From 8922e50150f4a917007d3316c3d19dd9523fa5b2 Mon Sep 17 00:00:00 2001 From: xufulong <839789740@qq.com> Date: Tue, 14 Apr 2020 00:21:15 +0800 Subject: [PATCH] translate audio_player.c into English translate audio_player.c into English --- app/src/main/cpp/audio_lame.c | 9 +- app/src/main/cpp/audio_player.c | 230 ++++++++++++++++---------------- 2 files changed, 121 insertions(+), 118 deletions(-) diff --git a/app/src/main/cpp/audio_lame.c b/app/src/main/cpp/audio_lame.c index 9bc6553..487f7c9 100644 --- a/app/src/main/cpp/audio_lame.c +++ b/app/src/main/cpp/audio_lame.c @@ -190,7 +190,8 @@ AUDIO_PLAYER_FUNC(void, lameInit, jint inSampleRate, jint outChannel, jint outSampleRate, jint outBitrate, jfloat scaleInput, jint mode, jint vbrMode, jint quality, jint vbrQuality, jint abrMeanBitrate, jint lowPassFreq, jint highPassFreq, jstring id3tagTitle, - jstring id3tagArtist, jstring id3tagAlbum, jstring id3tagYear, jstring id3tagComment) { + jstring id3tagArtist, jstring id3tagAlbum, jstring id3tagYear, + jstring id3tagComment) { glf = initialize(env, inSampleRate, outChannel, outSampleRate, outBitrate, scaleInput, mode, vbrMode, @@ -201,17 +202,17 @@ AUDIO_PLAYER_FUNC(void, lameInit, } AUDIO_PLAYER_FUNC(jint, lameEncode, - jshortArray buffer_l, jshortArray buffer_r, jint samples, jbyteArray mp3buf) { + jshortArray buffer_l, jshortArray buffer_r, jint samples, jbyteArray mp3buf) { return encode(env, glf, buffer_l, buffer_r, samples, mp3buf); } AUDIO_PLAYER_FUNC(jint, encodeBufferInterleaved, - jshortArray pcm, jint samples, jbyteArray mp3buf) { + jshortArray pcm, jint samples, jbyteArray mp3buf) { return encodeBufferInterleaved(env, glf, pcm, samples, mp3buf); } AUDIO_PLAYER_FUNC(jint, lameFlush, - jbyteArray mp3buf) { + jbyteArray mp3buf) { return flush(env, glf, mp3buf); } diff --git a/app/src/main/cpp/audio_player.c b/app/src/main/cpp/audio_player.c index 99d22f4..eae822d 100644 --- a/app/src/main/cpp/audio_player.c +++ b/app/src/main/cpp/audio_player.c @@ -4,15 +4,12 @@ #include #include #include -//封装格式 +#include + #include "libavformat/avformat.h" -//解码 #include "libavcodec/avcodec.h" -//缩放 #include "libswscale/swscale.h" -//重采样 #include "libswresample/swresample.h" -#include #include "ffmpeg_jni_define.h" #define TAG "AudioPlayer" @@ -20,133 +17,138 @@ #define MAX_AUDIO_FRAME_SIZE 48000 * 4 AUDIO_PLAYER_FUNC(void, play, jstring input_jstr) { - const char* input_cstr = (*env)->GetStringUTFChars(env,input_jstr,NULL); - LOGI(TAG, "input_cstr=%s", input_cstr); - //注册组件 - av_register_all(); - AVFormatContext *pFormatCtx = avformat_alloc_context(); - //打开音频文件 - if(avformat_open_input(&pFormatCtx,input_cstr,NULL,NULL) != 0){ - LOGE(TAG, "无法打开音频文件"); - return; - } - //获取输入文件信息 - if(avformat_find_stream_info(pFormatCtx,NULL) < 0){ - LOGE(TAG, "无法获取输入文件信息"); - return; - } - //获取音频流索引位置 - int i = 0, audio_stream_idx = -1; - for(; i < pFormatCtx->nb_streams;i++){ - if(pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO){ - audio_stream_idx = i; - break; - } - } + const char *input_cstr = (*env)->GetStringUTFChars(env, input_jstr, NULL); + LOGI(TAG, "input_cstr=%s", input_cstr); + //register all modules + av_register_all(); + AVFormatContext *pFormatCtx = avformat_alloc_context(); + //open the audio file + if (avformat_open_input(&pFormatCtx, input_cstr, NULL, NULL) != 0) { + LOGE(TAG, "Couldn't open the audio file!"); + return; + } + //find all streams info + if (avformat_find_stream_info(pFormatCtx, NULL) < 0) { + LOGE(TAG, "Couldn't find stream info!"); + return; + } + //get the audio stream index in the stream array + int i = 0, audio_stream_idx = -1; + for (; i < pFormatCtx->nb_streams; i++) { + if (pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) { + audio_stream_idx = i; + break; + } + } - //获取音频解码器 - AVCodecContext *codecCtx = pFormatCtx->streams[audio_stream_idx]->codec; - AVCodec *codec = avcodec_find_decoder(codecCtx->codec_id); - if(codec == NULL){ - LOGE(TAG, "无法获取解码器"); - return; - } - //打开解码器 - if(avcodec_open2(codecCtx,codec,NULL) < 0){ - LOGE(TAG, "无法打开解码器"); - return; - } - //压缩数据 - AVPacket *packet = (AVPacket *)av_malloc(sizeof(AVPacket)); - //解压缩数据 - AVFrame *frame = av_frame_alloc(); - //frame->16bit 44100 PCM 统一音频采样格式与采样率 - SwrContext *swrCtx = swr_alloc(); + //find audio decoder + AVCodecContext *codecCtx = pFormatCtx->streams[audio_stream_idx]->codec; + AVCodec *codec = avcodec_find_decoder(codecCtx->codec_id); + if (codec == NULL) { + LOGE(TAG, "Couldn't find audio decoder!"); + return; + } + //open audio decoder + if (avcodec_open2(codecCtx, codec, NULL) < 0) { + LOGE(TAG, "Couldn't open audio decoder"); + return; + } + //malloc packet memory + AVPacket *packet = (AVPacket *) av_malloc(sizeof(AVPacket)); + //malloc frame memory + AVFrame *frame = av_frame_alloc(); + //frame->16bit 44100 PCM + SwrContext *swrCtx = swr_alloc(); - //输入的采样格式 - enum AVSampleFormat in_sample_fmt = codecCtx->sample_fmt; - //输出采样格式16bit PCM - enum AVSampleFormat out_sample_fmt = AV_SAMPLE_FMT_S16; - //输入采样率 - int in_sample_rate = codecCtx->sample_rate; - //输出采样率 - int out_sample_rate = in_sample_rate; - //声道布局（2个声道，默认立体声stereo） - uint64_t in_ch_layout = codecCtx->channel_layout; - //输出的声道布局（立体声） - uint64_t out_ch_layout = AV_CH_LAYOUT_STEREO; + //input sampleFormat + enum AVSampleFormat in_sample_fmt = codecCtx->sample_fmt; + //output sampleFormat: 16bit PCM + enum AVSampleFormat out_sample_fmt = AV_SAMPLE_FMT_S16; + //input sampleRate + int in_sample_rate = codecCtx->sample_rate; + //output sampleRate + int out_sample_rate = in_sample_rate; + //input channel layout(2 channels, stereo by default) + uint64_t in_ch_layout = codecCtx->channel_layout; + //output channel layout + uint64_t out_ch_layout = AV_CH_LAYOUT_STEREO; - swr_alloc_set_opts(swrCtx, - out_ch_layout,out_sample_fmt,out_sample_rate, - in_ch_layout,in_sample_fmt,in_sample_rate, - 0, NULL); - swr_init(swrCtx); + swr_alloc_set_opts(swrCtx, + out_ch_layout, out_sample_fmt, out_sample_rate, + in_ch_layout, in_sample_fmt, in_sample_rate, + 0, NULL); + swr_init(swrCtx); - //输出的声道个数 - int out_channel_nb = av_get_channel_layout_nb_channels(out_ch_layout); + //output channel number + int out_channel_nb = av_get_channel_layout_nb_channels(out_ch_layout); - jclass player_class = (*env)->GetObjectClass(env,thiz); - if(!player_class){ + jclass player_class = (*env)->GetObjectClass(env, thiz); + if (!player_class) { LOGE(TAG, "player_class not found..."); } - //AudioTrack对象 - jmethodID audio_track_method = (*env)->GetMethodID(env,player_class,"createAudioTrack","(II)Landroid/media/AudioTrack;"); - if(!audio_track_method){ + //get AudioTrack by reflection + jmethodID audio_track_method = (*env)->GetMethodID(env, player_class, "createAudioTrack", + "(II)Landroid/media/AudioTrack;"); + if (!audio_track_method) { LOGE(TAG, "audio_track_method not found..."); } - jobject audio_track = (*env)->CallObjectMethod(env,thiz,audio_track_method,out_sample_rate,out_channel_nb); + jobject audio_track = (*env)->CallObjectMethod(env, thiz, audio_track_method, out_sample_rate, + out_channel_nb); - //调用play方法 - jclass audio_track_class = (*env)->GetObjectClass(env,audio_track); - jmethodID audio_track_play_mid = (*env)->GetMethodID(env,audio_track_class,"play","()V"); - (*env)->CallVoidMethod(env,audio_track,audio_track_play_mid); + //call play method + jclass audio_track_class = (*env)->GetObjectClass(env, audio_track); + jmethodID audio_track_play_mid = (*env)->GetMethodID(env, audio_track_class, "play", "()V"); + (*env)->CallVoidMethod(env, audio_track, audio_track_play_mid); - //获取write()方法 - jmethodID audio_track_write_mid = (*env)->GetMethodID(env,audio_track_class,"write","([BII)I"); + //get write method + jmethodID audio_track_write_mid = (*env)->GetMethodID(env, audio_track_class, "write", + "([BII)I"); - //16bit 44100 PCM 数据 - uint8_t *out_buffer = (uint8_t *)av_malloc(MAX_AUDIO_FRAME_SIZE); + uint8_t *out_buffer = (uint8_t *) av_malloc(MAX_AUDIO_FRAME_SIZE); - int got_frame = 0,index = 0, ret; - //不断读取编码数据 - while(av_read_frame(pFormatCtx,packet) >= 0){ - //解码音频类型的Packet - if(packet->stream_index == audio_stream_idx){ - //解码 - ret = avcodec_decode_audio4(codecCtx,frame,&got_frame,packet); - if(ret < 0){ + int got_frame = 0, index = 0, ret; + //read audio frame + while (av_read_frame(pFormatCtx, packet) >= 0) { + //is audio stream index + if (packet->stream_index == audio_stream_idx) { + //do decode + ret = avcodec_decode_audio4(codecCtx, frame, &got_frame, packet); + if (ret < 0) { break; - } - //解码一帧成功 - if(got_frame > 0){ - LOGI(TAG, "decode frame count=%d", index++); - //音频格式转换 - swr_convert(swrCtx, &out_buffer, MAX_AUDIO_FRAME_SIZE,(const uint8_t **)frame->data,frame->nb_samples); + } + //decode success + if (got_frame > 0) { + LOGI(TAG, "decode frame count=%d", index++); + //convert audio format + swr_convert(swrCtx, &out_buffer, MAX_AUDIO_FRAME_SIZE, + (const uint8_t **) frame->data, frame->nb_samples); int out_buffer_size = av_samples_get_buffer_size(NULL, out_channel_nb, - frame->nb_samples, out_sample_fmt, 1); + frame->nb_samples, out_sample_fmt, + 1); - jbyteArray audio_sample_array = (*env)->NewByteArray(env,out_buffer_size); - jbyte* sample_byte_array = (*env)->GetByteArrayElements(env,audio_sample_array,NULL); - //拷贝缓冲数据 + jbyteArray audio_sample_array = (*env)->NewByteArray(env, out_buffer_size); + jbyte *sample_byte_array = (*env)->GetByteArrayElements(env, audio_sample_array, + NULL); + //copy buffer data memcpy(sample_byte_array, out_buffer, (size_t) out_buffer_size); - //释放数组 - (*env)->ReleaseByteArrayElements(env,audio_sample_array,sample_byte_array,0); - //调用AudioTrack的write方法进行播放 - (*env)->CallIntMethod(env,audio_track,audio_track_write_mid, - audio_sample_array,0,out_buffer_size); - //释放局部引用 - (*env)->DeleteLocalRef(env,audio_sample_array); + //release byteArray + (*env)->ReleaseByteArrayElements(env, audio_sample_array, sample_byte_array, 0); + //call write method to play + (*env)->CallIntMethod(env, audio_track, audio_track_write_mid, + audio_sample_array, 0, out_buffer_size); + //delete local reference + (*env)->DeleteLocalRef(env, audio_sample_array); usleep(1000 * 16); - } - } - av_free_packet(packet); - } + } + } + av_free_packet(packet); + } LOGI(TAG, "decode audio finish"); - av_frame_free(&frame); - av_free(out_buffer); - swr_free(&swrCtx); - avcodec_close(codecCtx); - avformat_close_input(&pFormatCtx); - (*env)->ReleaseStringUTFChars(env,input_jstr,input_cstr); + av_frame_free(&frame); + av_free(out_buffer); + swr_free(&swrCtx); + avcodec_close(codecCtx); + avformat_close_input(&pFormatCtx); + (*env)->ReleaseStringUTFChars(env, input_jstr, input_cstr); } \ No newline at end of file