frank
7 years ago
parent
709b994c7d
commit
17b18af513
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//
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// Created by frank on 2018/2/1.
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//
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#include <jni.h> |
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#include <stdlib.h> |
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#include <unistd.h> |
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//封装格式
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#include "libavformat/avformat.h" |
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//解码
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#include "libavcodec/avcodec.h" |
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//缩放
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#include "libswscale/swscale.h" |
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//重采样
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#include "libswresample/swresample.h" |
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#include <android/log.h> |
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#define TAG "MediaPlayer" |
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#define LOGI(FORMAT,...) __android_log_print(ANDROID_LOG_INFO, TAG, FORMAT, ##__VA_ARGS__); |
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#define LOGE(FORMAT,...) __android_log_print(ANDROID_LOG_ERROR, TAG, FORMAT, ##__VA_ARGS__); |
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#define MAX_AUDIO_FRAME_SIZE 48000 * 4 |
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JNIEXPORT void JNICALL Java_com_frank_ffmpeg_AudioPlayer_play |
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(JNIEnv *env, jobject jthiz, jstring input_jstr){ |
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const char* input_cstr = (*env)->GetStringUTFChars(env,input_jstr,NULL); |
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LOGI("input_cstr=%s", input_cstr); |
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//注册组件
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av_register_all(); |
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AVFormatContext *pFormatCtx = avformat_alloc_context(); |
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//打开音频文件
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if(avformat_open_input(&pFormatCtx,input_cstr,NULL,NULL) != 0){ |
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LOGI("%s","无法打开音频文件"); |
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return; |
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} |
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//获取输入文件信息
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if(avformat_find_stream_info(pFormatCtx,NULL) < 0){ |
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LOGI("%s","无法获取输入文件信息"); |
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return; |
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} |
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//获取音频流索引位置
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int i = 0, audio_stream_idx = -1; |
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for(; i < pFormatCtx->nb_streams;i++){ |
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if(pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO){ |
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audio_stream_idx = i; |
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break; |
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} |
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} |
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//获取音频解码器
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AVCodecContext *codecCtx = pFormatCtx->streams[audio_stream_idx]->codec; |
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AVCodec *codec = avcodec_find_decoder(codecCtx->codec_id); |
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if(codec == NULL){ |
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LOGI("%s","无法获取解码器"); |
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return; |
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} |
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//打开解码器
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if(avcodec_open2(codecCtx,codec,NULL) < 0){ |
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LOGI("%s","无法打开解码器"); |
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return; |
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} |
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//压缩数据
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AVPacket *packet = (AVPacket *)av_malloc(sizeof(AVPacket)); |
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//解压缩数据
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AVFrame *frame = av_frame_alloc(); |
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//frame->16bit 44100 PCM 统一音频采样格式与采样率
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SwrContext *swrCtx = swr_alloc(); |
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//输入的采样格式
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enum AVSampleFormat in_sample_fmt = codecCtx->sample_fmt; |
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//输出采样格式16bit PCM
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enum AVSampleFormat out_sample_fmt = AV_SAMPLE_FMT_S16; |
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//输入采样率
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int in_sample_rate = codecCtx->sample_rate; |
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//输出采样率
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int out_sample_rate = in_sample_rate; |
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//声道布局（2个声道，默认立体声stereo）
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uint64_t in_ch_layout = codecCtx->channel_layout; |
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//输出的声道布局（立体声）
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uint64_t out_ch_layout = AV_CH_LAYOUT_STEREO; |
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swr_alloc_set_opts(swrCtx, |
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out_ch_layout,out_sample_fmt,out_sample_rate, |
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in_ch_layout,in_sample_fmt,in_sample_rate, |
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0, NULL); |
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swr_init(swrCtx); |
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//输出的声道个数
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int out_channel_nb = av_get_channel_layout_nb_channels(out_ch_layout); |
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jclass player_class = (*env)->GetObjectClass(env,jthiz); |
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if(!player_class){ |
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LOGE("player_class not found..."); |
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} |
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//AudioTrack对象
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jmethodID audio_track_method = (*env)->GetMethodID(env,player_class,"createAudioTrack","(II)Landroid/media/AudioTrack;"); |
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if(!audio_track_method){ |
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LOGE("audio_track_method not found..."); |
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} |
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jobject audio_track = (*env)->CallObjectMethod(env,jthiz,audio_track_method,out_sample_rate,out_channel_nb); |
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//调用play方法
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jclass audio_track_class = (*env)->GetObjectClass(env,audio_track); |
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jmethodID audio_track_play_mid = (*env)->GetMethodID(env,audio_track_class,"play","()V"); |
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(*env)->CallVoidMethod(env,audio_track,audio_track_play_mid); |
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//获取write()方法
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jmethodID audio_track_write_mid = (*env)->GetMethodID(env,audio_track_class,"write","([BII)I"); |
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//16bit 44100 PCM 数据
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uint8_t *out_buffer = (uint8_t *)av_malloc(MAX_AUDIO_FRAME_SIZE); |
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int got_frame = 0,index = 0, ret; |
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//不断读取编码数据
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while(av_read_frame(pFormatCtx,packet) >= 0){ |
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//解码音频类型的Packet
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if(packet->stream_index == audio_stream_idx){ |
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//解码
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ret = avcodec_decode_audio4(codecCtx,frame,&got_frame,packet); |
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if(ret < 0){ |
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break; |
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} |
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//解码一帧成功
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if(got_frame > 0){ |
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LOGI("decode frame count=%d",index++); |
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//音频格式转换
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swr_convert(swrCtx, &out_buffer, MAX_AUDIO_FRAME_SIZE,(const uint8_t **)frame->data,frame->nb_samples); |
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int out_buffer_size = av_samples_get_buffer_size(NULL, out_channel_nb, |
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frame->nb_samples, out_sample_fmt, 1); |
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jbyteArray audio_sample_array = (*env)->NewByteArray(env,out_buffer_size); |
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jbyte* sample_byte_array = (*env)->GetByteArrayElements(env,audio_sample_array,NULL); |
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//拷贝缓冲数据
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memcpy(sample_byte_array, out_buffer, (size_t) out_buffer_size); |
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//释放数组
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(*env)->ReleaseByteArrayElements(env,audio_sample_array,sample_byte_array,0); |
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//调用AudioTrack的write方法进行播放
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(*env)->CallIntMethod(env,audio_track,audio_track_write_mid, |
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audio_sample_array,0,out_buffer_size); |
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//释放局部引用
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(*env)->DeleteLocalRef(env,audio_sample_array); |
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usleep(1000 * 16); |
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} |
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} |
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av_free_packet(packet); |
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} |
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LOGI("decode audio finish"); |
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av_frame_free(&frame); |
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av_free(out_buffer); |
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swr_free(&swrCtx); |
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avcodec_close(codecCtx); |
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avformat_close_input(&pFormatCtx); |
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(*env)->ReleaseStringUTFChars(env,input_jstr,input_cstr); |
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} |
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@ -0,0 +1,319 @@ |
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//
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// Created by frank on 2018/2/1.
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//
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#include <jni.h> |
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#include <string.h> |
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#include "libavcodec/avcodec.h" |
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#include "libavformat/avformat.h" |
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#include "libswscale/swscale.h" |
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#include "libswresample/swresample.h" |
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#include "libavutil/samplefmt.h" |
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#include <SLES/OpenSLES.h> |
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#include <SLES/OpenSLES_Android.h> |
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#include <android/log.h> |
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#include <libavutil/opt.h> |
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#define TAG "OpenSLPlayer" |
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#define LOGI(FORMAT,...) __android_log_print(ANDROID_LOG_INFO, TAG, FORMAT,##__VA_ARGS__); |
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#define LOGE(FORMAT,...) __android_log_print(ANDROID_LOG_ERROR, TAG, FORMAT,##__VA_ARGS__); |
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//引擎接口
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SLObjectItf engineObject = NULL; |
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SLEngineItf engineEngine; |
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//输出混音器接口
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SLObjectItf outputMixObject = NULL; |
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SLEnvironmentalReverbItf outputMixEnvironmentalReverb = NULL; |
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//缓冲播放器接口
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SLObjectItf bqPlayerObject = NULL; |
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SLPlayItf bqPlayerPlay; |
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SLAndroidSimpleBufferQueueItf bqPlayerBufferQueue; |
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SLEffectSendItf bqPlayerEffectSend; |
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SLVolumeItf bqPlayerVolume; |
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//音效设置
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const SLEnvironmentalReverbSettings reverbSettings = SL_I3DL2_ENVIRONMENT_PRESET_STONECORRIDOR; |
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void *buffer; |
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size_t bufferSize; |
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uint8_t *outputBuffer; |
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size_t outputBufferSize; |
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//FFmpeg相关
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AVPacket packet; |
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int audioStream; |
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AVFrame *aFrame; |
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SwrContext *swr; |
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AVFormatContext *aFormatCtx; |
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AVCodecContext *aCodecCtx; |
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int frame_count = 0; |
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int createAudioPlayer(int *rate, int *channel, const char *file_name) ; |
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// 释放相关资源
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int releaseAudioPlayer(); |
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// 获取PCM数据, 自动回调获取
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int getPCM(void **pcm, size_t *pcmSize) ; |
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//播放回调方法
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void bqPlayerCallback(SLAndroidSimpleBufferQueueItf bufferQueueItf, void *context) { |
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bufferSize = 0; |
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getPCM(&buffer, &bufferSize); |
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//如果buffer不为空，入待播放队列
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if (NULL != buffer && 0 != bufferSize) { |
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SLresult result; |
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result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, buffer, bufferSize); |
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if(result < 0){ |
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LOGE("Enqueue error..."); |
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} else{ |
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LOGI("decode frame count=%d", frame_count++); |
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} |
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} |
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} |
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//创建OpenSLES引擎
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void createEngine() { |
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SLresult result; |
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//创建引擎
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result = slCreateEngine(&engineObject, 0, NULL, 0, NULL, NULL); |
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LOGI("slCreateEngine=%d", result); |
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result = (*engineObject)->Realize(engineObject, SL_BOOLEAN_FALSE); |
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LOGI("engineObject->Realize=%d", result); |
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//获取引擎接口
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result = (*engineObject)->GetInterface(engineObject, SL_IID_ENGINE, &engineEngine); |
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LOGI("engineObject->GetInterface=%d", result); |
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//创建输出混音器
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result = (*engineEngine)->CreateOutputMix(engineEngine, &outputMixObject, 0, 0, 0); |
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LOGI("CreateOutputMix=%d", result); |
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//关联输出混音器
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result = (*outputMixObject)->Realize(outputMixObject, SL_BOOLEAN_FALSE); |
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LOGI("outputMixObject->Realize=%d", result); |
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//获取reverb接口
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result = (*outputMixObject)->GetInterface(outputMixObject, SL_IID_ENVIRONMENTALREVERB, |
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&outputMixEnvironmentalReverb); |
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LOGI("outputMixObject->GetInterface=%d", result); |
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if (SL_RESULT_SUCCESS == result) { |
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result = (*outputMixEnvironmentalReverb)->SetEnvironmentalReverbProperties( |
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outputMixEnvironmentalReverb, &reverbSettings); |
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} |
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LOGI("SetEnvironmentalReverbProperties=%d", result); |
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} |
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//创建带有缓冲队列的音频播放器
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void createBufferQueueAudioPlayer(int rate, int channel, int bitsPerSample) { |
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SLresult result; |
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//配置音频源
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SLDataLocator_AndroidSimpleBufferQueue buffer_queue = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, 2}; |
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SLDataFormat_PCM format_pcm; |
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format_pcm.formatType = SL_DATAFORMAT_PCM; |
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format_pcm.numChannels = (SLuint32) channel; |
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format_pcm.bitsPerSample = (SLuint32) bitsPerSample; |
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format_pcm.samplesPerSec = (SLuint32) (rate * 1000); |
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format_pcm.containerSize = 16; |
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if (channel == 2) |
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format_pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT; |
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else |
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format_pcm.channelMask = SL_SPEAKER_FRONT_CENTER; |
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format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN; |
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SLDataSource audioSrc = {&buffer_queue, &format_pcm}; |
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//配置音频池
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SLDataLocator_OutputMix loc_outmix = {SL_DATALOCATOR_OUTPUTMIX, outputMixObject}; |
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SLDataSink audioSnk = {&loc_outmix, NULL}; |
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//创建音频播放器
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const SLInterfaceID ids[3] = {SL_IID_BUFFERQUEUE, SL_IID_EFFECTSEND, SL_IID_VOLUME}; |
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const SLboolean req[3] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE}; |
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result = (*engineEngine)->CreateAudioPlayer(engineEngine, &bqPlayerObject, &audioSrc, &audioSnk, |
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3, ids, req); |
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LOGI("CreateAudioPlayer=%d", result); |
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//关联播放器
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result = (*bqPlayerObject)->Realize(bqPlayerObject, SL_BOOLEAN_FALSE); |
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LOGI("bqPlayerObject Realize=%d", result); |
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//获取播放接口
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result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_PLAY, &bqPlayerPlay); |
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LOGI("GetInterface bqPlayerPlay=%d", result); |
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//获取缓冲队列接口
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result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_BUFFERQUEUE, |
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&bqPlayerBufferQueue); |
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LOGI("GetInterface bqPlayerBufferQueue=%d", result); |
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//注册缓冲队列回调
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result = (*bqPlayerBufferQueue)->RegisterCallback(bqPlayerBufferQueue, bqPlayerCallback, NULL); |
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LOGI("RegisterCallback=%d", result); |
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//获取音效接口
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result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_EFFECTSEND, |
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&bqPlayerEffectSend); |
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LOGI("GetInterface effect=%d", result); |
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//获取音量接口
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result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_VOLUME, &bqPlayerVolume); |
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LOGI("GetInterface volume=%d", result); |
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//开始播放音乐
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result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PLAYING); |
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LOGI("SetPlayState=%d", result); |
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} |
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int createAudioPlayer(int *rate, int *channel, const char *file_name) { |
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//注册相关组件
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av_register_all(); |
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aFormatCtx = avformat_alloc_context(); |
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//打开音频文件
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if (avformat_open_input(&aFormatCtx, file_name, NULL, NULL) != 0) { |
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LOGE("Couldn't open file:%s\n", file_name); |
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return -1; // Couldn't open file
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} |
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//寻找stream信息
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if (avformat_find_stream_info(aFormatCtx, NULL) < 0) { |
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LOGE("Couldn't find stream information."); |
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return -1; |
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} |
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//寻找音频stream
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int i; |
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audioStream = -1; |
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for (i = 0; i < aFormatCtx->nb_streams; i++) { |
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if (aFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO && |
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audioStream < 0) { |
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audioStream = i; |
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} |
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} |
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if (audioStream == -1) { |
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LOGE("Couldn't find audio stream!"); |
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return -1; |
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} |
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//获取解码器context
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aCodecCtx = aFormatCtx->streams[audioStream]->codec; |
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//寻找音频解码器
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AVCodec *aCodec = avcodec_find_decoder(aCodecCtx->codec_id); |
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if (!aCodec) { |
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fprintf(stderr, "Unsupported codec!\n"); |
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return -1; |
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} |
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//打开解码器
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if (avcodec_open2(aCodecCtx, aCodec, NULL) < 0) { |
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LOGE("Could not open codec."); |
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return -1; |
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} |
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aFrame = av_frame_alloc(); |
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// 设置格式转换
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swr = swr_alloc(); |
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av_opt_set_int(swr, "in_channel_layout", aCodecCtx->channel_layout, 0); |
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av_opt_set_int(swr, "out_channel_layout", aCodecCtx->channel_layout, 0); |
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av_opt_set_int(swr, "in_sample_rate", aCodecCtx->sample_rate, 0); |
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av_opt_set_int(swr, "out_sample_rate", aCodecCtx->sample_rate, 0); |
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av_opt_set_sample_fmt(swr, "in_sample_fmt", aCodecCtx->sample_fmt, 0); |
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av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0); |
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swr_init(swr); |
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// 分配PCM数据缓存
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outputBufferSize = 8196; |
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outputBuffer = (uint8_t *) malloc(sizeof(uint8_t) * outputBufferSize); |
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// 返回sample rate和channels
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*rate = aCodecCtx->sample_rate; |
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*channel = aCodecCtx->channels; |
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return 0; |
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} |
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// 获取PCM数据, 自动回调获取
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int getPCM(void **pcm, size_t *pcmSize) { |
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while (av_read_frame(aFormatCtx, &packet) >= 0) { |
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int frameFinished = 0; |
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//音频流
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if (packet.stream_index == audioStream) { |
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avcodec_decode_audio4(aCodecCtx, aFrame, &frameFinished, &packet); |
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//解码完一帧数据
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if (frameFinished) { |
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// data_size为音频数据所占的字节数
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int data_size = av_samples_get_buffer_size( |
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aFrame->linesize, aCodecCtx->channels, |
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aFrame->nb_samples, aCodecCtx->sample_fmt, 1); |
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if (data_size > outputBufferSize) { |
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outputBufferSize = (size_t) data_size; |
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outputBuffer = (uint8_t *) realloc(outputBuffer, sizeof(uint8_t) * outputBufferSize); |
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} |
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// 音频格式转换
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swr_convert(swr, &outputBuffer, aFrame->nb_samples, |
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(uint8_t const **) (aFrame->extended_data), |
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aFrame->nb_samples); |
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// 返回pcm数据
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*pcm = outputBuffer; |
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*pcmSize = (size_t) data_size; |
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return 0; |
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} |
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} |
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} |
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return -1; |
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} |
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// 释放相关资源
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int releaseAudioPlayer() { |
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av_packet_unref(&packet); |
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av_free(outputBuffer); |
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av_free(aFrame); |
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avcodec_close(aCodecCtx); |
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avformat_close_input(&aFormatCtx); |
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return 0; |
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} |
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JNIEXPORT void JNICALL Java_com_frank_ffmpeg_AudioPlayer_playAudio |
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(JNIEnv * env, jclass jobject, jstring filePath) { |
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int rate, channel; |
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const char *file_name = (*env)->GetStringUTFChars(env, filePath, NULL); |
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LOGI("file_name=%s", file_name); |
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// 创建音频解码器
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createAudioPlayer(&rate, &channel, file_name); |
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// 创建播放引擎
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createEngine(); |
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// 创建缓冲队列音频播放器
|
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createBufferQueueAudioPlayer(rate, channel, SL_PCMSAMPLEFORMAT_FIXED_16); |
||||
|
||||
// 启动音频播放
|
||||
bqPlayerCallback(bqPlayerBufferQueue, NULL); |
||||
} |
||||
|
||||
//停止播放，释放相关资源
|
||||
JNIEXPORT jint JNICALL Java_com_frank_ffmpeg_AudioPlayer_stop |
||||
(JNIEnv * env, jclass jobject) { |
||||
if (bqPlayerObject != NULL) { |
||||
(*bqPlayerObject)->Destroy(bqPlayerObject); |
||||
bqPlayerObject = NULL; |
||||
bqPlayerPlay = NULL; |
||||
bqPlayerBufferQueue = NULL; |
||||
bqPlayerEffectSend = NULL; |
||||
bqPlayerVolume = NULL; |
||||
} |
||||
|
||||
if (outputMixObject != NULL) { |
||||
(*outputMixObject)->Destroy(outputMixObject); |
||||
outputMixObject = NULL; |
||||
outputMixEnvironmentalReverb = NULL; |
||||
} |
||||
|
||||
if (engineObject != NULL) { |
||||
(*engineObject)->Destroy(engineObject); |
||||
engineObject = NULL; |
||||
engineEngine = NULL; |
||||
} |
||||
|
||||
// 释放解码器相关资源
|
||||
releaseAudioPlayer(); |
||||
} |
||||
@ -0,0 +1,45 @@ |
||||
package com.frank.ffmpeg; |
||||
|
||||
import android.media.AudioFormat; |
||||
import android.media.AudioManager; |
||||
import android.media.AudioTrack; |
||||
|
||||
/** |
||||
* 音频播放器 |
||||
* Created by frank on 2018/2/1. |
||||
*/ |
||||
|
||||
public class AudioPlayer { |
||||
static { |
||||
System.loadLibrary("media-handle"); |
||||
} |
||||
//调用AudioTrack播放
|
||||
public native void play(String audioPath); |
||||
//调用OpenSL ES播放
|
||||
public native void playAudio(String audioPath); |
||||
//调用OpenSL ES播放
|
||||
public native void stop(String audioPath); |
||||
|
||||
/** |
||||
* 创建一个AudioTrack对象 |
||||
* @param sampleRate 采样率 |
||||
* @param channels 声道布局 |
||||
* @return AudioTrack |
||||
*/ |
||||
public AudioTrack createAudioTrack(int sampleRate, int channels){ |
||||
int audioFormat = AudioFormat.ENCODING_PCM_16BIT; |
||||
int channelConfig; |
||||
if(channels == 1){ |
||||
channelConfig = AudioFormat.CHANNEL_OUT_MONO; |
||||
}else if(channels == 2){ |
||||
channelConfig = AudioFormat.CHANNEL_OUT_STEREO; |
||||
}else{ |
||||
channelConfig = AudioFormat.CHANNEL_OUT_STEREO; |
||||
} |
||||
|
||||
int bufferSizeInBytes = AudioTrack.getMinBufferSize(sampleRate, channelConfig, audioFormat); |
||||
|
||||
return new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate, channelConfig, audioFormat, |
||||
bufferSizeInBytes, AudioTrack.MODE_STREAM); |
||||
} |
||||
} |
||||
Loading…
Reference in new issue