audio: Add support for tunnel mode recording (01cb926c) · Commits · e / os / android_frameworks_av

include/media/AudioRecord.h

+5 −1

Original line number	Diff line number	Diff line
		@@ -186,7 +186,11 @@ public:
		audio_format_t format() const;
		uint32_t channelCount() const;
		size_t frameCount() const;
		size_t frameSize() const { return mFrameSize; }
		#ifdef QCOM_HARDWARE
		size_t frameSize() const;
		#else
		size_t frameSize() const { return mFrameSize; };
		#endif
		audio_source_t inputSource() const;

include/media/stagefright/AudioSource.h

+14 −1

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.
		@@ -104,8 +105,20 @@ private:

		AudioSource(const AudioSource &);
		AudioSource &operator=(const AudioSource &);
		};

		#ifdef QCOM_HARDWARE
		//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 );
		#endif
		};
		} // namespace android

		#endif // AUDIO_SOURCE_H_

media/libmedia/AudioRecord.cpp

+88 −1

Original line number	Diff line number	Diff line
		@@ -3,6 +3,7 @@
		** Not a Contribution.
		**
		** Copyright 2008, The Android Open Source Project
		** 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.
		@@ -136,6 +137,8 @@ status_t AudioRecord::set(
		bool threadCanCallJava,
		int sessionId)
		{
		status_t status;

		// FIXME "int" here is legacy and will be replaced by size_t later
		if (frameCountInt < 0) {
		ALOGE("Invalid frame count %d", frameCountInt);
		@@ -184,11 +187,15 @@ status_t AudioRecord::set(
		#endif
		mChannelCount = channelCount;

		#ifdef QCOM_HARDWARE
		mFrameSize = frameSize();
		#else
		if (audio_is_linear_pcm(format)) {
		mFrameSize = channelCount * audio_bytes_per_sample(format);
		} else {
		mFrameSize = sizeof(uint8_t);
		}
		#endif

		if (sessionId == 0 ) {
		mSessionId = AudioSystem::newAudioSessionId();
		@@ -207,12 +214,60 @@ status_t AudioRecord::set(
		return BAD_VALUE;
		}

		#ifdef QCOM_HARDWARE
		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 * AMR_FRAMESIZE; // Full rate framesize
		} else if (format ==AUDIO_FORMAT_EVRC) {
		frameSizeInBytes = channelCount * EVRC_FRAMESIZE; // Full rate framesize
		} else if (format ==AUDIO_FORMAT_QCELP) {
		frameSizeInBytes = channelCount * QCELP_FRAMESIZE; // Full rate framesize
		} else if (format ==AUDIO_FORMAT_AAC) {
		frameSizeInBytes = AAC_FRAMESIZE;
		} 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 * AMR_WB_FRAMESIZE;

		}
		}
		// We use 2* size of input buffer for ping pong use of record buffer.
		int minFrameCount = 2 * inputBuffSizeInBytes / frameSizeInBytes;
		#else
		// validate framecount
		size_t minFrameCount = 0;
		status_t status = getMinFrameCount(&minFrameCount, sampleRate, format, channelMask);
		status = getMinFrameCount(&minFrameCount, sampleRate, format, channelMask);
		if (status != NO_ERROR) {
		return status;
		}
		#endif
		ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);

		if (frameCount == 0) {
		@@ -286,6 +341,38 @@ size_t AudioRecord::frameCount() const
		return mFrameCount;
		}

		#ifdef QCOM_HARDWARE
		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() * AMR_FRAMESIZE; // Full rate framesize
		} else if (format() == AUDIO_FORMAT_EVRC) {
		return channelCount() * EVRC_FRAMESIZE; // Full rate framesize
		} else if (format() == AUDIO_FORMAT_QCELP) {
		return channelCount() * QCELP_FRAMESIZE; // 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 AAC_FRAMESIZE;
		} else if(format() == AUDIO_FORMAT_AMR_WB) {
		return channelCount() * AMR_WB_FRAMESIZE;
		}
		if (audio_is_linear_pcm(mFormat)) {
		return channelCount()*audio_bytes_per_sample(mFormat);
		} else {
		return sizeof(uint8_t);
		}
		}
		}
		#endif

		audio_source_t AudioRecord::inputSource() const
		{
		return mInputSource;

media/libmediaplayerservice/Android.mk

+1 −0

Original line number	Diff line number	Diff line
		@@ -43,6 +43,7 @@ LOCAL_SHARED_LIBRARIES := \
		LOCAL_STATIC_LIBRARIES := \
		libstagefright_nuplayer \
		libstagefright_rtsp \
		libmedia_helper \

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

media/libmediaplayerservice/StagefrightRecorder.cpp

+51 −1

Original line number	Diff line number	Diff line
		@@ -20,7 +20,7 @@
		//#define LOG_NDEBUG 0
		#define LOG_TAG "StagefrightRecorder"
		#include <utils/Log.h>

		#include <media/AudioParameter.h>
		#include "StagefrightRecorder.h"

		#include <binder/IPCThreadState.h>
		@@ -47,6 +47,7 @@
		#include <camera/ICamera.h>
		#include <camera/CameraParameters.h>
		#include <gui/Surface.h>
		#include <utils/String8.h>

		#include <utils/Errors.h>
		#include <sys/types.h>
		@@ -864,6 +865,55 @@ status_t StagefrightRecorder::start() {
		}

		sp<MediaSource> StagefrightRecorder::createAudioSource() {
		#ifdef QCOM_HARDWARE
		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;
		}
		#ifdef ENABLE_QC_AV_ENHANCEMENTS
		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;
		}
		#endif
		else if ( mAudioEncoder == AUDIO_ENCODER_AMR_WB &&
		result.getInt(String8("AWB"),value) == NO_ERROR ) {
		tunneledSource = true;
		tunnelMime = MEDIA_MIMETYPE_AUDIO_AMR_WB;
		}
		}

		if ( tunneledSource ) {
		ALOGD("tunnel recording");
		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;
		}
		#endif
		sp<AudioSource> audioSource =
		new AudioSource(
		mAudioSource,