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opusenc: Add GstAudioClippingMeta to buffers that need to be clipped

Browse source 
https://bugzilla.gnome.org/show_bug.cgi?id=757153
This commit is contained in:
Sebastian Dröge 2015-11-02 16:52:28 +02:00
parent 6b751360ae
commit 328f9088f3
1 changed files with 20 additions and 4 deletions
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24
ext/opus/gstopusenc.c
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@ -761,8 +761,8 @@ gst_opus_enc_setup (GstOpusEnc * enc)
lookahead);
/* lookahead is samples, the Opus header wants it in 48kHz samples */
enc->lookahead = enc->pending_lookahead = lookahead;
lookahead = lookahead * 48000 / enc->sample_rate;
enc->lookahead = enc->pending_lookahead = lookahead;
gst_opus_header_create_caps (&caps, NULL, lookahead, enc->sample_rate,
enc->n_channels, enc->n_stereo_streams, enc->channel_mapping_family,
@ -894,6 +894,7 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
GstBuffer *outbuf;
GstSegment *segment;
GstClockTime duration;
guint64 trim_start = 0, trim_end = 0;
guint max_payload_size;
gint frame_samples, input_samples, output_samples;
@ -945,6 +946,7 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
"%" G_GINT64_FORMAT " extra samples of padding in this frame",
diff);
output_samples = frame_samples - diff;
trim_end = diff * 48000 / enc->sample_rate;
} else {
GST_DEBUG_OBJECT (enc,
"Need to add %" G_GINT64_FORMAT " extra samples in the next frame",
@ -966,11 +968,16 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
/* Adjust for lookahead here */
if (enc->pending_lookahead) {
if (input_samples > enc->pending_lookahead) {
output_samples = input_samples - enc->pending_lookahead;
guint scaled_lookahead =
enc->pending_lookahead * enc->sample_rate / 48000;
if (input_samples > scaled_lookahead) {
output_samples = input_samples - scaled_lookahead;
trim_start = enc->pending_lookahead;
enc->pending_lookahead = 0;
} else {
enc->pending_lookahead -= input_samples;
trim_start = input_samples * 48000 / enc->sample_rate;
enc->pending_lookahead -= trim_start;
output_samples = 0;
}
} else {
@ -988,6 +995,7 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
output_samples = enc->consumed_samples - enc->encoded_samples;
input_samples = 0;
GST_DEBUG_OBJECT (enc, "draining %d samples", output_samples);
trim_end = (frame_samples - output_samples) * 48000 / enc->sample_rate;
} else if (enc->encoded_samples == enc->consumed_samples) {
GST_DEBUG_OBJECT (enc, "nothing to drain");
goto done;
@ -1008,6 +1016,14 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
frame_samples, (int) bytes);
if (trim_start || trim_end) {
GST_DEBUG_OBJECT (enc,
"Adding trim-start %" G_GUINT64_FORMAT " trim-end %" G_GUINT64_FORMAT,
trim_start, trim_end);
gst_buffer_add_audio_clipping_meta (outbuf, GST_FORMAT_DEFAULT, trim_start,
trim_end);
}
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
outsize =