void
write_audio_frame (GeglProperties *o, AVFormatContext * oc, AVStream * st)
{
Priv *p = (Priv*)o->user_data;
AVCodecContext *c = st->codec;
int sample_count = 100000;
static AVPacket pkt = { 0 };
if (pkt.size == 0)
{
av_init_packet (&pkt);
}
/* first we add incoming frames audio samples */
{
int i;
int sample_count = gegl_audio_fragment_get_sample_count (o->audio);
GeglAudioFragment *af = gegl_audio_fragment_new (gegl_audio_fragment_get_sample_rate (o->audio),
gegl_audio_fragment_get_channels (o->audio),
gegl_audio_fragment_get_channel_layout (o->audio),
sample_count);
gegl_audio_fragment_set_sample_count (af, sample_count);
for (i = 0; i < sample_count; i++)
{
af->data[0][i] = o->audio->data[0][i];
af->data[1][i] = o->audio->data[1][i];
}
gegl_audio_fragment_set_pos (af, p->audio_pos);
p->audio_pos += sample_count;
p->audio_track = g_list_append (p->audio_track, af);
}
if (!(c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE))
sample_count = c->frame_size;
/* then we encode as much as we can in a loop using the codec frame size */
while (p->audio_pos - p->audio_read_pos > sample_count)
{
long i;
int ret;
int got_packet = 0;
AVFrame *frame = alloc_audio_frame (c->sample_fmt, c->channel_layout,
c->sample_rate, sample_count);
switch (c->sample_fmt) {
case AV_SAMPLE_FMT_FLT:
for (i = 0; i < sample_count; i++)
{
float left = 0, right = 0;
get_sample_data (p, i + p->audio_read_pos, &left, &right);
((float*)frame->data[0])[c->channels*i+0] = left;
((float*)frame->data[0])[c->channels*i+1] = right;
}
break;
case AV_SAMPLE_FMT_FLTP:
for (i = 0; i < sample_count; i++)
{
float left = 0, right = 0;
get_sample_data (p, i + p->audio_read_pos, &left, &right);
((float*)frame->data[0])[i] = left;
((float*)frame->data[1])[i] = right;
}
break;
case AV_SAMPLE_FMT_S16:
for (i = 0; i < sample_count; i++)
{
float left = 0, right = 0;
get_sample_data (p, i + p->audio_read_pos, &left, &right);
((int16_t*)frame->data[0])[c->channels*i+0] = left * (1<<15);
((int16_t*)frame->data[0])[c->channels*i+1] = right * (1<<15);
}
break;
case AV_SAMPLE_FMT_S32:
for (i = 0; i < sample_count; i++)
{
float left = 0, right = 0;
get_sample_data (p, i + p->audio_read_pos, &left, &right);
((int32_t*)frame->data[0])[c->channels*i+0] = left * (1<<31);
((int32_t*)frame->data[0])[c->channels*i+1] = right * (1<<31);
}
break;
case AV_SAMPLE_FMT_S32P:
for (i = 0; i < sample_count; i++)
{
float left = 0, right = 0;
get_sample_data (p, i + p->audio_read_pos, &left, &right);
((int32_t*)frame->data[0])[i] = left * (1<<31);
((int32_t*)frame->data[1])[i] = right * (1<<31);
}
break;
case AV_SAMPLE_FMT_S16P:
for (i = 0; i < sample_count; i++)
{
float left = 0, right = 0;
get_sample_data (p, i + p->audio_read_pos, &left, &right);
((int16_t*)frame->data[0])[i] = left * (1<<15);
((int16_t*)frame->data[1])[i] = right * (1<<15);
}
break;
default:
fprintf (stderr, "eeeek unhandled audio format\n");
break;
}
frame->pts = p->next_apts;
p->next_apts += sample_count;
av_frame_make_writable (frame);
ret = avcodec_encode_audio2 (c, &pkt, frame, &got_packet);
av_packet_rescale_ts (&pkt, st->codec->time_base, st->time_base);
if (ret < 0) {
fprintf (stderr, "Error encoding audio frame: %s\n", av_err2str (ret));
}
if (got_packet)
{
pkt.stream_index = st->index;
av_interleaved_write_frame (oc, &pkt);
av_free_packet (&pkt);
}
av_frame_free (&frame);
p->audio_read_pos += sample_count;
}
}
/* maintain list of audio samples */
static int
decode_audio (GeglOperation *operation,
gdouble pts1,
gdouble pts2)
{
GeglProperties *o = GEGL_PROPERTIES (operation);
Priv *p = (Priv*)o->user_data;
pts1 -= 2.0;
if (pts1 < 0.0)pts1 = 0.0;
if(pts1 - 15.0 > p->prevapts){
int64_t seek_target = av_rescale_q (pts1 * AV_TIME_BASE, AV_TIME_BASE_Q, p->audio_stream->time_base);
clear_audio_track (o);
p->prevapts = 0.0;
if (av_seek_frame (p->audio_fcontext, p->audio_stream->index, seek_target, (AVSEEK_FLAG_BACKWARD)) < 0)
fprintf (stderr, "audio seek error!\n");
else
avcodec_flush_buffers (p->audio_stream->codec);
}
while (p->prevapts <= pts2)
{
AVPacket pkt = {0,};
int decoded_bytes;
if (av_read_frame (p->audio_fcontext, &pkt) < 0)
{
av_free_packet (&pkt);
return -1;
}
if (pkt.stream_index==p->audio_index && p->audio_stream)
{
static AVFrame frame;
int got_frame;
decoded_bytes = avcodec_decode_audio4(p->audio_stream->codec,
&frame, &got_frame, &pkt);
if (decoded_bytes < 0)
{
fprintf (stderr, "avcodec_decode_audio4 failed for %s\n",
o->path);
}
if (got_frame) {
int samples_left = frame.nb_samples;
int si = 0;
while (samples_left)
{
int sample_count = samples_left;
int channels = MIN(p->audio_stream->codec->channels, GEGL_MAX_AUDIO_CHANNELS);
GeglAudioFragment *af = gegl_audio_fragment_new (o->audio_sample_rate, channels,
AV_CH_LAYOUT_STEREO, samples_left);
//);
switch (p->audio_stream->codec->sample_fmt)
{
case AV_SAMPLE_FMT_FLT:
for (gint i = 0; i < sample_count; i++)
for (gint c = 0; c < channels; c++)
af->data[c][i] = ((int16_t *)frame.data[0])[(i + si) * channels + c];
break;
case AV_SAMPLE_FMT_FLTP:
for (gint i = 0; i < sample_count; i++)
for (gint c = 0; c < channels; c++)
{
af->data[c][i] = ((float *)frame.data[c])[i + si];
}
break;
case AV_SAMPLE_FMT_S16:
for (gint i = 0; i < sample_count; i++)
for (gint c = 0; c < channels; c++)
af->data[c][i] = ((int16_t *)frame.data[0])[(i + si) * channels + c] / 32768.0;
break;
case AV_SAMPLE_FMT_S16P:
for (gint i = 0; i < sample_count; i++)
for (gint c = 0; c < channels; c++)
af->data[c][i] = ((int16_t *)frame.data[c])[i + si] / 32768.0;
break;
case AV_SAMPLE_FMT_S32:
for (gint i = 0; i < sample_count; i++)
for (gint c = 0; c < channels; c++)
af->data[c][i] = ((int32_t *)frame.data[0])[(i + si) * channels + c] / 2147483648.0;
break;
case AV_SAMPLE_FMT_S32P:
for (gint i = 0; i < sample_count; i++)
for (gint c = 0; c < channels; c++)
af->data[c][i] = ((int32_t *)frame.data[c])[i + si] / 2147483648.0;
break;
default:
g_warning ("undealt with sample format\n");
}
gegl_audio_fragment_set_sample_count (af, sample_count);
gegl_audio_fragment_set_pos (af,
(long int)av_rescale_q ((pkt.pts), p->audio_stream->time_base, AV_TIME_BASE_Q) * o->audio_sample_rate /AV_TIME_BASE);
p->audio_pos += sample_count;
p->audio_track = g_list_append (p->audio_track, af);
samples_left -= sample_count;
si += sample_count;
}
p->prevapts = pkt.pts * av_q2d (p->audio_stream->time_base);
}
}
av_free_packet (&pkt);
}
return 0;
}
