11 - FFmpeg - 编码 AAC

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