mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-01-21 22:58:16 +00:00
Code Issues 1 Projects Releases Wiki Activity

avauddec: Add clipping meta support for gapless playback

Browse source 
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1028>
This commit is contained in:
Carlos Rafael Giani 2019-09-10 17:30:07 +02:00 committed by Sebastian Dröge
parent 0431a0845c
commit 492dc666df
1 changed files with 63 additions and 8 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

71
subprojects/gst-libav/ext/libav/gstavauddec.c
View file

@ -30,6 +30,7 @@
#include <libavutil/channel_layout.h>
#include <gst/gst.h>
#include <gst/base/gstbytewriter.h>
#include "gstav.h"
#include "gstavcodecmap.h"
@ -463,7 +464,8 @@ gst_avpacket_init (AVPacket * packet, guint8 * data, guint size)
*/
static gboolean
gst_ffmpegauddec_audio_frame (GstFFMpegAudDec * ffmpegdec,
AVCodec * in_plugin, GstBuffer ** outbuf, GstFlowReturn * ret)
AVCodec * in_plugin, GstBuffer ** outbuf, GstFlowReturn * ret,
gboolean * need_more_data)
{
gboolean got_frame = FALSE;
gint res;
@ -533,6 +535,7 @@ gst_ffmpegauddec_audio_frame (GstFFMpegAudDec * ffmpegdec,
GST_BUFFER_FLAG_SET (*outbuf, GST_BUFFER_FLAG_CORRUPTED);
} else if (res == AVERROR (EAGAIN)) {
*outbuf = NULL;
*need_more_data = TRUE;
} else if (res == AVERROR_EOF) {
*ret = GST_FLOW_EOS;
GST_DEBUG_OBJECT (ffmpegdec, "Context was entirely flushed");
@ -552,7 +555,8 @@ beach:
* Returns: whether a frame was decoded
*/
static gboolean
gst_ffmpegauddec_frame (GstFFMpegAudDec * ffmpegdec, GstFlowReturn * ret)
gst_ffmpegauddec_frame (GstFFMpegAudDec * ffmpegdec, GstFlowReturn * ret,
gboolean * need_more_data)
{
GstFFMpegAudDecClass *oclass;
GstBuffer *outbuf = NULL;
@ -567,7 +571,8 @@ gst_ffmpegauddec_frame (GstFFMpegAudDec * ffmpegdec, GstFlowReturn * ret)
oclass = (GstFFMpegAudDecClass *) (G_OBJECT_GET_CLASS (ffmpegdec));
got_frame =
gst_ffmpegauddec_audio_frame (ffmpegdec, oclass->in_plugin, &outbuf, ret);
gst_ffmpegauddec_audio_frame (ffmpegdec, oclass->in_plugin, &outbuf, ret,
need_more_data);
if (outbuf) {
GST_LOG_OBJECT (ffmpegdec, "Decoded data, buffer %" GST_PTR_FORMAT, outbuf);
@ -594,16 +599,17 @@ gst_ffmpegauddec_drain (GstFFMpegAudDec * ffmpegdec, gboolean force)
{
GstFlowReturn ret = GST_FLOW_OK;
gboolean got_any_frames = FALSE;
gboolean need_more_data = FALSE;
gboolean got_frame;
if (avcodec_send_packet (ffmpegdec->context, NULL))
goto send_packet_failed;
do {
got_frame = gst_ffmpegauddec_frame (ffmpegdec, &ret);
got_frame = gst_ffmpegauddec_frame (ffmpegdec, &ret, &need_more_data);
if (got_frame)
got_any_frames = TRUE;
} while (got_frame);
} while (got_frame && !need_more_data);
avcodec_flush_buffers (ffmpegdec->context);
/* FFMpeg will return AVERROR_EOF if it's internal was fully drained
@ -653,6 +659,10 @@ gst_ffmpegauddec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf)
GstFlowReturn ret = GST_FLOW_OK;
gboolean is_header;
AVPacket packet;
GstAudioClippingMeta *clipping_meta = NULL;
guint32 num_clipped_samples = 0;
gboolean fully_clipped = FALSE;
gboolean need_more_data = FALSE;
ffmpegdec = (GstFFMpegAudDec *) decoder;
@ -684,6 +694,18 @@ gst_ffmpegauddec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf)
inbuf = gst_buffer_make_writable (inbuf);
}
/* mpegaudioparse is setting buffer flags for the Xing/LAME header. This
* should not be passed to the decoder as it results in unnecessary silence
* samples to be output */
if (oclass->in_plugin->id == AV_CODEC_ID_MP3 &&
GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DECODE_ONLY) &&
GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DROPPABLE)) {
gst_buffer_unref (inbuf);
return gst_audio_decoder_finish_frame (decoder, NULL, 1);
}
clipping_meta = gst_buffer_get_audio_clipping_meta (inbuf);
gst_buffer_map (inbuf, &map, GST_MAP_READ);
data = map.data;
@ -711,13 +733,37 @@ gst_ffmpegauddec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf)
if (!packet.size)
goto unmap;
if (clipping_meta != NULL) {
if (clipping_meta->format == GST_FORMAT_DEFAULT) {
uint8_t *p = av_packet_new_side_data (&packet, AV_PKT_DATA_SKIP_SAMPLES,
10);
if (p != NULL) {
GstByteWriter writer;
guint32 start = clipping_meta->start;
guint32 end = clipping_meta->end;
num_clipped_samples = start + end;
gst_byte_writer_init_with_data (&writer, p, 10, FALSE);
gst_byte_writer_put_uint32_le (&writer, start);
gst_byte_writer_put_uint32_le (&writer, end);
GST_LOG_OBJECT (ffmpegdec, "buffer has clipping metadata; added skip "
"side data to avpacket with start %u and end %u", start, end);
}
} else {
GST_WARNING_OBJECT (ffmpegdec,
"buffer has clipping metadata in unsupported format %s",
gst_format_get_name (clipping_meta->format));
}
}
if (avcodec_send_packet (ffmpegdec->context, &packet) < 0) {
goto send_packet_failed;
}
do {
/* decode a frame of audio now */
got_frame = gst_ffmpegauddec_frame (ffmpegdec, &ret);
got_frame = gst_ffmpegauddec_frame (ffmpegdec, &ret, &need_more_data);
if (got_frame)
got_any_frames = TRUE;
@ -728,9 +774,18 @@ gst_ffmpegauddec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf)
/* bad flow return, make sure we discard all data and exit */
break;
}
} while (got_frame);
} while (got_frame && !need_more_data);
if (is_header || got_any_frames) {
/* The frame was fully clipped if we have samples to be clipped and
* it's either more than the known fixed frame size, or the decoder returned
* that it needs more data (EAGAIN) and we didn't decode any frames at all.
*/
fully_clipped = (clipping_meta != NULL && num_clipped_samples > 0)
&& ((ffmpegdec->context->frame_size != 0
&& num_clipped_samples >= ffmpegdec->context->frame_size)
|| (need_more_data && !got_any_frames));
if (is_header || got_any_frames || fully_clipped) {
/* Even if previous return wasn't GST_FLOW_OK, we need to call
* _finish_frame() since baseclass is expecting that _finish_frame()
* is followed by _finish_subframe()