零基础自学做网站,wordpress 图灵机器人,手机视频网站建设,网站自适应源码Planar 模式是 ffmpeg内部存储模式#xff0c;我们实际使用的音频文件都是Packed模式的。 FFmpeq解码不同格式的音频输出的音频采样格式不是一样。 其中AAC解码输出的数据为浮点型的 AV_SAMPLE_FMT_FLTP 格式#xff0c;MP3 解码输出的数据为 AV_SAMPLE_FMT_S16P 格式(使用的…Planar 模式是 ffmpeg内部存储模式我们实际使用的音频文件都是Packed模式的。 FFmpeq解码不同格式的音频输出的音频采样格式不是一样。 其中AAC解码输出的数据为浮点型的 AV_SAMPLE_FMT_FLTP 格式MP3 解码输出的数据为 AV_SAMPLE_FMT_S16P 格式(使用的mp3文件为16位深)。 具体采样格式可以査看解码后的 AVframe 中的 format 成员或解码器的AVCodecContext中的sample_fmt成员。 Planar或者Packed模式直接影响到保存文件时写文件的操作操作数据的时候一定要先检测音频采样格式。
方法1
int encodeAudioInterface(AVCodecContext *encoderCtx, AVFrame *frame, AVPacket *packet, FILE *dest_fp)
{int ret avcodec_send_frame(encoderCtx, frame);if (ret 0){av_log(NULL, AV_LOG_ERROR, send frame to encoder failed:%s\n, av_err2str(ret));ret -1;}while (ret 0){ret avcodec_receive_packet(encoderCtx, packet);if (ret AVERROR(EAGAIN) || ret AVERROR_EOF){av_log(NULL, AV_LOG_WARNING, [encodeAudioInterface] -- AVERROR(EAGAIN) || AVERROR_EOF \n);return 0;}else if (ret 0){av_log(NULL, AV_LOG_ERROR, encode frame failed:%s\n, av_err2str(ret));return -1;}fwrite(packet-data, 1, packet-size, dest_fp);av_packet_unref(packet);}return 0;
}int encodeAudio(const char *inFileName, const char *outFileName)
{int ret 0;/*****************************************************************************/FILE *src_fp fopen(inFileName, rb);if (src_fp NULL){av_log(NULL, AV_LOG_ERROR, open infile:%s failed!\n, inFileName);ret -1;goto end;}FILE *dest_fp fopen(outFileName, wb);if (dest_fp NULL){av_log(NULL, AV_LOG_ERROR, open outfile:%s failed!\n, outFileName);ret -1;goto end;}/*****************************************************************************/AVFrame *frame av_frame_alloc();frame-sample_rate 48 * 1000; // 采样率 - 48Kframe-channels 2;frame-channel_layout AV_CH_LAYOUT_STEREO;frame-format AV_SAMPLE_FMT_S16;frame-nb_samples 1024;// libfdk_aacav_frame_get_buffer(frame, 0);AVCodec *encoder avcodec_find_encoder_by_name(libfdk_aac);if (encoder NULL){av_log(NULL, AV_LOG_ERROR, find encoder failed!\n);ret -1;goto end;}AVCodecContext *encoderCtx avcodec_alloc_context3(encoder);if (encoderCtx NULL){av_log(NULL, AV_LOG_ERROR, alloc encoder context failed!\n);ret -1;goto end;}encoderCtx-sample_fmt frame-format;encoderCtx-sample_rate frame-sample_rate;encoderCtx-channels frame-channels;encoderCtx-channel_layout frame-channel_layout;ret avcodec_open2(encoderCtx, encoder, NULL);if (ret 0){av_log(NULL, AV_LOG_ERROR, open encoder failed:%s\n, av_err2str(ret));goto end;}AVPacket packet;av_init_packet(packet);while (1){// packet L R L R// 2 * 2 * 1024 4096int readSize fread(frame-data[0], 1, frame-linesize[0], src_fp);if (readSize 0){av_log(NULL, AV_LOG_INFO, finish read infile!\n);break;}encodeAudioInterface(encoderCtx, frame, packet, dest_fp);}encodeAudioInterface(encoderCtx, NULL, packet, dest_fp);end:if (frame){av_frame_free(frame);}if (encoderCtx){avcodec_free_context(encoderCtx);return ret;}if (src_fp){fclose(src_fp);}if (dest_fp){fclose(dest_fp);}
}------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
方法 2
int CheckSampleRate(const AVCodec *encoder, const int SampleRate)
{// encoder-supported_samplerates 支持的音频采样数组如果未知则为NULL数组以0结尾const int *SupportSampleRate encoder-supported_samplerates;while (*SupportSampleRate ! 0){av_log(NULL, AV_LOG_DEBUG, [%s] encoder-name: %s, support %d hz -- line:%d\n, __FUNCTION__, encoder-name, *SupportSampleRate, __LINE__); // 受 frame-format等参数影响if (*SupportSampleRate SampleRate){av_log(NULL, AV_LOG_INFO, [%s] This sampling rate is supported by the encoder %d hz -- line:%d\n, __FUNCTION__, *SupportSampleRate, __LINE__); // 受 frame-format等参数影响return 1;}SupportSampleRate;}return 0;
}int CheckChannelLayout(const AVCodec *encoder, const uint64_t ChannelLayout)
{// 支持通道布局的数组如果未知则为NULL。数组以0结尾const uint64_t *SupportsChannelLayout encoder-channel_layouts;if (!SupportsChannelLayout){ // 不是每个AVCodec都给出支持的channel_layoutav_log(NULL, AV_LOG_WARNING, [%s] the encoder %s no set channel_layouts -- line:%d\n, __FUNCTION__, encoder-name, __LINE__);return 1;}while (*SupportsChannelLayout ! 0){av_log(NULL, AV_LOG_DEBUG, [%s] encoder-name: %s, support channel_layout %ld -- line:%d\n, __FUNCTION__, encoder-name, *SupportsChannelLayout, __LINE__); // 受 frame-format等参数影响if (*SupportsChannelLayout ChannelLayout){av_log(NULL, AV_LOG_INFO, [%s] This channel layout is supported by the encoder %d -- line:%d\n, __FUNCTION__, *SupportsChannelLayout, __LINE__); // 受 frame-format等参数影响return 1;}*SupportsChannelLayout;}return 0;
}int CheckSampleFmt(const AVCodec *codecCtx, enum AVSampleFormat SampleFmt)
{ // 数组支持的样例格式如果未知则为NULL数组以-1结尾const enum AVSampleFormat *SampleFmts codecCtx-sample_fmts;while (*SampleFmts ! AV_SAMPLE_FMT_NONE) // 通过 AV_SAMPLE_FMT_NONE 作为结束{if (*SampleFmts SampleFmt){return 1;}*SampleFmts;}return 0;
}void GetAdtsHeader(AVCodecContext *codecCtx, uint8_t *adtsHeader, int aacLength)
{uint8_t freqIdx 0; // 0: 96000HZ 3:48000Hz 4:44100Hzswitch (codecCtx-sample_rate){case 96000:freqIdx 0;break;case 88200:freqIdx 1;break;case 64000:freqIdx 2;break;case 48000:freqIdx 3;break;case 44100:freqIdx 4;break;case 32000:freqIdx 5;break;case 24000:freqIdx 6;break;case 22050:freqIdx 7;break;case 16000:freqIdx 8;break;case 12000:freqIdx 9;break;case 11025:freqIdx 10;break;case 8000:freqIdx 11;break;case 7350:freqIdx 12;break;default:freqIdx 4;break;}uint8_t chanCfg codecCtx-channels;uint32_t frameLength aacLength 7;adtsHeader[0] 0xff;adtsHeader[1] 0xF1;adtsHeader[2] ((codecCtx-profile) 6) (freqIdx 2) (chanCfg 2);adtsHeader[3] (((chanCfg 3) 6) (frameLength 11));adtsHeader[4] ((frameLength 0x7FF) 3);adtsHeader[5] (((frameLength 7) 5) 0x1F);adtsHeader[6] 0xFC;
}int decodeAudioInterface(AVCodecContext *codecCtx, AVFrame *frame, AVPacket *pkt, FILE *output)
{int ret avcodec_send_frame(codecCtx, frame);if (ret 0){av_log(NULL, AV_LOG_ERROR, [%s] sending the frame to the encoder error! -- line:%d\n, __FUNCTION__, __LINE__);return -1;}// 编码和解码都是一样的都是send 1次然后 receive 多次直到AVERROR(EAGAIN)或者AVERROR_EOFwhile (ret 0){ret avcodec_receive_packet(codecCtx, pkt);if (ret AVERROR(EAGAIN) || ret AVERROR_EOF){return 0;}else if (ret 0){av_log(NULL, AV_LOG_ERROR, [%s] encoding audio frameerror! -- line:%d\n, __FUNCTION__, __LINE__);return -1;}uint8_t aacHeader[7];GetAdtsHeader(codecCtx, aacHeader, pkt-size);size_t len fwrite(aacHeader, 1, 7, output);if (len ! 7){av_log(NULL, AV_LOG_ERROR, [%s] fwrite aac_header failed! -- line:%d\n, __FUNCTION__, __LINE__);return -1;}len fwrite(pkt-data, 1, pkt-size, output);if (len ! pkt-size){av_log(NULL, AV_LOG_ERROR, [%s] fwrite aac data failed! -- line:%d\n, __FUNCTION__, __LINE__);return -1;}}return -1;
}void f32leConvert2fltp(float *f32le, float *fltp, int nb_samples)
{float *fltp_l fltp; // 左通道float *fltp_r fltp nb_samples; // 右声道for (int i 0; i nb_samples; i){fltp_l[i] f32le[i * 2]; // 1 0 - 2 3fltp_r[i] f32le[i * 2 1]; // 可以尝试注释左声道或者右声道听听声音}
}int decodeAudio(const char *pcmFileName, const char *aacFileName, const char *encoderName)
{FILE *pcmfile fopen(pcmFileName, rb);FILE *aacfile fopen(aacFileName, wb);if (pcmfile NULL || aacfile NULL){av_log(NULL, AV_LOG_ERROR, [%s] open %s or %s file failed -- line:%d \n, __FUNCTION__, aacFileName, pcmFileName, __LINE__);goto _end;}const AVCodec *encoder NULL;if (encoderName ! NULL (strcmp(encoderName, libfdk_aac) 0 || strcmp(encoderName, aac) 0)) // encoderName 如果制定了编码器{encoder avcodec_find_encoder_by_name(encoderName); // 设置为指定编码器av_log(NULL, AV_LOG_INFO, [%s] force codec name: %s -- line:%d\n, __FUNCTION__, encoderName, __LINE__);}else{encoder avcodec_find_encoder(AV_CODEC_ID_AAC);av_log(NULL, AV_LOG_INFO, [%s] default codec name: %s -- line:%d\n, __FUNCTION__, aac, __LINE__);}if (encoder NULL){av_log(NULL, AV_LOG_ERROR, [%s] Codec found error! -- line:%d\n, __FUNCTION__, __LINE__);goto _end;}// 创建编码器上下文AVCodecContext *codecCtx avcodec_alloc_context3(encoder);if (codecCtx NULL){av_log(NULL, AV_LOG_ERROR, [%s] Conld not allocate audio codec context -- line:%d\n, __FUNCTION__, __LINE__);goto _end;}codecCtx-codec_id AV_CODEC_ID_AAC;codecCtx-codec_type AVMEDIA_TYPE_AUDIO;codecCtx-bit_rate 128 * 1024;codecCtx-channel_layout AV_CH_LAYOUT_STEREO;codecCtx-sample_rate 48000;codecCtx-channels av_get_channel_layout_nb_channels(codecCtx-channel_layout);codecCtx-profile FF_PROFILE_AAC_LOW;if (strcmp(encoder-name, libfdk_aac) 0){codecCtx-sample_fmt AV_SAMPLE_FMT_S16;}else{codecCtx-sample_fmt AV_SAMPLE_FMT_FLTP;}// 检测采样格式的支持情况if (!CheckSampleFmt(encoder, codecCtx-sample_fmt)){av_log(NULL, AV_LOG_ERROR, [%s] Encoder does not support sample format %s -- line:%d\n, __FUNCTION__, av_get_sample_fmt_name(codecCtx-sample_fmt), __LINE__);goto _end;}if (!CheckSampleRate(encoder, codecCtx-sample_rate)){av_log(NULL, AV_LOG_ERROR, [%s] Encoder does not support sample rate codecCtx-sample_rate:%d -- line:%d\n, __FUNCTION__, codecCtx-sample_rate, __LINE__);goto _end;}if (!CheckChannelLayout(encoder, codecCtx-channel_layout)){av_log(NULL, AV_LOG_ERROR, [%s] Encoder does not support sample channel_layout %lu -- line:%d\n, __FUNCTION__, codecCtx-channel_layout, __LINE__);goto _end;}av_log(NULL, AV_LOG_INFO, \n[%s] ------------------------ Audio encode config ------------------------ line:%d \n, __FUNCTION__, __LINE__);av_log(NULL, AV_LOG_INFO, [%s] codecCtx-bit_rate: %ld kbps -- line:%d\n, __FUNCTION__, codecCtx-bit_rate / 1024, __LINE__);av_log(NULL, AV_LOG_INFO, [%s] codecCtx-sample_rate: %d -- line:%d\n, __FUNCTION__, codecCtx-sample_rate, __LINE__);av_log(NULL, AV_LOG_INFO, [%s] codecCtx-sample_fmt: %s -- line:%d\n, __FUNCTION__, av_get_sample_fmt_name(codecCtx-sample_fmt), __LINE__);av_log(NULL, AV_LOG_INFO, [%s] codecCtx-channels: %d -- line:%d\n, __FUNCTION__, codecCtx-channels, __LINE__);// frame_size 是在 av_coedc_open2后进行关联av_log(NULL, AV_LOG_INFO, [%s] Before frame size %d -- line:%d\n, __FUNCTION__, codecCtx-frame_size, __LINE__);if (avcodec_open2(codecCtx, encoder, NULL) 0){av_log(NULL, AV_LOG_ERROR, [%s] Could not open codec -- line:%d\n, __FUNCTION__, __LINE__);goto _end;}av_log(NULL, AV_LOG_INFO, [%s] Once frame_size %d -- line:%d\n\n, __FUNCTION__, codecCtx-frame_size, __LINE__); // 决定每次送多少个采样点AVPacket *packet av_packet_alloc();if (!packet){av_log(NULL, AV_LOG_ERROR, [%s] packet alloc error! -- line:%d \n, __FUNCTION__, __LINE__);goto _end;}AVFrame *frame av_frame_alloc();if (!frame){av_log(NULL, AV_LOG_ERROR, [%s] Could not allocate audio frame -- line:%d\n, __FUNCTION__, __LINE__);goto _end;}frame-nb_samples codecCtx-frame_size;frame-format codecCtx-sample_fmt;frame-channel_layout codecCtx-channel_layout;frame-channels av_get_channel_layout_nb_channels(frame-channel_layout);av_log(NULL, AV_LOG_INFO, [%s] frame nb_samples: %d -- line:%d\n, __FUNCTION__, frame-nb_samples, __LINE__);av_log(NULL, AV_LOG_INFO, [%s] frame sample_fmt: %s -- line:%d\n, __FUNCTION__, av_get_sample_fmt_name(frame-format), __LINE__);av_log(NULL, AV_LOG_INFO, [%s] frame channel_layout: %lu -- line:%d\n, __FUNCTION__, frame-channel_layout, __LINE__);// 为frame分配bufferint ret av_frame_get_buffer(frame, 0);if (ret 0){av_log(NULL, AV_LOG_ERROR, [%s] Could not allocate audio data buffers -- line:%d\n, __FUNCTION__, __LINE__);goto _end;}// 计算出每一帧的数据 单个采样点的字节 * 通道数目 * 每帧采样点的数量size_t FrameByteSize av_get_bytes_per_sample(frame-format) * frame-channels * frame-nb_samples;av_log(NULL, AV_LOG_INFO, [%s] frame_bytes %ld frame-channels %d frame-nb_samples %d -- line:%d\n, __FUNCTION__, FrameByteSize, frame-channels, frame-nb_samples, __LINE__);uint8_t *pcmBuf (uint8_t *)malloc(FrameByteSize);uint8_t *pcmBufTemp (uint8_t *)malloc(FrameByteSize);if (!pcmBuf || !pcmBufTemp){av_log(NULL, AV_LOG_ERROR, [%s] pcmBuf or pcmBufTemp malloc failed -- line:%d\n, __FUNCTION__, __LINE__);goto _end;}memset(pcmBuf, 0, FrameByteSize);memset(pcmBufTemp, 0, FrameByteSize);int64_t pts 0;av_log(NULL, AV_LOG_INFO, \n[%s] ------------------------ start enode ------------------------ line:%d \n, __FUNCTION__, __LINE__);while (1){memset(pcmBuf, 0, FrameByteSize);size_t ReadByteSize fread(pcmBuf, 1, FrameByteSize, pcmfile);if (ReadByteSize 0){av_log(NULL, AV_LOG_INFO, [%s] read file finish -- line:%d \n, __FUNCTION__, __LINE__);break;}/*确保该 frame 可写, 如果编码器内部保持了内存参数计数则需要重新拷贝一个备份目的是新写入的数据和编码器保存的数据不能产生冲突*/ret av_frame_make_writable(frame);if (ret ! 0){av_log(NULL, AV_LOG_ERROR, [%s] av_frame_make_writable failed!!! -- line:%d\n, __FUNCTION__, __LINE__);}if (AV_SAMPLE_FMT_S16 frame-format){// 将读取到的PCM数据填充到frame去但是要注意匹配格式planner | packet ret av_samples_fill_arrays(frame-data, frame-linesize, pcmBuf, frame-channels, frame-nb_samples, frame-format, 0);}else{// 将读取到的PCM数据填充到frame去但是要注意匹配格式planner | packet // 将本地的f32le packed 模式的数据转为 float palannermemset(pcmBufTemp, 0, FrameByteSize);f32leConvert2fltp((float *)pcmBuf, (float *)pcmBufTemp, frame-nb_samples);ret av_samples_fill_arrays(frame-data, frame-linesize, pcmBufTemp, frame-channels, frame-nb_samples, frame-format, 0);}// 设置 ptspts frame-nb_samples;frame-pts pts; // 使用采样率作为pts的单位具体换算成秒 pts*1/采样率ret decodeAudioInterface(codecCtx, frame, packet, aacfile);if (ret 0){av_log(NULL, AV_LOG_ERROR, [%s] encode failed -- line:%d\n, __FUNCTION__, __LINE__);break;}}/*冲刷编码器*/decodeAudioInterface(codecCtx, NULL, packet, aacfile);
_end:if (aacfile){fclose(aacfile);}if (pcmfile){fclose(pcmfile);}if (pcmBuf){free(pcmBuf);}if (pcmBufTemp){free(pcmBufTemp);}if (packet){av_packet_free(packet);}if (frame){av_frame_free(frame);}if (codecCtx){avcodec_free_context(codecCtx);}return ret;
}对于 flush encoder 的操作: 编码器通常的冲洗方法:调用一次 avcodec_send_frame(NULL)(返回成功) 然后不停调用 avcodec_receive_packet() 直到其返回 AVERROR_EOF取出所有缓存帧, avcodec_receive_packet() 返回 AVERROR EOF 这一次是没有有效数据的仅仅获取到一个结束标志
