【FFmpeg】PCM编码成AAC

使用FFmpeg把PCM裸数据编码成AAC音频流，具体步骤跟YUV编码成H264差不多。

1、命令行

ffmpeg -f s16le -ar 44100 -ac 2 -i bb1.pcm output.aac

-f PCM数据为s16le

-ar 采样率为44100

-ac 通道数为2

这样就通过命令把PCM数据编码成AAC了。

2、使用API编码

FFmpeg内部AAC格式只支持AV_SAMPLE_FMT_FLTP格式的PCM，由于我们的PCM数据是s16le的，因此我们需要把s16le格式转换成fltp格式再进行编码。我们可以在AVCodec结构体中的sample_fmts字段中判断编码器是否支持你的格式。

初始化输出文件上下文
```
int avformat_alloc_output_context2(AVFormatContext **ctx, ff_const59 AVOutputFormat *oformat,const char *format_name, const char *filename);
```
ctx 输出文件的上下文

oformat 输出文件的AVOutputFormat，传NULL，FFmpeg会根据filename的格式初始化oformat

format_name 输出文件的格式，传NULL，FFmpeg会根据filename的格式初始化format_name

filename 输出文件路径

初始化编码器上下文

dec = avcodec_find_encoder(ofmt_ctx->oformat->audio_codec);
if (!dec) {printf("avcodec_find_encoder fail \n");goto __FAIL;
}
dec_ctx = avcodec_alloc_context3(dec);
dec_ctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (!check_sample_fmt(dec, dec_ctx->sample_fmt)) {fprintf(stderr, "Encoder does not support sample format %s",av_get_sample_fmt_name(dec_ctx->sample_fmt));goto __FAIL;
}
dec_ctx->channel_layout = select_channel_layout(dec);
dec_ctx->channels = av_get_channel_layout_nb_channels(dec_ctx->channel_layout);
dec_ctx->sample_rate = select_sample_rate(dec);
dec_ctx->bit_rate = 64000;
ret = avcodec_open2(dec_ctx, dec, NULL);

FFmpeg内部AAC音频流只支持fltp格式的PCM，使用check_sample_fmt函数可以检测编码器是否支持AV_SAMPLE_FMT_FLTP，通过select_channel_layout函数选择最佳的音频通道布局，通过select_sample_rate函数选择最佳的采样率。

检测是否支持AVSampleFormat

static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt)
{const enum AVSampleFormat *p = codec->sample_fmts;
while (*p != AV_SAMPLE_FMT_NONE) {if (*p == sample_fmt)return 1;p++;}return 0;
}

选择最佳采样率

static int select_sample_rate(const AVCodec *codec)
{const int *p;int best_samplerate = 0;
if (!codec->supported_samplerates)return 44100;
p = codec->supported_samplerates;while (*p) {if (!best_samplerate || abs(44100 - *p) < abs(44100 - best_samplerate))best_samplerate = *p;p++;}return best_samplerate;
}

选择最佳通道布局

static int select_channel_layout(const AVCodec *codec)
{const uint64_t *p;uint64_t best_ch_layout = 0;int best_nb_channels   = 0;
if (!codec->channel_layouts)return AV_CH_LAYOUT_STEREO;
p = codec->channel_layouts;while (*p) {int nb_channels = av_get_channel_layout_nb_channels(*p);
if (nb_channels > best_nb_channels) {best_ch_layout    = *p;best_nb_channels = nb_channels;}p++;}return best_ch_layout;
}

创建输入文件音频流

AVStream *st = avformat_new_stream(ofmt_ctx, dec);
ret = avcodec_parameters_from_context(st->codecpar, dec_ctx);
if (ret<0) {printf("avcodec_parameters_from_context fail \n");goto __FAIL;
}

把编码器上下文参数拷贝给新建的AVSteam

打开输出文件

avio_open(&ofmt_ctx->pb, aacPath.UTF8String, AVIO_FLAG_WRITE);

写入文件头
```
avformat_write_header(ofmt_ctx, NULL);
```

读取PCM数据，放到AVFrame中

初始化AVFrame用来存放PCM数据

AVFrame *s16_frame = av_frame_alloc();
if (!s16_frame) {printf("av_frame_alloc fail \n");goto __FAIL;
}
s16_frame->nb_samples = dec_ctx->frame_size;
s16_frame->format = AV_SAMPLE_FMT_S16;
s16_frame->channel_layout = AV_CH_LAYOUT_STEREO;
s16_frame->sample_rate = 44100;
ret = av_frame_get_buffer(s16_frame, 0);

AVFrame的参数要与你的PCM数据参数一致，我用到的PCM数据是s16le、采样率44100Hz、通道数为2。

从文件中读取PCM数据

size_t size = fread(pcm_buffer, 1, pcm_buffer_size, pcm_f);

存放到AVFrame中去

av_samples_fill_arrays(s16_frame->data, s16_frame->linesize, pcm_buffer, s16_frame->channels, s16_frame->nb_samples, s16_frame->format, 0);
int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,const uint8_t *buf,int nb_channels, int nb_samples,enum AVSampleFormat sample_fmt, int align);

audio_data 输出buffer，传frame->data即可

linesize 输出buffer的行大小，传frame->linesize即可

buf 音频数据

nb_channels 音频通道数

nb_samples 音频采样数

sample_fmt 音频数据格式

align buffer的对齐方式默认为0，不对齐传1

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s16le->fltp格式转换
- 创建SwrContext
```
struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,int log_offset, void *log_ctx);
```
  s 传NULL即可，会自动分配空间创建SwrContext
  
  in_ch_layout out_ch_layout 输入、输出的通道布局
  
  in_sample_fmt out_sample_fmt 输入、输出的PCM数据格式
  
  in_sample_rate out_sample_rate 输入、输出的采样率
- 初始化SwrContext
```
int swr_init(struct SwrContext *s);
```
- 格式转换
```
int swr_convert(struct SwrContext *s, uint8_t **out, int out_count,const uint8_t **in , int in_count);
```
  in out 输入、输出的buffer
  
  in_count out_count 输入、输出的采样数，需要注意的是，这里传的是一个通道的采样数，而不是多个通道数相加的。

编码

int avcodec_send_frame(AVCodecContext *avctx, const AVFrame *frame);
int avcodec_receive_packet(AVCodecContext *avctx, AVPacket *avpkt);

写文件尾

int av_write_trailer(AVFormatContext *s);

完整代码如下

/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt)
{const enum AVSampleFormat *p = codec->sample_fmts;
while (*p != AV_SAMPLE_FMT_NONE) {if (*p == sample_fmt)return 1;p++;}return 0;
}

/* just pick the highest supported samplerate */
static int select_sample_rate(const AVCodec *codec)
{const int *p;int best_samplerate = 0;
if (!codec->supported_samplerates)return 44100;
p = codec->supported_samplerates;while (*p) {if (!best_samplerate || abs(44100 - *p) < abs(44100 - best_samplerate))best_samplerate = *p;p++;}return best_samplerate;
}

/* select layout with the highest channel count */
static int select_channel_layout(const AVCodec *codec)
{const uint64_t *p;uint64_t best_ch_layout = 0;int best_nb_channels   = 0;
if (!codec->channel_layouts)return AV_CH_LAYOUT_STEREO;
p = codec->channel_layouts;while (*p) {int nb_channels = av_get_channel_layout_nb_channels(*p);
if (nb_channels > best_nb_channels) {best_ch_layout    = *p;best_nb_channels = nb_channels;}p++;}return best_ch_layout;
}


+ (void)convert
{NSString *pcmPath = [[NSBundle mainBundle] pathForResource:@"bb1_44100_2_s16le.pcm" ofType:nil];NSString *aacPath = [NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES).firstObject stringByAppendingPathComponent:@"bb1.aac"];
NSLog(@"%@", aacPath);int ret;AVFormatContext *ofmt_ctx = NULL;AVCodecContext *dec_ctx = NULL;AVCodec *dec = NULL;AVPacket *pkt = NULL;AVFrame *s16_frame = NULL;AVFrame *fltp_frame = NULL;SwrContext *swr_ctx = NULL;FILE *pcm_f = fopen(pcmPath.UTF8String, "rb+");ret = avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, aacPath.UTF8String);if (ret<0) {printf("avformat_alloc_output_context2 fail \n");goto __FAIL;}dec = avcodec_find_encoder(ofmt_ctx->oformat->audio_codec);if (!dec) {printf("avcodec_find_encoder fail \n");goto __FAIL;}dec_ctx = avcodec_alloc_context3(dec);dec_ctx->sample_fmt = AV_SAMPLE_FMT_FLTP;if (!check_sample_fmt(dec, dec_ctx->sample_fmt)) {fprintf(stderr, "Encoder does not support sample format %s",av_get_sample_fmt_name(dec_ctx->sample_fmt));goto __FAIL;}dec_ctx->channel_layout = select_channel_layout(dec);dec_ctx->channels = av_get_channel_layout_nb_channels(dec_ctx->channel_layout);dec_ctx->sample_rate = select_sample_rate(dec);dec_ctx->bit_rate = 64000;
ret = avio_open(&ofmt_ctx->pb, aacPath.UTF8String, AVIO_FLAG_WRITE);if (ret<0) {printf("avio_open fail \n");goto __FAIL;}ret = avcodec_open2(dec_ctx, dec, NULL);if (ret<0) {printf("avcodec_open2 fail \n");goto __FAIL;}AVStream *st = avformat_new_stream(ofmt_ctx, dec);ret = avcodec_parameters_from_context(st->codecpar, dec_ctx);if (ret<0) {printf("avcodec_parameters_from_context fail \n");goto __FAIL;}ret = avformat_write_header(ofmt_ctx, NULL);if (ret<0) {printf("avformat_write_header fail \n");goto __FAIL;}s16_frame = av_frame_alloc();if (!s16_frame) {printf("av_frame_alloc fail \n");goto __FAIL;}s16_frame->nb_samples = dec_ctx->frame_size;s16_frame->format = AV_SAMPLE_FMT_S16;s16_frame->channel_layout = AV_CH_LAYOUT_STEREO;s16_frame->sample_rate = 44100;
//    s16_frame->channels = av_get_channel_layout_nb_channels(s16_frame->channel_layout);ret = av_frame_get_buffer(s16_frame, 0);if (ret<0) {printf("av_frame_get_buffer fail \n");goto __FAIL;}pkt = av_packet_alloc();if (!pkt) {printf("av_packet_alloc fail \n");goto __FAIL;}int pts_i = 0;swr_ctx = swr_alloc_set_opts(NULL, dec_ctx->channel_layout, dec_ctx->sample_fmt, dec_ctx->sample_rate, s16_frame->channel_layout, s16_frame->format, s16_frame->sample_rate, 0, NULL);if (!swr_ctx) {printf("swr_alloc_set_opts fail \n");goto __FAIL;}ret = swr_init(swr_ctx);if (ret<0) {printf("swr_init fail \n");goto __FAIL;}fltp_frame = av_frame_alloc();fltp_frame->nb_samples = dec_ctx->frame_size;fltp_frame->format = dec_ctx->sample_fmt;fltp_frame->channel_layout = dec_ctx->channel_layout;fltp_frame->sample_rate = dec_ctx->sample_rate;
//    fltp_frame->channels = av_get_channel_layout_nb_channels(s16_frame->channel_layout);ret = av_frame_get_buffer(fltp_frame, 0);if (ret<0) {printf("av_frame_get_buffer fail \n");goto __FAIL;}uint64_t pcm_buffer_size = s16_frame->nb_samples*av_get_bytes_per_sample(s16_frame->format)*s16_frame->channels;uint8_t *pcm_buffer = av_malloc(pcm_buffer_size);
while (feof(pcm_f)==0) {size_t size = fread(pcm_buffer, 1, pcm_buffer_size, pcm_f);int nb_samples = size/(av_get_bytes_per_sample(s16_frame->format)*s16_frame->channels);s16_frame->nb_samples = nb_samples;fltp_frame->nb_samples = nb_samples;av_samples_fill_arrays(s16_frame->data, s16_frame->linesize, pcm_buffer, s16_frame->channels, s16_frame->nb_samples, s16_frame->format, 0);ret = swr_convert(swr_ctx, fltp_frame->data, fltp_frame->nb_samples, s16_frame->data, s16_frame->nb_samples);if (size==0) {printf("fread fail \n");break;}pts_i+=fltp_frame->nb_samples;fltp_frame->pts = pts_i;ret = avcodec_send_frame(dec_ctx, fltp_frame);if (ret<0) {printf("avcodec_send_frame fail \n");break;}while (1) {ret = avcodec_receive_packet(dec_ctx, pkt);if (ret==AVERROR(EAGAIN) || ret == AVERROR_EOF) {break;} else if (ret<0) {printf("avcodec_receive_packet fail \n");break;}ret = av_interleaved_write_frame(ofmt_ctx, pkt);if (ret<0) {printf("av_interleaved_write_frame fail \n");break;}av_packet_unref(pkt);}}ret = avcodec_send_frame(dec_ctx, NULL);if (ret<0) {printf("avcodec_send_frame fail \n");goto __FAIL;}while (1) {ret = avcodec_receive_packet(dec_ctx, pkt);if (ret==AVERROR(EINVAL) || ret == AVERROR_EOF) {break;} else if (ret<0) {printf("avcodec_receive_packet fail \n");break;}ret = av_interleaved_write_frame(ofmt_ctx, pkt);if (ret<0) {printf("av_interleaved_write_frame fail \n");break;}av_packet_unref(pkt);}ret = av_write_trailer(ofmt_ctx);if (ret<0) {printf("av_write_trailer fail \n");}
__FAIL:if (ofmt_ctx->pb) {avio_close(ofmt_ctx->pb);}if (dec_ctx) {avcodec_close(dec_ctx);}if (pcm_buffer) {av_free(pcm_buffer);}if (ofmt_ctx) {avformat_free_context(ofmt_ctx);}if (s16_frame) {av_frame_free(&s16_frame);}if (fltp_frame) {av_frame_free(&fltp_frame);}if (pkt) {av_packet_free(&pkt);}
}

原文链接：【FFmpeg】PCM编码成AAC - 简书