我正在尝试使用ffmpeg解码音频文件，而我正在这样做，我得到许多未解决的外部错误。我是ffmpeg图书馆的新手，对这个问题的任何建议都会有很大的帮助。
谢谢。
void audioDecode(char* filename)
{
FILE *file;
AVFormatContext *audioInputFormatContext;
AVInputFormat *audioInputFormat = NULL;
AVCodec *audioCodec;
AVCodecContext *audioCodecContext;
av_register_all();
char *audioInputDeviceName = filename;
int ret;
int audioIndex = 0;
AVPacket pkt;
av_init_packet(&pkt);
avformat_network_init();
audioInputFormatContext = avformat_alloc_context();
ret = avformat_open_input(&audioInputFormatContext, audioInputDeviceName, audioInputFormat, NULL);
if (ret == 0)
{
ret = avformat_find_stream_info(audioInputFormatContext, 0);
if (ret >= 0)
{
for (int i = 0; i < audioInputFormatContext->nb_streams; i++) {
if (audioInputFormatContext->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
audioIndex = i;
break;
}
}
audioCodec = avcodec_find_decoder(audioInputFormatContext->streams[audioIndex]->codecpar->codec_id);
audioCodecContext = avcodec_alloc_context3(audioCodec);
avcodec_parameters_to_context(audioCodecContext, audioInputFormatContext->streams[audioIndex]->codecpar);
if (avcodec_open2(audioCodecContext, audioCodec, NULL) >= 0)
{
ret = av_read_frame(audioInputFormatContext, &pkt);
AVPacket encodePacket;
AVFrame* decodeFrame = av_frame_alloc();
int dec_got_frame = 0;
if (ret == 0)
{
ret = avcodec_send_packet(audioCodecContext, &pkt);
if (ret < 0)
printf("Error");
}
ret = avcodec_receive_frame(audioCodecContext, decodeFrame);
if (ret >= 0)
dec_got_frame = 1;
if (dec_got_frame)
{
fopen_s(&file, filename, "wb");
fwrite(pkt.data, 1, pkt.size, file);
fclose(file);
}
av_frame_free(&decodeFrame);
}
}
}
avformat_close_input(&audioInputFormatContext);
avcodec_free_context(&audioCodecContext);
av_packet_unref(&pkt);
}
答案 0 :(得分：0)
我只是假设您已经连接到流源，并通过您在评论中提到的内容来获得编解码器上下文。
这些是我自己的项目解码音频帧的片段。
解码音频数据包：
void FFMPEG::process_audio_packet(AVPacket *pkt) {
int got;
avcodec_decode_audio4(aud_stream.context, aud_stream.frame, &got, pkt);
if (got) Audio.add_av_frame(aud_stream.frame);
}
处理完成的帧并提取立体声16位带符号缓冲区：
void AudioManager::add_av_frame(AVFrame *frame) {
int nsamples = frame->nb_samples;
int sample_rate = frame->sample_rate;
int channels = frame->channels;
AVSampleFormat format = (AVSampleFormat) frame->format;
bool planar = av_sample_fmt_is_planar(format) == 1;
int64_t pts = av_frame_get_best_effort_timestamp(frame);
//double ftime;
/*if (ffmpeg.vid_stream.stream_id != -1)
ftime = av_q2d(ffmpeg.aud_stream.context->time_base) * pts;
else
ftime = av_q2d(ffmpeg.vid_stream.context->time_base) * pts;*/
AudioBuffer *buffer = NULL;
if (planar) { // handle planar audio frames
/*
* PLANAR frames conversion
* ------------------------
*/
if (channels == 1) { // MONO
//LOGD("Processing PLANAR MONO");
/*
* MONO
*/
if (format == AV_SAMPLE_FMT_S16P) { // 16 bit signed
if ((buffer = alloc_buffer(frame))) { // allocated okay?
short *channel = (short*)frame->data[0];
short *buff = buffer->data;
for (int c = 0; c < nsamples; c++) {
*buff++ = *channel;
*buff++ = *channel++;
}
queue_new_buffer(buffer);
}
return;
}
if (format == AV_SAMPLE_FMT_S32P) { // 32 bit signed
if ((buffer = alloc_buffer(frame))) { // allocated okay?
int32_t *channel = (int32_t*)frame->data[0];
short *buff = buffer->data;
for (int c = 0; c < nsamples; c++) {
int16_t s = (int16_t) (*channel++ >> 16);
*buff++ = s;
*buff++ = s;
}
queue_new_buffer(buffer);
}
return;
}
if (format == AV_SAMPLE_FMT_U8P) { // 8 bit unsigned
if ((buffer = alloc_buffer(frame))) { // allocated okay?
uint8_t *channel = (uint8_t*)frame->data[0];
short *buff = buffer->data;
for (int c = 0; c < nsamples; c++) {
int16_t s = ((int16_t)(*channel++ - 128) << 8);
*buff++ = s;
*buff++ = s;
}
queue_new_buffer(buffer);
}
}
return; // scrap if no audio buffer (highly unlikely)
} else if (channels == 2) { // STEREO
//LOGD("Processing PLANAR STEREO");
/*
* STEREO
*/
if (format == AV_SAMPLE_FMT_S16P) { // 16 bit signed
if ((buffer = alloc_buffer(frame))) { // allocated okay
short *channel1 = (short*)frame->data[0];
short *channel2 = (short*)frame->data[1];
short *buff = buffer->data;
for (int c = 0; c < nsamples; c++) {
*buff++ = *channel1++;
*buff++ = *channel2++;
}
queue_new_buffer(buffer);
}
return;
}
if (format == AV_SAMPLE_FMT_S32P) { // 32 bit signed
if ((buffer = alloc_buffer(frame))) { // allocated okay?
int32_t *channel1 = (int32_t*)frame->data[0];
int32_t *channel2 = (int32_t*)frame->data[1];
short *buff = buffer->data;
for (int c = 0; c < nsamples; c++) {
int16_t s1 = (int16_t) (*channel1++ >> 16);
int16_t s2 = (int16_t) (*channel2++ >> 16);
*buff++ = s1;
*buff++ = s2;
}
queue_new_buffer(buffer);
}
return;
}
if (format == AV_SAMPLE_FMT_U8P) { // 8 bit unsigned
if ((buffer = alloc_buffer(frame))) { // allocated okay?
uint8_t *channel1 = (uint8_t*)frame->data[0];
uint8_t *channel2 = (uint8_t*)frame->data[1];
short *buff = buffer->data;
for (int c = 0; c < nsamples; c++) {
int16_t s1 = ((int16_t)(*channel1++ - 128) << 8);
int16_t s2 = ((int16_t)(*channel2++ - 128) << 8);
*buff++ = s1;
*buff++ = s2;
}
queue_new_buffer(buffer);
}
}
return;
} // TODO: Handle more channels at a later date
} else { // handle non-planar audio frames
/*
* INTERLEAVED conversion
* ----------------------
*/
}
}
处理音频缓冲区：
void AudioManager::queue_new_buffer(AudioBuffer *buffer) {
if (buffer) { // valid buffer
// apply volume gain (only working with stereo)
if (volume != 100) {
short *data = buffer->data;
int num_samples = buffer->nsamples << 1;
while (num_samples--) {
long sample = ((long)*data * volume) / 100;
if (sample < -32768) sample = -32768;
if (sample > 32767) sample = 32767;
*data++ = (short)sample;
}
}
// add buffer to queue
buffer->used = true;
double pts_start = get_pts_start_time();
decode_pos = (++decode_pos) % MAX_AUD_BUFFERS;
if (decode_pos == playback_pos)
playback_pos = (++playback_pos) % MAX_AUD_BUFFERS;
if (ffmpeg.vid_stream.stream_id == -1 && pts_start < 0.0) {
set_pts_start_time(buffer->frame_time);
set_sys_start_time(Display.get_current_render_time());
LOGD("START TIME FROM AUDIO STREAM...");
}
//LOGI("Audio buffer queued %d (%d)", decode_pos, playback_pos);
}
}