G711/G726编解码示例

概述

本文基于ffmpeg，演示g711a，g711u，g726，g726le的编码以及编码后的文件如何播放测试。

g711本质还是pcm，不过是将16位带符号pcm数据压缩为8位pcm数据。g711a是取s16le格式的高13位；g711u则是取s16le格式的高14位。实际压缩比是2:1，具体压缩算法可参考如下博客：https://blog.csdn.net/q2519008/article/details/80900838。

编码代码

encode_audio.c ：

/**
 * func: g711a/g711u/g726/g726le encoder.
 * author: Francis Fan
 */

#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>

#include <libavcodec/avcodec.h>

#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/frame.h>
#include <libavutil/samplefmt.h>

/* 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;
}

static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
                   FILE *output)
{
    int ret;

    /* send the frame for encoding */
    ret = avcodec_send_frame(ctx, frame);
    if (ret < 0) {
        fprintf(stderr, "Error sending the frame to the encoder\n");
        exit(1);
    }

    /* read all the available output packets (in general there may be any
     * number of them */
    while (ret >= 0) {
        ret = avcodec_receive_packet(ctx, pkt);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
            return;
        else if (ret < 0) {
            fprintf(stderr, "Error encoding audio frame\n");
            exit(1);
        }
        fwrite(pkt->data, 1, pkt->size, output);
        av_packet_unref(pkt);
    }
}

static char optstr[] = "?i:o:c:r:f:b:t:";

static void print_usage(char *name)
{
  fprintf(stderr, "Usage: \n");
  fprintf(stderr, "\t%s -i <infile> -o <outfile> -c <channel>"
                  " -r <samplerate> -f <formate> -b <bitrate>"
                  "-t <encodertype>\n", name);
  fprintf(stderr, "Notice: encoder type:\n");
  fprintf(stderr, "\t[alaw  ]:1\n");
  fprintf(stderr, "\t[mulaw ]:2\n");
  fprintf(stderr, "\t[g726  ]:3\n");
  fprintf(stderr, "\t[g726le]:4\n");
}

int main(int argc, char **argv)
{
    const char *outfilename = NULL;
    const char *infilename = NULL;
    const AVCodec *codec;
    AVCodecContext *c= NULL;
    AVFrame *frame;
    AVPacket *pkt;
    int readsize, perframesize;
    int ret, opt;
    FILE *fin, *fout;
    char *inbuff;
    int bitrate = 64000;
    int channel = 2;
    int samplerate = 44100;
    int sampleformat = AV_SAMPLE_FMT_S16; //s16le
    int encodertype = 0;

    opterr = 1;
    while ((opt = getopt(argc, argv, optstr)) != -1) {
      switch (opt) {
      case 'i':
        infilename = optarg;
        fprintf(stderr, "input file path: %s\n", infilename);
        break;
      case 'o':
        outfilename = optarg;
        fprintf(stderr, "output file path: %s\n", outfilename);
        break;
      case 'c':
        channel = atoi(optarg);
        fprintf(stderr, "input channel: %d\n", channel);
        break;
      case 'r':
        samplerate = atoi(optarg);
        fprintf(stderr, "input samplerate: %d\n", samplerate);
        break;
      case 'f':
        sampleformat = atoi(optarg);
        fprintf(stderr, "input sampleformat: %d\n", sampleformat);
        break;
      case 'b':
        bitrate = atoi(optarg);
        fprintf(stderr, "input bitrate: %d\n", bitrate);
        break;
      case 't':
        encodertype = atoi(optarg);
        fprintf(stderr, "input encodertype: %d\n", encodertype);
        break;
      case '?':
      default:
        print_usage(argv[0]);
        exit(0);
      }
    }

    /* Arguments check */
    if (!infilename || !outfilename || !encodertype) {
      print_usage(argv[0]);
      exit(1);
    }

    if ((sampleformat < 1) || (sampleformat > 4)) {
      fprintf(stderr, "ERROR: invalid encoder type:%d\n", encodertype);
      print_usage(argv[0]);
      exit(1);
    }

    if (sampleformat != AV_SAMPLE_FMT_S16) {
      fprintf(stderr, "ERROR: Sample format must be AV_SAMPLE_FMT_S16 for this test!\n");
      exit(1);
    }

    if (((encodertype == 3) || (encodertype == 4)) &&
        ((samplerate > 8000) || (channel != 1) || (bitrate != 16000))) {
      fprintf(stderr, "ERROR: [G726 or G726LE] input file should be:"
                      "channel:1, samplerate:8000, bitrate:16000\n");
      exit(1);
    }

    /* Create encoder */
    if (encodertype == 1) {
      fprintf(stderr, "# creating alaw encoder...\n");
      codec = avcodec_find_encoder(AV_CODEC_ID_PCM_ALAW);
    } else if (encodertype == 2) {
      fprintf(stderr, "# creating mulaw encoder...\n");
      codec = avcodec_find_encoder(AV_CODEC_ID_PCM_MULAW);
    } else if (encodertype == 3) {
      fprintf(stderr, "# creating g726 encoder...\n");
      codec = avcodec_find_encoder(AV_CODEC_ID_ADPCM_G726);
    } else if (encodertype == 4) {
      fprintf(stderr, "# creating g726le encoder...\n");
      codec = avcodec_find_encoder(AV_CODEC_ID_ADPCM_G726LE);
    }
    if (!codec) {
        fprintf(stderr, "ERROR: Codec not found!\n");
        exit(1);
    }

    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "ERROR: Could not allocate audio codec context\n");
        exit(1);
    }

    /* put sample parameters */
    c->bit_rate = bitrate;

    /* check that the encoder supports s16 pcm input */
    c->sample_fmt = AV_SAMPLE_FMT_S16;
    if (!check_sample_fmt(codec, c->sample_fmt)) {
        fprintf(stderr, "ERROR: Encoder does not support sample format %s",
                av_get_sample_fmt_name(c->sample_fmt));
        exit(1);
    }

    c->sample_rate    = samplerate;
    c->channels       = channel;
    if (channel == 1)
      c->channel_layout = AV_CH_LAYOUT_MONO;
    else if (channel == 2)
      c->channel_layout = AV_CH_LAYOUT_STEREO;

#if 0
    if ((encodertype == 3) || (encodertype == 4))
      c->bits_per_coded_sample = 2;
#endif

    fprintf(stderr, "### INFO: sample_rate:%d\n", c->sample_rate);
    fprintf(stderr, "### INFO: channels:%d\n", c->channels);

    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }

    fprintf(stderr, "### INFO: frame_size:%d\n", c->frame_size);
    fprintf(stderr, "### INFO: bits_per_coded_sample:%d\n", c->bits_per_coded_sample);

    fin = fopen(infilename, "rb");
    if (!fin) {
        fprintf(stderr, "Could not open input: %s\n", infilename);
        exit(1);
    }

    fout = fopen(outfilename, "wb");
    if (!fout) {
        fprintf(stderr, "Could not open output: %s\n", outfilename);
        exit(1);
    }

    /* packet for holding encoded output */
    pkt = av_packet_alloc();
    if (!pkt) {
        fprintf(stderr, "could not allocate the packet\n");
        exit(1);
    }

    /* frame containing input raw audio */
    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate audio frame\n");
        exit(1);
    }

    if (c->frame_size == 0) {
      fprintf(stderr, "# Codec has no default frame_size, set 1024 default\n");
      c->frame_size = 1024;
    }

    perframesize = c->channels * av_get_bytes_per_sample(c->sample_fmt);
    fprintf(stderr, "### INFO: perframesize:%d\n", perframesize);
    inbuff = (char *)malloc(c->frame_size * perframesize);
    if (!inbuff) {
      fprintf(stderr, "ERROR: no space left!\n");
      exit(1);
    }

    printf("+++ FLC-DBG: readsize:%d\n", c->frame_size * perframesize);

    while (1) {
        readsize = fread(inbuff, 1, c->frame_size * perframesize, fin);
        if (readsize <= 0) {
          fprintf(stderr, "# Get the end of file! error:%s\n", strerror(errno));
          break;
        }
        fprintf(stderr, "# encoder %d size...\n", readsize);
        frame->nb_samples = readsize / perframesize;
        ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, (unsigned char *)inbuff, readsize, 1);
        if (ret < 0) {
          fprintf(stderr, "ERROR: fill audio frame failed!\n");
          break;
        }

        encode(c, frame, pkt, fout);
    }

    /* flush the encoder */
    encode(c, NULL, pkt, fout);

    free(inbuff);
    fclose(fin);
    fclose(fout);

    av_frame_free(&frame);
    av_packet_free(&pkt);
    avcodec_free_context(&c);

    return 0;
}

编码演示

使用说明：

• g711a

  编码：

  encode_audio -i test8000.pcm -r 8000 -c 1 -o test.alaw 
  -t 1
  

  播放：

  ffplay -ar 8000 -channels 1  -f alaw -i test.alaw -nod
  isp
  

• g711u
  编码：

  encode_audio -i test8000.pcm -r 8000 -c 1 -o test.ulaw 
  -t 2
  

  播放：

  ffplay -ar 8000 -channels 1  -f mulaw -i test.ulaw -nod
  isp
  

• g726

  encode_audio -i test8000.pcm -r 8000 -c 1 -o test.g726 
  	-t 3
  

  播放：
  由于ffplay播放码率为32kbps，编码器要求的编码码率只能为16kbps，因此使用ffplay播放失败。可使用下面章节的解码程序对test.g726进行解码然后再进行pcm的播放。

  注意：
  g726要求必须是采样率：8k，单通道，16kbps的音频数据。本文使用S16LE格式进行测试，其他格式是否支持尚未深究。

解码代码

decode_audio.c: 仅支持G726解码。

/**
 * func: Only decode G726 file.
 * author: Francis Fan
 */

#include <stdio.h>
#include <stdlib.h>
#include <string.h>

#include <libavutil/frame.h>
#include <libavutil/mem.h>

#include <libavcodec/avcodec.h>

#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096

static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
                   FILE *outfile)
{
    int i, ch;
    int ret, data_size;

    /* send the packet with the compressed data to the decoder */
    ret = avcodec_send_packet(dec_ctx, pkt);
    if (ret < 0) {
        fprintf(stderr, "Error submitting the packet to the decoder\n");
        exit(1);
    }

    /* read all the output frames (in general there may be any number of them */
    while (ret >= 0) {
        ret = avcodec_receive_frame(dec_ctx, frame);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
            return;
        else if (ret < 0) {
            fprintf(stderr, "Error during decoding\n");
            exit(1);
        }
        data_size = av_get_bytes_per_sample(dec_ctx->sample_fmt);
        if (data_size < 0) {
            /* This should not occur, checking just for paranoia */
            fprintf(stderr, "Failed to calculate data size\n");
            exit(1);
        }
        for (i = 0; i < frame->nb_samples; i++)
            for (ch = 0; ch < dec_ctx->channels; ch++)
                fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
    }
}

int main(int argc, char **argv)
{
    const char *outfilename, *filename;
    const AVCodec *codec;
    AVCodecContext *c= NULL;
    AVCodecParserContext *parser = NULL;
    int len, ret;
    FILE *f, *outfile;
    uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
    uint8_t *data;
    size_t   data_size;
    AVPacket *pkt;
    AVFrame *decoded_frame = NULL;

    if (argc <= 2) {
        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
        exit(0);
    }
    filename    = argv[1];
    outfilename = argv[2];

    pkt = av_packet_alloc();

    /* find the MPEG audio decoder */
    codec = avcodec_find_decoder(AV_CODEC_ID_ADPCM_G726);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }

    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate audio codec context\n");
        exit(1);
    }

    c->bits_per_coded_sample = 2;   //g726压缩比为8:1 编码前采样用bit数为那么编码后应该占/8 = 2
    c->channels = 1;
    c->sample_fmt = AV_SAMPLE_FMT_S16;
    c->sample_rate = 8000;
    c->codec_type = AVMEDIA_TYPE_AUDIO;
    c->bit_rate = 16000;

    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }

    f = fopen(filename, "rb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }
    outfile = fopen(outfilename, "wb");
    if (!outfile) {
        av_free(c);
        exit(1);
    }

    while (1) {
        if (!decoded_frame) {
            if (!(decoded_frame = av_frame_alloc())) {
                fprintf(stderr, "Could not allocate audio frame\n");
                exit(1);
            }
        }

        data      = inbuf;
        data_size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
        if (data_size <= 0) {
          printf("INFO: Get end of the file\n");
          break;
        }

        pkt->data = (uint8_t *)inbuf;
        pkt->size = data_size;

        if (pkt->size)
            decode(c, pkt, decoded_frame, outfile);
    }

    /* flush the decoder */
    pkt->data = NULL;
    pkt->size = 0;
    decode(c, pkt, decoded_frame, outfile);

    fclose(outfile);
    fclose(f);

    avcodec_free_context(&c);
    av_frame_free(&decoded_frame);
    av_packet_free(&pkt);

    return 0;
}

解码演示

decode_audio test.g726 g726.pcm

注：有些设备不支持mono pcm，可使用ffplay播放或者pull出源文件，使用Audacity播放。