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Commit 809b4b7a authored by Mingming Yin's avatar Mingming Yin Committed by Steve Kondik
Browse files

audio: Add support for tunnel mode recording 

- Add support for tunnel mode recording.

Change-Id: I95cdfff729affd784141487521c9f2f714221d11
parent 47c98e7e

include/media/stagefright/AudioSource.h

+12 −0
Original line number Diff line number Diff line 
/*

 * Copyright (C) 2009 The Android Open Source Project

 * Copyright (c) 2013, The Linux Foundation. All rights reserved.

 *

 * Licensed under the Apache License, Version 2.0 (the "License");

 * you may not use this file except in compliance with the License.
@@ -104,6 +105,17 @@ private: 


    AudioSource(const AudioSource &);

    AudioSource &operator=(const AudioSource &);



    //additions for tunnel source

public:

    AudioSource(

        audio_source_t inputSource, const sp<MetaData>& meta );



private:

    audio_format_t mFormat;

    String8 mMime;

    int32_t mMaxBufferSize;

    int64_t bufferDurationUs( ssize_t n );

};



}  // namespace android

include/media/stagefright/MediaDefs.h

+20 −0
Original line number Diff line number Diff line 
/*

 * Copyright (C) 2009 The Android Open Source Project

 * Copyright (c) 2013, The Linux Foundation. All rights reserved.

 *

 * Licensed under the Apache License, Version 2.0 (the "License");

 * you may not use this file except in compliance with the License.
@@ -43,6 +44,8 @@ extern const char *MEDIA_MIMETYPE_AUDIO_G711_MLAW; 
extern const char *MEDIA_MIMETYPE_AUDIO_RAW;

extern const char *MEDIA_MIMETYPE_AUDIO_FLAC;

extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS;

extern const char *MEDIA_MIMETYPE_AUDIO_DTS;

extern const char *MEDIA_MIMETYPE_AUDIO_DTS_LBR;



extern const char *MEDIA_MIMETYPE_CONTAINER_MPEG4;

extern const char *MEDIA_MIMETYPE_CONTAINER_WAV;
@@ -57,6 +60,23 @@ extern const char *MEDIA_MIMETYPE_CONTAINER_WVM; 
extern const char *MEDIA_MIMETYPE_TEXT_3GPP;

extern const char *MEDIA_MIMETYPE_TEXT_SUBRIP;



extern const char *MEDIA_MIMETYPE_AUDIO_EVRC;



extern const char *MEDIA_MIMETYPE_VIDEO_WMV;

extern const char *MEDIA_MIMETYPE_AUDIO_WMA;

extern const char *MEDIA_MIMETYPE_CONTAINER_ASF;

extern const char *MEDIA_MIMETYPE_VIDEO_DIVX;

extern const char *MEDIA_MIMETYPE_AUDIO_AC3;

extern const char *MEDIA_MIMETYPE_AUDIO_EAC3;

extern const char *MEDIA_MIMETYPE_CONTAINER_AAC;

extern const char *MEDIA_MIMETYPE_CONTAINER_QCP;

extern const char *MEDIA_MIMETYPE_VIDEO_DIVX311;

extern const char *MEDIA_MIMETYPE_VIDEO_DIVX4;

extern const char *MEDIA_MIMETYPE_CONTAINER_MPEG2;



extern const char *MEDIA_MIMETYPE_CONTAINER_3G2;

extern const char *MEDIA_MIMETYPE_AUDIO_DTS;



}  // namespace android



#endif  // MEDIA_DEFS_H_

media/libmedia/AudioRecord.cpp

+70 −10
Original line number Diff line number Diff line 
/*

**

** Copyright 2008, The Android Open Source Project

** Copyright (c) 2011-2012, The Linux Foundation. All rights reserved.

** Copyright (c) 2011-2013, The Linux Foundation. All rights reserved.

**

** Licensed under the Apache License, Version 2.0 (the "License");

** you may not use this file except in compliance with the License.
@@ -185,12 +185,51 @@ status_t AudioRecord::set( 
        return BAD_VALUE;

    }



    // validate framecount

    int minFrameCount = 0;

    status_t status = getMinFrameCount(&minFrameCount, sampleRate, format, channelMask);

    if (status != NO_ERROR) {

        return status;

    size_t inputBuffSizeInBytes = -1;

    if (AudioSystem::getInputBufferSize(sampleRate, format, channelCount, &inputBuffSizeInBytes)

            != NO_ERROR) {

        ALOGE("AudioSystem could not query the input buffer size.");

        return NO_INIT;

    }

    ALOGV("AudioRecord::set() inputBuffSizeInBytes = %d", inputBuffSizeInBytes );



    if (inputBuffSizeInBytes == 0) {

        ALOGE("Recording parameters are not supported: sampleRate %d, channelCount %d, format %d",

            sampleRate, channelCount, format);

        return BAD_VALUE;

    }



    // Change for Codec type

    int frameSizeInBytes = 0;

    if(inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION) {

        if (audio_is_linear_pcm(format)) {

             frameSizeInBytes = channelCount * (format == AUDIO_FORMAT_PCM_16_BIT ? sizeof(int16_t)

: sizeof(int8_t));

        } else {

             frameSizeInBytes = channelCount *sizeof(int16_t);

        }

    } else {

        if (format ==AUDIO_FORMAT_AMR_NB) {

             frameSizeInBytes = channelCount * 32; // Full rate framesize

        } else if (format ==AUDIO_FORMAT_EVRC) {

             frameSizeInBytes = channelCount * 23; // Full rate framesize

        } else if (format ==AUDIO_FORMAT_QCELP) {

             frameSizeInBytes = channelCount * 35; // Full rate framesize

        } else if (format ==AUDIO_FORMAT_AAC) {

             frameSizeInBytes = 2048;

        } else if ((format ==AUDIO_FORMAT_PCM_16_BIT) || (format ==AUDIO_FORMAT_PCM_8_BIT)) {

             if (audio_is_linear_pcm(format)) {

                  frameSizeInBytes = channelCount * (format == AUDIO_FORMAT_PCM_16_BIT ? sizeof(int16_t) : sizeof(int8_t));

             } else {

                  frameSizeInBytes = sizeof(int8_t);

             }

        } else if(format == AUDIO_FORMAT_AMR_WB) {

            frameSizeInBytes = channelCount * 61;



        }

    }

    // We use 2* size of input buffer for ping pong use of record buffer.

    int minFrameCount = 2 * inputBuffSizeInBytes / frameSizeInBytes;

    ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);



    if (frameCount == 0) {
@@ -204,7 +243,7 @@ status_t AudioRecord::set( 
    }



    // create the IAudioRecord

    status = openRecord_l(sampleRate, format, channelMask,

    status_t status = openRecord_l(sampleRate, format, channelMask,

                        frameCount, input);

    if (status != NO_ERROR) {

        return status;
@@ -269,12 +308,33 @@ uint32_t AudioRecord::frameCount() const 


size_t AudioRecord::frameSize() const

{

    if(inputSource() == AUDIO_SOURCE_VOICE_COMMUNICATION) {

        if (audio_is_linear_pcm(mFormat)) {

             return channelCount()*audio_bytes_per_sample(mFormat);

        } else {

            return channelCount()*sizeof(int16_t);

        }

    } else {

        if (format() ==AUDIO_FORMAT_AMR_NB) {

             return channelCount() * 32; // Full rate framesize

        } else if (format() == AUDIO_FORMAT_EVRC) {

             return channelCount() * 23; // Full rate framesize

        } else if (format() == AUDIO_FORMAT_QCELP) {

             return channelCount() * 35; // Full rate framesize

        } else if (format() == AUDIO_FORMAT_AAC) {

            // Not actual framsize but for variable frame rate AAC encoding,

           // buffer size is treated as a frame size

             return 2048;

        } else if(format() == AUDIO_FORMAT_AMR_WB) {

            return channelCount() * 61;

        }

        if (audio_is_linear_pcm(mFormat)) {

            return channelCount()*audio_bytes_per_sample(mFormat);

        } else {

            return sizeof(uint8_t);

        }

    }

}



audio_source_t AudioRecord::inputSource() const

{

media/libmediaplayerservice/Android.mk

+2 −0
Original line number Diff line number Diff line
@@ -41,10 +41,12 @@ LOCAL_SHARED_LIBRARIES := \ 
LOCAL_STATIC_LIBRARIES :=       \

    libstagefright_nuplayer     \

    libstagefright_rtsp         \

    libmedia_helper             \



LOCAL_C_INCLUDES :=                                                 \

    $(call include-path-for, graphics corecg)                       \

    $(TOP)/frameworks/av/media/libstagefright/include               \

    $(TOP)/frameworks/av/include/media                              \

    $(TOP)/frameworks/av/media/libstagefright/rtsp                  \

    $(TOP)/frameworks/av/media/libstagefright/wifi-display          \

    $(TOP)/frameworks/native/include/media/openmax                  \

media/libmediaplayerservice/StagefrightRecorder.cpp

+41 −1
Original line number Diff line number Diff line
@@ -21,7 +21,7 @@ 
//#define LOG_NDEBUG 0

#define LOG_TAG "StagefrightRecorder"

#include <utils/Log.h>



#include <media/AudioParameter.h>

#include "StagefrightRecorder.h"



#include <binder/IPCThreadState.h>
@@ -53,6 +53,7 @@ 
#include <camera/ICamera.h>

#include <camera/CameraParameters.h>

#include <gui/Surface.h>

#include <utils/String8.h>



#include <utils/Errors.h>

#include <sys/types.h>
@@ -855,6 +856,45 @@ status_t StagefrightRecorder::start() { 
}



sp<MediaSource> StagefrightRecorder::createAudioSource() {

    bool tunneledSource = false;

    const char *tunnelMime;

    {

        AudioParameter param;

        String8 key("tunneled-input-formats");

        param.add( key, String8("get") );

        String8 valueStr = AudioSystem::getParameters( 0, param.toString());

        AudioParameter result(valueStr);

        int value;

        if ( mAudioEncoder == AUDIO_ENCODER_AMR_NB &&

            result.getInt(String8("AMR"),value) == NO_ERROR ) {

            tunneledSource = true;

            tunnelMime = MEDIA_MIMETYPE_AUDIO_AMR_NB;

        }

        else if ( mAudioEncoder == AUDIO_ENCODER_QCELP &&

            result.getInt(String8("QCELP"),value) == NO_ERROR ) {

            tunneledSource = true;

            tunnelMime = MEDIA_MIMETYPE_AUDIO_QCELP;

        }

        else if ( mAudioEncoder == AUDIO_ENCODER_EVRC &&

            result.getInt(String8("EVRC"),value) == NO_ERROR ) {

            tunneledSource = true;

            tunnelMime = MEDIA_MIMETYPE_AUDIO_EVRC;

        }

    }



    if ( tunneledSource ) {

        sp<AudioSource> audioSource = NULL;

        sp<MetaData> meta = new MetaData;

        meta->setInt32(kKeyChannelCount, mAudioChannels);

        meta->setInt32(kKeySampleRate, mSampleRate);

        meta->setInt32(kKeyBitRate, mAudioBitRate);

        if (mAudioTimeScale > 0) {

            meta->setInt32(kKeyTimeScale, mAudioTimeScale);

        }

        meta->setCString( kKeyMIMEType, tunnelMime );

        audioSource = new AudioSource( mAudioSource, meta);

        return audioSource->initCheck( ) == OK ? audioSource : NULL;

    }

    sp<AudioSource> audioSource =

        new AudioSource(

                mAudioSource,