/**
* encodes raw audio to various encodings like mp3
* @param args it holds various parameters like desired output encoding
* @see libav_encode_args
*/
void audio_encode(libav_encode_args* a) {
libav_encode_args& args = *a;
avcodec_register_all();
AVCodec *codec;
AVCodecContext *c = NULL;
AVFrame *frame;
AVPacket pkt;
int i, j, k, ret, got_output;
int max_input_buffer_size = args.input_buffer.periodbuffer->size();
int input_buffer_size = args.initptr < args.termptr ? (args.termptr - args.
initptr + 1) : (args.termptr + args.input_buffer.periodbuffer->
size() - args.initptr + 1);
FILE *f;
uint16_t *samples;
float t, tincr;
// printf("Encode audio file %s\n", filename);
/* find the MP2 encoder */
codec = avcodec_find_encoder(args.codecID);
if (!codec) {
// fprintf(stderr, "Codec not found\n");
// exit(1);
ErrnoExit("Codec not found");
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* put sample parameters */
c->bit_rate = args.bitrate;
/* check that the encoder supports s16 pcm input */
c->sample_fmt = args.av_smpl_fmt; //AV_SAMPLE_FMT_S16;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "Encoder does not support sample format %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
// f = fopen(filename, "wb");
// if (!f) {
// fprintf(stderr, "Could not open %s\n", filename);
// exit(1);
// }
/* frame containing input raw audio */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* the codec gives us the frame size, in samples,
* we calculate the size of the samples buffer in bytes */
int computed_buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size, c->sample_fmt, 0);
samples = (uint16_t*) av_malloc(computed_buffer_size);
if (!samples) {
fprintf(stderr, "Could not allocate %d bytes for samples buffer\n", computed_buffer_size);
exit(1);
}
/* setup the data pointers in the AVFrame */
if (computed_buffer_size != (*args.input_buffer.periodbuffer)[(args.initptr) % max_input_buffer_size].length) {
fp_err(FPOL_LAV, "buffer statement mismatch. Encoder exited");
return;
}
ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, (const uint8_t*) samples, computed_buffer_size, 0);
if (ret < 0) {
fprintf(stderr, "Could not setup audio frame\n");
exit(1);
}
/* encode a single tone sound */
// t = 0;
//tincr = 2 * M_PI * 440.0 / c->sample_rate;
args.output_buffer = (char*) malloc(input_buffer_size * (*args.input_buffer.periodbuffer)[(args.initptr) % max_input_buffer_size].length);
args.output_buffer_size = 0;
int non_cycled_termptr = args.initptr + input_buffer_size;
for (i = args.initptr; i < non_cycled_termptr; i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
// for (j = 0; j < c->frame_size; j++) {
// samples[2 * j] = (int) (sin(t) * 10000);
// for (k = 1; k < c->channels; k++) {
// samples[2 * j + k] = samples[2 * j];
// }
// t += tincr;
// }
/* setup the data pointers in the AVFrame */
int k = 0;
int l = computed_buffer_size / 4;
uint16_t* obuffer = (uint16_t*) ((*args.input_buffer.periodbuffer)[i % max_input_buffer_size].period);
for (j = 0; j < computed_buffer_size / 2; j++) {
samples[k++] = obuffer[j];
j++;
samples[l++] = obuffer[j];
}
/* encode the samples */
ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame\n");
exit(1);
}
if (got_output) {
//fwrite(pkt.data, 1, pkt.size, f);
//pkt.data[pkt.size] = 0;
memcpy(args.output_buffer + args.output_buffer_size, pkt.data, pkt.size);
args.output_buffer_size += pkt.size;
av_packet_unref(&pkt);
}
}
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
//fwrite(pkt.data, 1, pkt.size, f);
memcpy(args.output_buffer + args.output_buffer_size, pkt.data, pkt.size);
args.output_buffer_size += pkt.size;
av_packet_unref(&pkt);
}
}
// fclose(f);
av_freep(&samples);
av_frame_free(&frame);
avcodec_close(c);
av_free(c);
}
/*
* Audio encoding example
*/
static void audio_encode_example(const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
AVFrame *frame;
AVPacket pkt;
int i, j, k, ret, got_output;
int buffer_size;
FILE *f;
uint16_t *samples;
float t, tincr;
printf("Encode audio file %s\n", filename);
/* find the MP2 encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
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);
}
/* put sample parameters */
c->bit_rate = 64000;
/* 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, "Encoder does not support sample format %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
/* frame containing input raw audio */
frame = avcodec_alloc_frame();
if (!frame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* the codec gives us the frame size, in samples,
* we calculate the size of the samples buffer in bytes */
buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
c->sample_fmt, 0);
samples = av_malloc(buffer_size);
if (!samples) {
fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
buffer_size);
exit(1);
}
/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
(const uint8_t*)samples, buffer_size, 0);
if (ret < 0) {
fprintf(stderr, "Could not setup audio frame\n");
exit(1);
}
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for(i=0;i<200;i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
for (j = 0; j < c->frame_size; j++) {
samples[2*j] = (int)(sin(t) * 10000);
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
t += tincr;
}
/* encode the samples */
ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
fclose(f);
av_freep(&samples);
avcodec_free_frame(&frame);
avcodec_close(c);
av_free(c);
}
int main(int argc, char **argv)
{
const char *filename;
const AVCodec *codec;
AVCodecContext *c = NULL;
AVFrame *frame;
AVPacket *pkt;
int i, j, k, ret;
FILE *f;
uint16_t *samples;
float t, tincr;
if (argc <= 1) {
fprintf(stderr, "Usage: %s <output file>\n", argv[0]);
return 0;
}
filename = argv[1];
/* find the MP2 encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
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);
}
/* put sample parameters */
c->bit_rate = 64000;
/* 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, "Encoder does not support sample format %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
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);
}
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* allocate the data buffers */
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate audio data buffers\n");
exit(1);
}
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for (i = 0; i < 200; i++) {
/* make sure the frame is writable -- makes a copy if the encoder
* kept a reference internally */
ret = av_frame_make_writable(frame);
if (ret < 0)
exit(1);
samples = (uint16_t*)frame->data[0];
for (j = 0; j < c->frame_size; j++) {
samples[2 * j] = (int)(sin(t) * 10000);
for (k = 1; k < c->channels; k++)
samples[2 * j + k] = samples[2 * j];
t += tincr;
}
encode(c, frame, pkt, f);
}
/* flush the encoder */
encode(c, NULL, pkt, f);
fclose(f);
av_frame_free(&frame);
av_packet_free(&pkt);
avcodec_free_context(&c);
return 0;
}
