hwoutput and add to mHwOutputs
+        } else {
+            ALOGW("%s source has neither SW nor HW output", __FUNCTION__);
+        }
     }
-
-    sourceDesc->patchDesc()->mAfPatchHandle = afPatchHandle;
-    addAudioPatch(sourceDesc->patchDesc()->mHandle, sourceDesc->patchDesc());
-
     return NO_ERROR;
+
+FailureSourceActive:
+    swOutput->stop();
+    releaseOutput(sourceDesc->portId());
+FailureSourceAdded:
+    sourceDesc->setSwOutput(nullptr);
+FailureReleasePatch:
+    releaseAudioPatchInternal(handle);
+    return INVALID_OPERATION;
 }
 
 status_t AudioPolicyManager::stopAudioSource(audio_port_handle_t portId)
@@ -4270,33 +4260,22 @@
 status_t AudioPolicyManager::disconnectAudioSource(const sp<SourceClientDescriptor>& sourceDesc)
 {
     ALOGV("%s port Id %d", __FUNCTION__, sourceDesc->portId());
-
-    sp<AudioPatch> patchDesc = mAudioPatches.valueFor(sourceDesc->patchDesc()->mHandle);
-    if (patchDesc == 0) {
-        ALOGW("%s source has no patch with handle %d", __FUNCTION__,
-              sourceDesc->patchDesc()->mHandle);
-        return BAD_VALUE;
-    }
-    removeAudioPatch(sourceDesc->patchDesc()->mHandle);
-
-    sp<SwAudioOutputDescriptor> swOutputDesc = sourceDesc->swOutput().promote();
-    if (swOutputDesc != 0) {
-        status_t status = stopSource(swOutputDesc, sourceDesc);
+    sp<SwAudioOutputDescriptor> swOutput = sourceDesc->swOutput().promote();
+    if (swOutput != 0) {
+        status_t status = stopSource(swOutput, sourceDesc);
         if (status == NO_ERROR) {
-            swOutputDesc->stop();
+            swOutput->stop();
         }
-        swOutputDesc->removeClient(sourceDesc->portId());
-        mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+        releaseOutput(sourceDesc->portId());
     } else {
         sp<HwAudioOutputDescriptor> hwOutputDesc = sourceDesc->hwOutput().promote();
         if (hwOutputDesc != 0) {
-          //   release patch between src device and output device
           //   close Hwoutput and remove from mHwOutputs
         } else {
             ALOGW("%s source has neither SW nor HW output", __FUNCTION__);
         }
     }
-    return NO_ERROR;
+    return releaseAudioPatchInternal(sourceDesc->getPatchHandle());
 }
 
 sp<SourceClientDescriptor> AudioPolicyManager::getSourceForAttributesOnOutput(
@@ -5004,7 +4983,8 @@
     ssize_t index = mAudioPatches.indexOfKey(closingOutput->getPatchHandle());
     if (index >= 0) {
         sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
-        (void) /*status_t status*/ mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+        (void) /*status_t status*/ mpClientInterface->releaseAudioPatch(
+                    patchDesc->getAfHandle(), 0);
         mAudioPatches.removeItemsAt(index);
         mpClientInterface->onAudioPatchListUpdate();
     }
@@ -5052,7 +5032,8 @@
     ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
     if (index >= 0) {
         sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
-        (void) /*status_t status*/ mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+        (void) /*status_t status*/ mpClientInterface->releaseAudioPatch(
+                    patchDesc->getAfHandle(), 0);
         mAudioPatches.removeItemsAt(index);
         mpClientInterface->onAudioPatchListUpdate();
     }
@@ -5265,7 +5246,7 @@
     ssize_t index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle());
     if (index >= 0) {
         sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
-        if (patchDesc->mUid != mUidCached) {
+        if (patchDesc->getUid() != mUidCached) {
             ALOGV("%s device %s forced by patch %d", __func__,
                   outputDesc->devices().toString().c_str(), outputDesc->getPatchHandle());
             return  outputDesc->devices();
@@ -5316,7 +5297,7 @@
     ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
     if (index >= 0) {
         sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
-        if (patchDesc->mUid != mUidCached) {
+        if (patchDesc->getUid() != mUidCached) {
             ALOGV("getNewInputDevice() device %s forced by patch %d",
                   inputDesc->getDevice()->toString().c_str(), inputDesc->getPatchHandle());
             return inputDesc->getDevice();
@@ -5643,10 +5624,10 @@
         return INVALID_OPERATION;
     }
     sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index);
-    status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, delayMs);
+    status_t status = mpClientInterface->releaseAudioPatch(patchDesc->getAfHandle(), delayMs);
     ALOGV("resetOutputDevice() releaseAudioPatch returned %d", status);
     outputDesc->setPatchHandle(AUDIO_PATCH_HANDLE_NONE);
-    removeAudioPatch(patchDesc->mHandle);
+    removeAudioPatch(patchDesc->getHandle());
     nextAudioPortGeneration();
     mpClientInterface->onAudioPatchListUpdate();
     return status;
@@ -5696,10 +5677,10 @@
         return INVALID_OPERATION;
     }
     sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index);
-    status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+    status_t status = mpClientInterface->releaseAudioPatch(patchDesc->getAfHandle(), 0);
     ALOGV("resetInputDevice() releaseAudioPatch returned %d", status);
     inputDesc->setPatchHandle(AUDIO_PATCH_HANDLE_NONE);
-    removeAudioPatch(patchDesc->mHandle);
+    removeAudioPatch(patchDesc->getHandle());
     nextAudioPortGeneration();
     mpClientInterface->onAudioPatchListUpdate();
     return status;
@@ -6112,8 +6093,8 @@
             }
         }
         if (release) {
-            ALOGV("%s releasing patch %u", __FUNCTION__, patchDesc->mHandle);
-            releaseAudioPatch(patchDesc->mHandle, patchDesc->mUid);
+            ALOGV("%s releasing patch %u", __FUNCTION__, patchDesc->getHandle());
+            releaseAudioPatch(patchDesc->getHandle(), patchDesc->getUid());
         }
     }
 
@@ -6288,7 +6269,7 @@
     status_t status = installPatch(
             caller, index, patchHandle, patch, delayMs, mUidCached, &patchDesc);
     if (status == NO_ERROR) {
-        ioDescriptor->setPatchHandle(patchDesc->mHandle);
+        ioDescriptor->setPatchHandle(patchDesc->getHandle());
     }
     return status;
 }
@@ -6305,7 +6286,7 @@
     audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
     if (index >= 0) {
         patchDesc = mAudioPatches.valueAt(index);
-        afPatchHandle = patchDesc->mAfPatchHandle;
+        afPatchHandle = patchDesc->getAfHandle();
     }
 
     status_t status = mpClientInterface->createAudioPatch(patch, &afPatchHandle, delayMs);
@@ -6314,13 +6295,13 @@
     if (status == NO_ERROR) {
         if (index < 0) {
             patchDesc = new AudioPatch(patch, uid);
-            addAudioPatch(patchDesc->mHandle, patchDesc);
+            addAudioPatch(patchDesc->getHandle(), patchDesc);
         } else {
             patchDesc->mPatch = *patch;
         }
-        patchDesc->mAfPatchHandle = afPatchHandle;
+        patchDesc->setAfHandle(afPatchHandle);
         if (patchHandle) {
-            *patchHandle = patchDesc->mHandle;
+            *patchHandle = patchDesc->getHandle();
         }
         nextAudioPortGeneration();
         mpClientInterface->onAudioPatchListUpdate();
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index d516d7b..634eb31 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -233,7 +233,10 @@
         virtual status_t getAudioPort(struct audio_port *port);
         virtual status_t createAudioPatch(const struct audio_patch *patch,
                                            audio_patch_handle_t *handle,
-                                           uid_t uid);
+                                           uid_t uid) {
+            return createAudioPatchInternal(patch, handle, uid);
+        }
+
         virtual status_t releaseAudioPatch(audio_patch_handle_t handle,
                                               uid_t uid);
         virtual status_t listAudioPatches(unsigned int *num_patches,
@@ -868,6 +871,29 @@
             param.addInt(String8(AudioParameter::keyMonoOutput), (int)mMasterMono);
             mpClientInterface->setParameters(output, param.toString());
         }
+
+        /**
+         * @brief createAudioPatchInternal internal function to manage audio patch creation
+         * @param[in] patch structure containing sink and source ports configuration
+         * @param[out] handle patch handle to be provided if patch installed correctly
+         * @param[in] uid of the client
+         * @param[in] delayMs if required
+         * @param[in] sourceDesc [optional] in case of external source, source client to be
+         * configured by the patch, i.e. assigning an Output (HW or SW)
+         * @return NO_ERROR if patch installed correclty, error code otherwise.
+         */
+        status_t createAudioPatchInternal(const struct audio_patch *patch,
+                                          audio_patch_handle_t *handle,
+                                          uid_t uid, uint32_t delayMs = 0,
+                                          const sp<SourceClientDescriptor>& sourceDesc = nullptr);
+        /**
+         * @brief releaseAudioPatchInternal internal function to remove an audio patch
+         * @param[in] handle of the patch to be removed
+         * @param[in] delayMs if required
+         * @return NO_ERROR if patch removed correclty, error code otherwise.
+         */
+        status_t releaseAudioPatchInternal(audio_patch_handle_t handle, uint32_t delayMs = 0);
+
         status_t installPatch(const char *caller,
                 audio_patch_handle_t *patchHandle,
                 AudioIODescriptorInterface *ioDescriptor,