Hi All, I want to convert any audio [libav recognised] format to raw
wave content of 1 channel and 8000 bitrate. Following is the source
code, however the read samples are always null. Could you please point
out the obvious msitake that I have overlooked. Any help is
appriciated.
TIA.
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/avutil.h>
#define RATE 8000
#define CHANNELS 1
#define SAMPLES_SIZE ((AVCODEC_MAX_AUDIO_FRAME_SIZE *
2)+FF_INPUT_BUFFER_PADDING_SIZE)
AVStream* get_stream(struct AVFormatContext *ic, enum CodecType type) {
unsigned int i;
for(i = 0; i < ic->nb_streams; i++) {
if(ic->streams[i]->codec->codec_type == type) {
return ic->streams[i];
}
}
return NULL;
}
static int init_audio_pcm(AVCodecContext * codec_context){
AVCodec * codec = avcodec_find_encoder(CODEC_ID_PCM_S16LE);
if (!codec){
return -1;
}
if (avcodec_open(codec_context, codec) < 0) {
return -1;
}
codec_context->sample_rate = RATE;
codec_context->channels = CHANNELS;
codec_context->bit_rate = 8000;
return 0;
}
int main(int argc, char* argv[])
{
AVFormatContext *in_format_ctx;
AVStream* in_audio_st ;
AVCodec *in_audio_codec;
AVPacket in_packet;
int ret_val;
int16_t *samples, *resamples, *audio_out;
int samples_size, new_sample_size , audio_out_size, frame_bytes;
/* output */
AVCodecContext *out_codec_ctx;
ReSampleContext *resample_ctx;
avcodec_init();
avcodec_register_all();
av_register_all();
av_log_set_level(AV_LOG_DEBUG|AV_LOG_VERBOSE|AV_LOG_INFO|AV_LOG_WARNING|AV_LOG_ERROR|AV_LOG_FATAL|AV_LOG_PANIC);
if((ret_val=av_open_input_file(&in_format_ctx, argv[1], NULL, 0, NULL))!=0){
return -1;
}
if(ret_val=av_find_stream_info(in_format_ctx)<0){
return -1;
}
dump_format(in_format_ctx, 0, argv[1], 0);
if(!((get_stream(in_format_ctx,CODEC_TYPE_VIDEO)==NULL) &&
((in_audio_st=get_stream(in_format_ctx,CODEC_TYPE_AUDIO))!=NULL) )) {
return -1;
}
in_audio_codec = avcodec_find_decoder(in_audio_st->codec->codec_id);
if(in_audio_codec == NULL) {
return -1;
}
if(avcodec_open(in_audio_st->codec, in_audio_codec) < 0) {
return -1;
}
av_init_packet(&in_packet);
samples=(int16_t *)av_malloc(SAMPLES_SIZE);
resamples=(int16_t *)av_malloc(SAMPLES_SIZE);
audio_out=(int16_t *)av_malloc(SAMPLES_SIZE);
if(samples == NULL || resamples == NULL||audio_out == NULL) {
return -1;
}
out_codec_ctx=avcodec_alloc_context();
if(out_codec_ctx == NULL) {
return -1;
}
ret_val=init_audio_pcm(out_codec_ctx);
if(ret_val < 0){
return -1;
}
frame_bytes = out_codec_ctx->frame_size * 2 * out_codec_ctx->channels;
resample_ctx=av_audio_resample_init(CHANNELS,in_audio_st->codec->channels,RATE,in_audio_st->codec->sample_rate,
SAMPLE_FMT_S16,in_audio_st->codec->sample_fmt,16, 10, 0, 0.8);
if(resample_ctx == NULL) {
return -1;
}
while ((ret_val = av_read_frame(in_format_ctx, &in_packet)) >= 0) {
if ( in_packet.size <1 ) {
continue;
}
samples_size = SAMPLES_SIZE;
ret_val = avcodec_decode_audio3(in_audio_st->codec,(int16_t*)
samples, &samples_size, &in_packet);
if(in_packet.size > ret_val) {
continue;
}
if(samples_size < 0 ) {
continue;
}
if(samples_size > (AVCODEC_MAX_AUDIO_FRAME_SIZE * 2)){
continue;
}
if(samples_size == 0) {
continue;
}
new_sample_size = audio_resample(resample_ctx, resamples,
samples,samples_size/( sizeof( short )*CHANNELS ) );
audio_out_size = avcodec_encode_audio(out_codec_ctx,
audio_out, new_sample_size, (short*)resamples);
av_free_packet(&in_packet);
}
av_close_input_file(in_format_ctx);
audio_resample_close(resample_ctx);
return 0;
}
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