Decoding and resampling audio with FFmpeg for output with libao - c

Question

I'm trying to write a program to read and play an audio file using FFmpeg and libao. I've been following the procedure outlined in the FFmpeg documentation for decoding audio using the new avcodec_send_packet and avcodec_receive_frame functions, but the examples I've been able to find are few and far between (the ones in the FFmpeg documentation either don't use libavformat or use the deprecated avcodec_decode_audio4). I've based a lot of my program off of the transcode_aac.c example (up to init_resampler) in the FFmpeg documentation, but that also uses the deprecated decoding function.
I believe I have the decoding part of the program working, but I need to resample the audio in order to convert it into an interleaved format to send to libao, for which I'm attempting to use libswresample. Whenever the program is run in its current state, it outputs (many times) "Error resampling: Output changed". The test file I've been using is just a YouTube rip that I had on hand. ffprobe reports the only stream as:
Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 125 kb/s (default)
This is my first program with FFmpeg (and I'm still relatively new to C), so any advice on how to improve/fix other parts of the program would be welcome.
#include<stdio.h>
#include<libavcodec/avcodec.h>
#include<libavformat/avformat.h>
#include<libavutil/avutil.h>
#include<libswresample/swresample.h>
#include<ao/ao.h>
#define OUTPUT_CHANNELS 2
#define OUTPUT_RATE 44100
#define BUFFER_SIZE 192000
#define OUTPUT_BITS 16
#define OUTPUT_FMT AV_SAMPLE_FMT_S16
static char *errtext (int err) {
static char errbuff[256];
av_strerror(err,errbuff,sizeof(errbuff));
return errbuff;
}
static int open_audio_file (const char *filename, AVFormatContext **context, AVCodecContext **codec_context) {
AVCodecContext *avctx;
AVCodec *codec;
int ret;
int stream_id;
int i;
// Open input file
if ((ret = avformat_open_input(context,filename,NULL,NULL)) < 0) {
fprintf(stderr,"Error opening input file '%s': %s\n",filename,errtext(ret));
*context = NULL;
return ret;
}
// Get stream info
if ((ret = avformat_find_stream_info(*context,NULL)) < 0) {
fprintf(stderr,"Unable to find stream info: %s\n",errtext(ret));
avformat_close_input(context);
return ret;
}
// Find the best stream
if ((stream_id = av_find_best_stream(*context,AVMEDIA_TYPE_AUDIO,-1,-1,&codec,0)) < 0) {
fprintf(stderr,"Unable to find valid audio stream: %s\n",errtext(stream_id));
avformat_close_input(context);
return stream_id;
}
// Allocate a decoding context
if (!(avctx = avcodec_alloc_context3(codec))) {
fprintf(stderr,"Unable to allocate decoder context\n");
avformat_close_input(context);
return AVERROR(ENOMEM);
}
// Initialize stream parameters
if ((ret = avcodec_parameters_to_context(avctx,(*context)->streams[stream_id]->codecpar)) < 0) {
fprintf(stderr,"Unable to get stream parameters: %s\n",errtext(ret));
avformat_close_input(context);
avcodec_free_context(&avctx);
return ret;
}
// Open the decoder
if ((ret = avcodec_open2(avctx,codec,NULL)) < 0) {
fprintf(stderr,"Could not open codec: %s\n",errtext(ret));
avformat_close_input(context);
avcodec_free_context(&avctx);
return ret;
}
*codec_context = avctx;
return 0;
}
static void init_packet (AVPacket *packet) {
av_init_packet(packet);
packet->data = NULL;
packet->size = 0;
}
static int init_resampler (AVCodecContext *codec_context, SwrContext **resample_context) {
int ret;
// Set resampler options
*resample_context = swr_alloc_set_opts(NULL,
av_get_default_channel_layout(OUTPUT_CHANNELS),
OUTPUT_FMT,
codec_context->sample_rate,
av_get_default_channel_layout(codec_context->channels),
codec_context->sample_fmt,
codec_context->sample_rate,
0,NULL);
if (!(*resample_context)) {
fprintf(stderr,"Unable to allocate resampler context\n");
return AVERROR(ENOMEM);
}
// Open the resampler
if ((ret = swr_init(*resample_context)) < 0) {
fprintf(stderr,"Unable to open resampler context: %s\n",errtext(ret));
swr_free(resample_context);
return ret;
}
return 0;
}
static int init_frame (AVFrame **frame) {
if (!(*frame = av_frame_alloc())) {
fprintf(stderr,"Could not allocate frame\n");
return AVERROR(ENOMEM);
}
return 0;
}
int main (int argc, char *argv[]) {
AVFormatContext *context = 0;
AVCodecContext *codec_context;
SwrContext *resample_context = NULL;
AVPacket packet;
AVFrame *frame = 0;
AVFrame *resampled = 0;
int16_t *buffer;
int ret, packet_ret, finished;
ao_device *device;
ao_sample_format format;
int default_driver;
if (argc != 2) {
fprintf(stderr,"Usage: %s <filename>\n",argv[0]);
return 1;
}
av_register_all();
printf("Opening file...\n");
if (open_audio_file(argv[1],&context,&codec_context) < 0)
return 1;
printf("Initializing resampler...\n");
if (init_resampler(codec_context,&resample_context) < 0) {
avformat_close_input(&context);
avcodec_free_context(&codec_context);
return 1;
}
// Setup libao
printf("Starting audio device...\n");
ao_initialize();
default_driver = ao_default_driver_id();
format.bits = OUTPUT_BITS;
format.channels = OUTPUT_CHANNELS;
format.rate = codec_context->sample_rate;
format.byte_format = AO_FMT_NATIVE;
format.matrix = 0;
if ((device = ao_open_live(default_driver,&format,NULL)) == NULL) {
fprintf(stderr,"Error opening audio device\n");
avformat_close_input(&context);
avcodec_free_context(&codec_context);
swr_free(&resample_context);
return 1;
}
// Mainloop
printf("Beginning mainloop...\n");
init_packet(&packet);
// Read packets until done
while (1) {
packet_ret = av_read_frame(context,&packet);
// Send a packet
if ((ret = avcodec_send_packet(codec_context,&packet)) < 0)
fprintf(stderr,"Error sending packet to decoder: %s\n",errtext(ret));
av_packet_unref(&packet);
while (1) {
if (!frame)
frame = av_frame_alloc();
ret = avcodec_receive_frame(codec_context,frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) // Need more input
break;
else if (ret < 0) {
fprintf(stderr,"Error receiving frame: %s\n",errtext(ret));
break;
}
// We have a valid frame, need to resample it
if (!resampled)
resampled = av_frame_alloc();
resampled->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
resampled->sample_rate = codec_context->sample_rate;
resampled->format = OUTPUT_FMT;
if ((ret = swr_convert_frame(resample_context,resampled,frame)) < 0) {
fprintf(stderr,"Error resampling: %s\n",errtext(ret));
} else {
ao_play(device,(char*)resampled->extended_data[0],resampled->linesize[0]);
}
av_frame_unref(resampled);
av_frame_unref(frame);
}
if (packet_ret == AVERROR_EOF)
break;
}
printf("Closing file and freeing contexts...\n");
avformat_close_input(&context);
avcodec_free_context(&codec_context);
swr_free(&resample_context);
printf("Closing audio device...\n");
ao_close(device);
ao_shutdown();
return 0;
}
UPDATE: I've got it playing sound now, but it sounds like samples are missing (and MP3 files warn that "Could not update timestamps for skipped samples"). The issue was that the resampled frame needed to have certain attributes set before being passed to swr_convert_frame. I've also added av_packet_unref and av_frame_unref, but I'm still unsure as to where to best locate them.

Answer 1

ao_play(device,(char*)resampled->extended_data[0],resampled->linesize[0]);
You have problem in this line. Resampled audio frame has incorrect linesize parameters. swr_convert_frame aligns data and extended_data fields with silence. This silence is included into linesize parameter so you pass incorrect frame size into ao_play function.
ao_play(device, (char*)resampled->extended_data[0], av_sample_get_buffer_size(resampled->linesize, resampled->channels, resampled->nb_samples, resampled->format, 0));
Function av_sample_get_buffer_size() returns true sample size, without align. When I faced similar problem, this was the solution.