From ca9528d0b0b62f32e42136647a097fbb80f96e3d Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Tue, 10 Jun 2014 10:04:07 +0100 Subject: [PATCH] speexenc: update output segment stop time to match clipped samples This will let oggmux generate a granpos on the last page that properly represents the clipped samples at the end of the stream. --- ext/speex/gstspeexenc.c | 29 +++++++++++++++++++++++++++++ ext/speex/gstspeexenc.h | 1 + 2 files changed, 30 insertions(+) diff --git a/ext/speex/gstspeexenc.c b/ext/speex/gstspeexenc.c index 52b97ca..79b3cb3 100644 --- a/ext/speex/gstspeexenc.c +++ b/ext/speex/gstspeexenc.c @@ -253,6 +253,7 @@ gst_speex_enc_start (GstAudioEncoder * benc) speex_bits_init (&enc->bits); enc->tags = gst_tag_list_new_empty (); enc->header_sent = FALSE; + enc->encoded_samples = 0; return TRUE; } @@ -511,6 +512,9 @@ gst_speex_enc_sink_event (GstAudioEncoder * benc, GstEvent * event) } break; } + case GST_EVENT_SEGMENT: + enc->encoded_samples = 0; + break; default: break; } @@ -530,6 +534,8 @@ gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf) gsize bsize, size; GstBuffer *outbuf; GstFlowReturn ret = GST_FLOW_OK; + GstSegment *segment; + GstClockTime duration; if (G_LIKELY (buf)) { gst_buffer_map (buf, &map, GST_MAP_READ); @@ -539,6 +545,28 @@ gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf) if (G_UNLIKELY (bsize % bytes)) { GST_DEBUG_OBJECT (enc, "draining; adding silence samples"); + /* If encoding part of a frame, and we have no set stop time on + * the output segment, we update the segment stop time to reflect + * the last sample. This will let oggmux set the last page's + * granpos to tell a decoder the dummy samples should be clipped. + */ + segment = &GST_AUDIO_ENCODER_OUTPUT_SEGMENT (enc); + GST_DEBUG_OBJECT (enc, "existing output segment %" GST_SEGMENT_FORMAT, + segment); + if (!GST_CLOCK_TIME_IS_VALID (segment->stop)) { + int input_samples = bsize / (enc->channels * 2); + GST_DEBUG_OBJECT (enc, + "No stop time and partial frame, updating segment"); + duration = + gst_util_uint64_scale (enc->encoded_samples + input_samples, + GST_SECOND, enc->rate); + segment->stop = segment->start + duration; + GST_DEBUG_OBJECT (enc, "new output segment %" GST_SEGMENT_FORMAT, + segment); + gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc), + gst_event_new_segment (segment)); + } + size = ((bsize / bytes) + 1) * bytes; data0 = data = g_malloc0 (size); memcpy (data, bdata, bsize); @@ -603,6 +631,7 @@ gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf) ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), outbuf, samples); + enc->encoded_samples += frame_size; done: g_free (data0); diff --git a/ext/speex/gstspeexenc.h b/ext/speex/gstspeexenc.h index 8d9568d..b089cb2 100644 --- a/ext/speex/gstspeexenc.h +++ b/ext/speex/gstspeexenc.h @@ -80,6 +80,7 @@ struct _GstSpeexEnc { gint rate; gboolean header_sent; + guint64 encoded_samples; GstTagList *tags; -- 2.7.4