要在Android/iOS端实现语音对讲,原型为微信与米聊,开始预演所用技术,找到以下资料。
在Android开发中,需要录音并发送到对方设备上。这时问题来了,手机常会是GPRS、3G等方式上网,所以节省流量是非常关键的,使用Speex来压缩音频文件,可以将音频压文件小数倍。
1.去Speex官网下载最新Speex源码。
2.创建一个新的应用(我创建的应用名为Audio),并创建一个jni目录($project/jni)。
3.把speex源码目录下的libspeex和include目录及其子目录文件全部拷贝到$project/jni目录下($project/jni/libspeex and $project/jni/include)。
4.在jni目录下新增Android.mk文件,编辑内容如下
LOCAL_PATH := $(call my-dir) include $(CLEAR_VARS) LOCAL_MODULE := libspeex LOCAL_CFLAGS = -DFIXED_POINT -DUSE_KISS_FFT -DEXPORT="" -UHAVE_CONFIG_H LOCAL_C_INCLUDES := $(LOCAL_PATH)/include LOCAL_SRC_FILES := \ ./speex_jni.cpp \ ./libspeex/bits.c \ ./libspeex/buffer.c \ ./libspeex/cb_search.c \ ./libspeex/exc_10_16_table.c \ ./libspeex/exc_10_32_table.c \ ./libspeex/exc_20_32_table.c \ ./libspeex/exc_5_256_table.c \ ./libspeex/exc_5_64_table.c \ ./libspeex/exc_8_128_table.c \ ./libspeex/fftwrap.c \ ./libspeex/filterbank.c \ ./libspeex/filters.c \ ./libspeex/gain_table.c \ ./libspeex/gain_table_lbr.c \ ./libspeex/hexc_10_32_table.c \ ./libspeex/hexc_table.c \ ./libspeex/high_lsp_tables.c \ ./libspeex/jitter.c \ ./libspeex/kiss_fft.c \ ./libspeex/kiss_fftr.c \ ./libspeex/lpc.c \ ./libspeex/lsp.c \ ./libspeex/lsp_tables_nb.c \ ./libspeex/ltp.c \ ./libspeex/mdf.c \ ./libspeex/modes.c \ ./libspeex/modes_wb.c \ ./libspeex/nb_celp.c \ ./libspeex/preprocess.c \ ./libspeex/quant_lsp.c \ ./libspeex/resample.c \ ./libspeex/sb_celp.c \ ./libspeex/scal.c \ ./libspeex/smallft.c \ ./libspeex/speex.c \ ./libspeex/speex_callbacks.c \ ./libspeex/speex_header.c \ ./libspeex/stereo.c \ ./libspeex/vbr.c \ ./libspeex/vq.c \ ./libspeex/window.c include $(BUILD_SHARED_LIBRARY)
APP_ABI := armeabi armeabi-v7a
#ifndef __SPEEX_TYPES_H__ #define __SPEEX_TYPES_H__ typedef short spx_int16_t; typedef unsigned short spx_uint16_t; typedef int spx_int32_t; typedef unsigned int spx_uint32_t; #endif
#include <jni.h> #include <string.h> #include <unistd.h> #include <speex/speex.h> static int codec_open = 0; static int dec_frame_size; static int enc_frame_size; static SpeexBits ebits, dbits; void *enc_state; void *dec_state; static JavaVM *gJavaVM; extern "C" JNIEXPORT jint JNICALL Java_com_audio_Speex_open (JNIEnv *env, jobject obj, jint compression) { int tmp; if (codec_open++ != 0) return (jint)0; speex_bits_init(&ebits); speex_bits_init(&dbits); enc_state = speex_encoder_init(&speex_nb_mode); dec_state = speex_decoder_init(&speex_nb_mode); tmp = compression; speex_encoder_ctl(enc_state, SPEEX_SET_QUALITY, &tmp); speex_encoder_ctl(enc_state, SPEEX_GET_FRAME_SIZE, &enc_frame_size); speex_decoder_ctl(dec_state, SPEEX_GET_FRAME_SIZE, &dec_frame_size); return (jint)0; } extern "C" JNIEXPORT jint Java_com_audio_Speex_encode (JNIEnv *env, jobject obj, jshortArray lin, jint offset, jbyteArray encoded, jint size) { jshort buffer[enc_frame_size]; jbyte output_buffer[enc_frame_size]; int nsamples = (size-1)/enc_frame_size + 1; int i, tot_bytes = 0; if (!codec_open) return 0; speex_bits_reset(&ebits); for (i = 0; i < nsamples; i++) { env->GetShortArrayRegion(lin, offset + i*enc_frame_size, enc_frame_size, buffer); speex_encode_int(enc_state, buffer, &ebits); } //env->GetShortArrayRegion(lin, offset, enc_frame_size, buffer); //speex_encode_int(enc_state, buffer, &ebits); tot_bytes = speex_bits_write(&ebits, (char *)output_buffer, enc_frame_size); env->SetByteArrayRegion(encoded, 0, tot_bytes, output_buffer); return (jint)tot_bytes; } extern "C" JNIEXPORT jint JNICALL Java_com_audio_Speex_decode (JNIEnv *env, jobject obj, jbyteArray encoded, jshortArray lin, jint size) { jbyte buffer[dec_frame_size]; jshort output_buffer[dec_frame_size]; jsize encoded_length = size; if (!codec_open) return 0; env->GetByteArrayRegion(encoded, 0, encoded_length, buffer); speex_bits_read_from(&dbits, (char *)buffer, encoded_length); speex_decode_int(dec_state, &dbits, output_buffer); env->SetShortArrayRegion(lin, 0, dec_frame_size, output_buffer); return (jint)dec_frame_size; } extern "C" JNIEXPORT jint JNICALL Java_com_audio_getFrameSize (JNIEnv *env, jobject obj) { if (!codec_open) return 0; return (jint)enc_frame_size; } extern "C" JNIEXPORT void JNICALL Java_com_audio_Speex_close (JNIEnv *env, jobject obj) { if (--codec_open != 0) return; speex_bits_destroy(&ebits); speex_bits_destroy(&dbits); speex_decoder_destroy(dec_state); speex_encoder_destroy(enc_state); }
package com.audio; class Speex { /* quality * 1 : 4kbps (very noticeable artifacts, usually intelligible) * 2 : 6kbps (very noticeable artifacts, good intelligibility) * 4 : 8kbps (noticeable artifacts sometimes) * 6 : 11kpbs (artifacts usually only noticeable with headphones) * 8 : 15kbps (artifacts not usually noticeable) */ private static final int DEFAULT_COMPRESSION = 8; Speex() { } public void init() { load(); open(DEFAULT_COMPRESSION); } private void load() { try { System.loadLibrary("speex"); } catch (Throwable e) { e.printStackTrace(); } } public native int open(int compression); public native int getFrameSize(); public native int decode(byte encoded[], short lin[], int size); public native int encode(short lin[], int offset, byte encoded[], int size); public native void close(); }
9.打开cygwin工具,切换到项目目录(我项目是在F:\workspace\Audio),输入$NDK/ndk-build
cygwin工具的安装与配置,可以看这篇文章——使用NDK与环境搭建
会在项目中生成libs目录和libspeex.so文件,这就是Speex类中System.loadLibrary("speex");代码引用的,系统会根据操作系统由"speex"找到对应的动态库libspeex.so,Windows下是.dll文件,linux下是.so文件。
当前,我的项目结构如下图
可以从android-recorder下载代码作为参考
其它文章:
http://hhuai.github.com/blog/2012/02/05/ios-and-andorid-voice/
Ios实现amr编解码
介绍
学习ios第一个练手功能就是给已有产品加上语音通信功能,能够互通ios与android。这里给出自己的一些心得,希望能给他人一些参考。
资料搜集与参考
类似产品使用的技术
talkbox Android版用的是ilbc的第三方编解码库,在iPhone上用的是caf
微信 Android版估计是amr估计转码的是交给腾讯强大的服务器了。
米聊 Android版和Iphone版用的都是speex
原文链接:http://blog.csdn.net/chenfeng0104/article/details/7088138