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----------------------------------------------------------------- PCM介绍 ----------------------------------------------------------------
PCM(Pulse Code Modulation)，脉冲编码调制，是一种用数字表示采样模拟信号的方法。
核心过程:采样-->量化-->编码
----------------------------------------------------------------- PCM关键要素 -----------------------------------------------------------------
·采样率(sampleRate):每秒中采集样本的个数，如8KHz，表示每秒采样8000次。
奈奎斯特定理，明确:按比声音最高频率高2倍以上的频率对声音进行采样;
人耳能接受的频率范围为20Hz～20kHz，故采样率一般为44.1KHz较好，采样率越高，质量越高，但存储空间增大。
·量化格式(sampleFormat) : ffmpeg支持的量化格式: ffmpeg -formats | grep PCM
声道数"(channel):单声道(mono)、双声道(stereo)
----------------------------------------------------------------- PCM数据格式 -----------------------------------------------------------------
存储格式:
> 双声道音频文件，采样数据按LRLR方式存储，存储的时候与字节序有关。
> 单声道音频文件，采样数据按时间先后依次存入(有时也会用LRLR方式存储，但另一个声道数据为0)。
· 存储格式分为Packed和Planner两种，对于双通道音频，Packed为两个声道的数据交错存储;Planner 为两个声道数据分开存储:
> Packed: LRLRLR
> Planner: LLLRRR
----------------------------------------------------------------- PCM计算 -----------------------------------------------------------------
· 大小计算:以CD的音质为例:量化格式为16比特(2字节)，采样率为44100，声道数为2
比特率为:44100*16*2=1378.125kbps
1分钟音频大小:1378.125 * 60/8/1024=10.09MB
· ffmpeg提取pcm数据命令:
ffmpeg -i break.aac -ar 48000 -ac 2 -f s16le out.pcm
· ffplay播放pcm数据:
ffplay -ar 48000 -ac 2 -f s16le out.pcm
方法一
int decodeAudioInterface(AVCodecContext *decoderCtx, AVPacket *packet, AVFrame *frame, FILE *dest_fp) { int ret = avcodec_send_packet(decoderCtx, packet); if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "send packet to decoder failed:%s\n", av_err2str(ret)); return -1; } while (ret >= 0) { ret = avcodec_receive_frame(decoderCtx, frame); if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) { av_log(NULL, AV_LOG_WARNING, "[decodeAudioInterface] -- AVERROR(EAGAIN) || AVERROR_EOF \n", av_err2str(ret)); return 0; } else if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "decode packet failed:%s\n", av_err2str(ret)); } // frame -- fltp int dataSize = av_get_bytes_per_sample(decoderCtx->sample_fmt); if (dataSize < 0) { av_log(NULL, AV_LOG_ERROR, "get bytes failed!\n"); return -1; } for (int i = 0; i < frame->nb_samples; i++) { for (int channel = 0; channel < decoderCtx->channels; channel++) { fwrite(frame->data[channel] + dataSize * i, 1, dataSize, dest_fp); } } } return 0; } int decodeAudio(const char *inFileName, const char *outFileName) { /********************************************************************/ FILE *dest_fp = fopen(outFileName, "wb+"); if (dest_fp == NULL) { av_log(NULL, AV_LOG_ERROR, "open outfile %s failed!\n", outFileName); goto end; } /********************************************************************/ AVFormatContext *inFmtCtx = NULL; int ret = avformat_open_input(&inFmtCtx, inFileName, NULL, NULL); if (ret != 0) { av_log(NULL, AV_LOG_ERROR, "open input file failed:%s\n", av_err2str(ret)); return -1; } ret = avformat_find_stream_info(inFmtCtx, NULL); if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "find stream info failed:%s\n", av_err2str(ret)); goto end; } int audioIndex = av_find_best_stream(inFmtCtx, AVMEDIA_TYPE_AUDIO, -1, -1, NULL, 0); if (audioIndex < 0) { av_log(NULL, AV_LOG_ERROR, "find bast stream failed:%s\n", av_err2str(audioIndex)); goto end; } AVCodecContext *decoderCtx = avcodec_alloc_context3(NULL); if (decoderCtx == NULL) { av_log(NULL, AV_LOG_ERROR, "avcodec alloc context failed\n"); ret = -1; goto end; } // 拷贝编码参数 avcodec_parameters_to_context(decoderCtx, inFmtCtx->streams[audioIndex]->codecpar); AVCodec *decoder = avcodec_find_decoder(decoderCtx->codec_id); if (decoder == NULL) { av_log(NULL, AV_LOG_ERROR, "find decoder %d failed!\n", decoderCtx->codec_id); } ret = avcodec_open2(decoderCtx, decoder, NULL); if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "open decoder failed:%s\n", av_err2str(ret)); goto end; } AVPacket packet; av_init_packet(&packet); AVFrame *frame = av_frame_alloc(); int frameSize = av_samples_get_buffer_size(NULL, decoderCtx->channels, frame->nb_samples, decoderCtx->sample_fmt, 1); uint8_t *frameBuffer = av_malloc(frameSize); avcodec_fill_audio_frame(frame, decoderCtx->channels, decoderCtx->sample_fmt, frameBuffer, frameSize, 1); while (av_read_frame(inFmtCtx, &packet) >= 0) { if (packet.stream_index == audioIndex) { decodeAudioInterface(decoderCtx, &packet, frame, dest_fp); } av_packet_unref(&packet); } decodeAudioInterface(decoderCtx, NULL, frame, dest_fp); end: if (inFmtCtx) { avformat_close_input(&inFmtCtx); } if (decoderCtx) { avcodec_free_context(&decoderCtx); } if (frame) { av_frame_free(&frame); } if (frameBuffer) { av_freep(frameBuffer); } if (dest_fp) { fclose(dest_fp); } return 0; } ====================================================================================
方法二、
#define AUDIO_INBUF_SIZE 20480 #define AUDIO_REFLT_THRESH 4096 int decodeAudioInterface(AVCodecContext *decoderCtx, AVPacket *packet, AVFrame *frame, FILE *dest_fp) { int ret = avcodec_send_packet(decoderCtx, packet); if (ret == AVERROR(EAGAIN)) { av_log(NULL, AV_LOG_WARNING, "[decodeAudioInterface] -- AVERROR(EAGAIN) \n"); } else if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "send packet to decoder failed: %s\n", av_err2str(ret)); return -1; } while (ret >= 0) { // 对于frame avcodec_receive_frame 内部每次都先调用 ret = avcodec_receive_frame(decoderCtx, frame); if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) { return; } else if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "decode packet failed:%s\n", av_err2str(ret)); } // 获取单个sample 占用的字节 size_t dataSize = av_get_bytes_per_sample(decoderCtx->sample_fmt); if (dataSize < 0) { /*This should not occur,checking just for paranoia*/ av_log(NULL, AV_LOG_ERROR, "get bytes failed!\n"); return -1; } av_log(NULL, AV_LOG_INFO,"采样率: %uHZ, 通道：%u, 编码格式：%u \n", frame->sample_rate,frame->channels, frame->format); for (int i = 0; i < frame->nb_samples; i++) { for (int channel = 0; channel < decoderCtx->channels; channel++) // 交错的方式写入，float的格式输出 { fwrite(frame->data[channel] + dataSize * i, 1, dataSize, dest_fp); } } } } // ffplay -ar 48000 -ac 2 -f f32le outTest.pcm int decodeAudio(const char *inFileName, const char *outFileName) { /************************************************************************************/ // 打开输入文件 FILE *inFile = fopen(inFileName, "rb"); if (!inFile) { av_log(NULL, AV_LOG_ERROR, "Could not open:%s\n", inFileName); goto _end; } // 打开输出文件 FILE *outFile = fopen(outFileName, "wb"); if (!outFile) { av_log(NULL, AV_LOG_ERROR, "Could not open:%s\n", inFileName); goto _end; } uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE]; uint8_t *data = inbuf; size_t dataSize = fread(inbuf, 1, AUDIO_INBUF_SIZE, inFile); /************************************************************************************/ enum AVCodecID AudioCodecID = AV_CODEC_ID_AAC; if (strstr(inFileName, "aac") != NULL) { AudioCodecID = AV_CODEC_ID_AAC; } else if (strstr(inFileName, "mp3") != NULL) { AudioCodecID = AV_CODEC_ID_MP3; } else { av_log(NULL, AV_LOG_WARNING, "default codec id:%d\n", AudioCodecID); } // 查找解码器 const AVCodec *decoder; decoder = avcodec_find_decoder(AudioCodecID); // AV_CODEC_ID_AAC if (!decoder) { av_log(NULL, AV_LOG_ERROR, "decoder not found\n"); goto _end; } // 获得裸流的解析器 -- 根据制定的解码器ID初始化相应裸流的解析器 AVCodecParserContext *parserCtx = av_parser_init(decoder->id); if (!parserCtx) { av_log(NULL, AV_LOG_ERROR, "parserCtx not found\n"); goto _end; } AVCodecContext *decoderCtx = NULL; decoderCtx = avcodec_alloc_context3(decoder); if (!decoderCtx) { av_log(NULL, AV_LOG_ERROR, "Conld not allocate audio codec context\n"); goto _end; } // 将解码器和解码器上下文进行关联 int ret = avcodec_open2(decoderCtx, decoder, NULL); if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "Could not open codec\n"); goto _end; } AVFrame *decodeFrame = av_frame_alloc(); if (decodeFrame == NULL) { av_log(NULL, AV_LOG_ERROR, "Could not allocate audio decodeFrame\n"); goto _end; } AVPacket *packet = NULL; packet = av_packet_alloc(); while (dataSize > 0) { ret = av_parser_parse2(parserCtx, decoderCtx, &packet->data, &packet->size, data, dataSize, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0); if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "Error while parsing\n"); goto _end; } data += ret; // 跳过已经解析的数据 dataSize -= ret; // 对应的缓存大小也做相应的减小 if (packet->size) decodeAudioInterface(decoderCtx, packet, decodeFrame, outFile); if (dataSize < AUDIO_REFLT_THRESH) // 如果数据少了则再次读取 { memmove(inbuf, data, dataSize); // 把之前剩的数据拷贝到 buffer 的其实位置 data = inbuf; // 读取数据 长度： AUDIO_INBUF_SIZE - dataSize int len = fread(data + dataSize, 1, AUDIO_INBUF_SIZE - dataSize, inFile); if (len > 0) dataSize += len; } } _end: if (outFile) fclose(outFile); if (inFile) fclose(inFile); if (decoderCtx) avcodec_free_context(&decoderCtx); if (parserCtx) av_parser_close(parserCtx); if (decodeFrame) av_frame_free(&decodeFrame); if (packet) av_packet_free(&packet); return 0; }