聊天平臺原始碼，聊天平臺如何獲取到音訊流的相關程式碼

我們重取樣的引數要和SDL的引數是一致的，還要考慮到雖然沒有AVPacket了，但是解碼器中還有未解碼的資料，我們要用avcodec_send_packet(aCodecCtx, NULL)告訴解碼器沒有資料了，重新整理解碼器。

下面這段程式碼的邏輯是如果沒有資料要解碼了，就跳轉到獲取解碼資料的邏輯。

int audio_decode_frame(AVCodecContext *aCodecCtx, uint8_t *audio_buf, int buf_size)
{

 static AVPacket pkt;
 static uint8_t *audio_pkt_data = NULL;
 static int audio_pkt_size = 0;
 static AVFrame frame;
 av_frame_get_buffer(&frame, 0);
 int len1, data_size = 0;
 int ret = 0;
 int is_no_pkt = 0;

 for (;;)
 {

  if (pkt.data)
   av_packet_unref(&pkt);

  if (quit)
  {
   return -1;
  }

  if (packet_queue_get(&audioq, &pkt, 1) < 0)
  {
   ret = avcodec_send_packet(aCodecCtx, NULL);
   if (ret < 0)
   {
    return -1;
   }
   is_no_pkt = 1;
   goto __RECEIVE;
  }

  ret = avcodec_send_packet(aCodecCtx, &pkt);
  if (ret < 0)
  {
   ret = -1;
   printf("decode error");
   av_packet_unref(&pkt);
   return -1;
  }
 __RECEIVE:
  if (is_no_pkt)
  {
   return -1;
  }
  else if (pkt.data)
  {
   av_packet_unref(&pkt);
  }
  ret = avcodec_receive_frame(aCodecCtx, &frame);
  if (ret < 0)
  {
   continue;
  }
 
  data_size = av_get_bytes_per_sample(out_format) * out_channel * SDL_AUDIO_BUFFER_SIZE;

  swr_convert(audio_convert_ctx,
        &audio_buf,
        SDL_AUDIO_BUFFER_SIZE,
        (const uint8_t **)frame.data,
        frame.nb_samples);
  return data_size;
 }
}

​

轉碼後的資料大小=取樣大小 * 單通道取樣個數 * 通道數

data_size = av_get_bytes_per_sample(out_format) * out_channel * SDL_AUDIO_BUFFER_SIZE;

下面是完整的程式碼

#include <stdio.h>
#include <assert.h>

#include <SDL2/SDL.h>

#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.h>
#include <libavutil/samplefmt.h>

#define SDL_AUDIO_BUFFER_SIZE 1024
#define MAX_AUDIO_FRAME_SIZE 192000
typedef struct PacketQueue
{
 AVPacketList *first_pkt, *last_pkt;
 int nb_packets;
 int size;
 SDL_mutex *mutex;
 SDL_cond *cond;
} PacketQueue;

PacketQueue audioq;
int quit = 0;
struct SwrContext *audio_convert_ctx = NULL;
static Uint8 out_channel = 2;
static SDL_AudioFormat out_format = AV_SAMPLE_FMT_S16;
static int out_nb_samples = 0;
static int sample_rate = 48000;


void audio_callback(void *userdata, Uint8 *stream, int len);
void packet_queue_init(PacketQueue *q);
int packet_queue_put(PacketQueue *q, AVPacket *pkt);
int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block);
int audio_decode_frame(AVCodecContext *aCodecCtx, uint8_t *audio_buf, int buf_size);

int main(int argc, char *argv[])
{
 int ret = 0;
 AVFormatContext *fmt_ctx = NULL;
 char *in_file;
 int video_stream_index = -1;
 int audio_stream_index = -1;
 AVCodec *video_codec = NULL, *audio_codec = NULL;
 AVCodecParameters *audio_codecpar = NULL, *video_codecpar = NULL;
 AVCodecContext *audio_codec_ctx = NULL, *video_codec_ctx = NULL;
 SDL_AudioSpec wanted_spec, src_spec;
 int64_t in_channel_layout;
 int64_t out_channel_layout;
 AVPacket packet;
 SDL_Event event;
 av_log_set_level(AV_LOG_INFO);

 if (argc < 2)
 {
  av_log(NULL, AV_LOG_ERROR, "need infile\n");
  return -1;
 }
 in_file = argv[1];
 //初始化SDL
 ret = SDL_Init(SDL_INIT_AUDIO | SDL_INIT_VIDEO | SDL_INIT_TIMER);
 if (ret != 0)
 {
  av_log(NULL, AV_LOG_ERROR, "sdl init error\n");
  return ret;
 }

 //獲取輸入檔案上下文
 ret = avformat_open_input(&fmt_ctx,
              in_file,
              NULL,
              NULL);
 if (!fmt_ctx)
 {
  av_log(NULL, AV_LOG_ERROR, "open input file fail\n");
  goto __FAIL;
 }

 //獲取視訊流索引和音訊流索引
 ret = avformat_find_stream_info(fmt_ctx, NULL);
 if (ret < 0)
 {
  av_log(NULL, AV_LOG_ERROR, "find stream fail\n");
  goto __FAIL;
 }
 av_dump_format(fmt_ctx, 0, in_file, 0);
 video_stream_index = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &video_codec, -1);
 audio_stream_index = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &audio_codec, -1);
 if (video_stream_index < 0 || audio_stream_index < 0)
 {
  av_log(NULL, AV_LOG_ERROR, "find audio or video fail\n");
  goto __FAIL;
 }
 if (!audio_codec || !video_codec)
 {
  ret = -1;
  av_log(NULL, AV_LOG_ERROR, "find audio codec or video codec fail");
  goto __FAIL;
 }
 //獲取解碼器上下文並開啟解碼器
 audio_codecpar = fmt_ctx->streams[audio_stream_index]->codecpar;
 out_nb_samples = audio_codecpar->frame_size;

 video_codecpar = fmt_ctx->streams[video_stream_index]->codecpar;

 audio_codec_ctx = avcodec_alloc_context3(audio_codec);
 video_codec_ctx = avcodec_alloc_context3(video_codec);

 avcodec_parameters_to_context(audio_codec_ctx, audio_codecpar);
 avcodec_parameters_to_context(video_codec_ctx, video_codecpar);

 ret = avcodec_open2(audio_codec_ctx, audio_codec, NULL);
 if (ret < 0)
 {
  av_log(NULL, AV_LOG_ERROR, "open audio codec fail\n");
  goto __FAIL;
 }

 ret = avcodec_open2(video_codec_ctx, video_codec, NULL);
 if (ret < 0)
 {
  av_log(NULL, AV_LOG_ERROR, "open video codec fail\n");
  goto __FAIL;
 }

 //設定SDL_音訊引數
 wanted_spec.freq = sample_rate; //取樣率
 wanted_spec.format = AUDIO_S16LSB; //取樣大小
 wanted_spec.channels = out_channel; //聲道數
 wanted_spec.silence = 0;
 wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE; //取樣個數
 wanted_spec.callback = audio_callback;
 wanted_spec.userdata = audio_codec_ctx;

 ret = SDL_OpenAudio(&wanted_spec, &src_spec);
 if (ret < 0)
 {
  av_log(NULL, AV_LOG_ERROR, "open audio device fail\n");
  goto __FAIL;
 }

 //設定重取樣
 in_channel_layout = audio_codecpar->channel_layout;
 out_channel_layout = av_get_default_channel_layout(out_channel);
 audio_convert_ctx = swr_alloc();
 if (!audio_convert_ctx)
 {
  ret = -1;
  av_log(NULL, AV_LOG_ERROR, "alloc swr fail\n");
  goto __FAIL;
 }

 swr_alloc_set_opts(audio_convert_ctx,
          out_channel_layout,
          out_format,
          sample_rate,
          in_channel_layout,
          audio_codecpar->format,
          audio_codecpar->sample_rate,
          0,
          NULL);

 swr_init(audio_convert_ctx);
 SDL_PauseAudio(0);

 packet_queue_init(&audioq);
 while (av_read_frame(fmt_ctx, &packet) >= 0)
 {
  if (packet.stream_index == audio_stream_index)
  {
   packet_queue_put(&audioq, &packet);
  }
  else
  {
   av_packet_unref(&packet);
  }
  SDL_PollEvent(&event);
  switch (event.type)
  {
  case SDL_QUIT:
   quit = 1;
   goto __FAIL;
   break;
  default:
   break;
  }
 }
 while (1)
 {
  SDL_WaitEvent(&event);
  switch (event.type)
  {
  case SDL_QUIT:
   quit = 1;
   goto __FAIL;
   break;
  default:
   break;
  }
 }

__FAIL:
 SDL_CloseAudio();
 SDL_Quit();
 if (audio_codecpar)
 {
  avcodec_parameters_free(&audio_codecpar);
 }

 if (video_codecpar)
 {
  avcodec_parameters_free(&video_codecpar);
 }

 if (video_codec_ctx)
 {
  avcodec_close(video_codec_ctx);
  avcodec_free_context(&video_codec_ctx);
 }

 if (audio_codec_ctx)
 {
  avcodec_close(audio_codec_ctx);
  avcodec_free_context(&audio_codec_ctx);
 }
 if (fmt_ctx)
 {
  avformat_close_input(&fmt_ctx);
  avformat_free_context(fmt_ctx);
 }

 return ret;
}

void audio_callback(void *userdata, Uint8 *stream, int len)
{

 AVCodecContext *aCodecCtx = (AVCodecContext *)userdata;
 int len1, audio_size;

 static uint8_t audio_buf[(MAX_AUDIO_FRAME_SIZE * 3) / 2];
 static unsigned int audio_buf_size = 0;
 static unsigned int audio_buf_index = 0;

 while (len > 0)
 {
  if (audio_buf_index >= audio_buf_size)
  {
   /* We have already sent all our data; get more */
   audio_size = audio_decode_frame(aCodecCtx, audio_buf, sizeof(audio_buf));
   if (audio_size < 0)
   {
    /* If error, output silence */
    audio_buf_size = 1024; // arbitrary?
    memset(audio_buf, 0, audio_buf_size);
   }
   else
   {
    audio_buf_size = audio_size;
   }
   audio_buf_index = 0;
  }
  len1 = audio_buf_size - audio_buf_index;
  if (len1 > len)
   len1 = len;
  fprintf(stderr, "index=%d, len1=%d, len=%d\n",
      audio_buf_index,
      len,
      len1);
  memcpy(stream, (uint8_t *)audio_buf + audio_buf_index, len1);
  len -= len1;
  stream += len1;
  audio_buf_index += len1;
 }
}

void packet_queue_init(PacketQueue *q)
{
 memset(q, 0, sizeof(PacketQueue));
 q->mutex = SDL_CreateMutex();
 q->cond = SDL_CreateCond();
}

int packet_queue_put(PacketQueue *q, AVPacket *pkt)
{

 AVPacketList *pkt1;
 AVPacket dst_pkt;
 av_init_packet(&dst_pkt);
 if (av_packet_ref(&dst_pkt, pkt) < 0)
 {
  return -1;
 }
 pkt1 = av_malloc(sizeof(AVPacketList));
 if (!pkt1)
  return -1;
 pkt1->pkt = dst_pkt;
 pkt1->next = NULL;

 SDL_LockMutex(q->mutex);

 if (!q->last_pkt)
 {
  q->first_pkt = pkt1;
 }
 else
 {
  q->last_pkt->next = pkt1;
 }

 q->last_pkt = pkt1;
 q->nb_packets++;
 q->size += pkt1->pkt.size;
 SDL_CondSignal(q->cond);

 SDL_UnlockMutex(q->mutex);
 return 0;
}

int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block)
{
 AVPacketList *pkt1;
 int ret;

 SDL_LockMutex(q->mutex);

 for (;;)
 {

  if (quit)
  {
   ret = -1;
   break;
  }

  pkt1 = q->first_pkt;
  if (pkt1)
  {
   q->first_pkt = pkt1->next;
   if (!q->first_pkt)
    q->last_pkt = NULL;
   q->nb_packets--;
   q->size -= pkt1->pkt.size;
   *pkt = pkt1->pkt;
   av_free(pkt1);
   ret = 1;
   break;
  }
  else if (!block)
  {
   ret = 0;
   break;
  }
  else
  {
   SDL_CondWait(q->cond, q->mutex);
  }
 }
 SDL_UnlockMutex(q->mutex);
 return ret;
}

int audio_decode_frame(AVCodecContext *aCodecCtx, uint8_t *audio_buf, int buf_size)
{

 static AVPacket pkt;
 static uint8_t *audio_pkt_data = NULL;
 static int audio_pkt_size = 0;
 static AVFrame frame;
 av_frame_get_buffer(&frame, 0);
 int len1, data_size = 0;
 int ret = 0;
 int is_no_pkt = 0;

 for (;;)
 {

  if (pkt.data)
   av_packet_unref(&pkt);

  if (quit)
  {
   return -1;
  }

  if (packet_queue_get(&audioq, &pkt, 1) < 0)
  {
   ret = avcodec_send_packet(aCodecCtx, NULL);
   if (ret < 0)
   {
    return -1;
   }
   is_no_pkt = 1;
   goto __RECEIVE;
  }

  ret = avcodec_send_packet(aCodecCtx, &pkt);
  if (ret < 0)
  {
   ret = -1;
   printf("decode error");
   av_packet_unref(&pkt);
   return -1;
  }
 __RECEIVE:
  if (is_no_pkt)
  {
   return -1;
  }
  else if (pkt.data)
  {
   av_packet_unref(&pkt);
  }
  ret = avcodec_receive_frame(aCodecCtx, &frame);
  if (ret < 0)
  {
   continue;
  }
 
  data_size = av_get_bytes_per_sample(out_format) * out_channel * SDL_AUDIO_BUFFER_SIZE;

  swr_convert(audio_convert_ctx,
        &audio_buf,
        SDL_AUDIO_BUFFER_SIZE,
        (const uint8_t **)frame.data,
        frame.nb_samples);
  return data_size;
 }
}

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