mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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3d500636a9
It is only meant to be used as a callback. The fallback macro uses strcmp which doesn't handle NULL strings gracefully. Instead use g_strcmp0 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6392>
2145 lines
75 KiB
C
2145 lines
75 KiB
C
/* GStreamer
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* Copyright (C) 2010 Marc-Andre Lureau <marcandre.lureau@gmail.com>
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* Copyright (C) 2010 Andoni Morales Alastruey <ylatuya@gmail.com>
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* Copyright (C) 2011, Hewlett-Packard Development Company, L.P.
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* Author: Youness Alaoui <youness.alaoui@collabora.co.uk>, Collabora Ltd.
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* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>, Collabora Ltd.
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* Copyright (C) 2014 Sebastian Dröge <sebastian@centricular.com>
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* Copyright (C) 2015 Tim-Philipp Müller <tim@centricular.com>
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*
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* Copyright (C) 2021-2022 Centricular Ltd
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* Author: Edward Hervey <edward@centricular.com>
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* Author: Jan Schmidt <jan@centricular.com>
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*
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* gsthlsdemux-stream.c:
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <string.h>
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#include <gst/base/gsttypefindhelper.h>
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#include <gst/tag/tag.h>
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#include <glib/gi18n-lib.h>
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#include "gsthlsdemux.h"
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#include "gsthlsdemux-stream.h"
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GST_DEBUG_CATEGORY_EXTERN (gst_hls_demux2_debug);
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#define GST_CAT_DEFAULT gst_hls_demux2_debug
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/* Maximum values for mpeg-ts DTS values */
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#define MPEG_TS_MAX_PTS (((((guint64)1) << 33) * (guint64)100000) / 9)
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static GstBuffer *gst_hls_demux_decrypt_fragment (GstHLSDemux * demux,
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GstHLSDemuxStream * stream, GstBuffer * encrypted_buffer, GError ** err);
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static gboolean
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gst_hls_demux_stream_decrypt_start (GstHLSDemuxStream * stream,
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const guint8 * key_data, const guint8 * iv_data);
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static void gst_hls_demux_stream_decrypt_end (GstHLSDemuxStream * stream);
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static gboolean
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gst_hls_demux_stream_start_fragment (GstAdaptiveDemux2Stream * stream);
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static GstFlowReturn
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gst_hls_demux_stream_finish_fragment (GstAdaptiveDemux2Stream * stream);
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static GstFlowReturn gst_hls_demux_stream_data_received (GstAdaptiveDemux2Stream
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* stream, GstBuffer * buffer);
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static gboolean gst_hls_demux_stream_has_next_fragment (GstAdaptiveDemux2Stream
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* stream);
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static GstFlowReturn
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gst_hls_demux_stream_advance_fragment (GstAdaptiveDemux2Stream * stream);
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static GstFlowReturn
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gst_hls_demux_stream_update_fragment_info (GstAdaptiveDemux2Stream * stream);
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static GstFlowReturn
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gst_hls_demux_stream_submit_request (GstAdaptiveDemux2Stream * stream,
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DownloadRequest * download_req);
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static void gst_hls_demux_stream_start (GstAdaptiveDemux2Stream * stream);
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static void gst_hls_demux_stream_stop (GstAdaptiveDemux2Stream * stream);
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static void gst_hls_demux_stream_create_tracks (GstAdaptiveDemux2Stream *
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stream);
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static gboolean gst_hls_demux_stream_select_bitrate (GstAdaptiveDemux2Stream *
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stream, guint64 bitrate);
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static GstClockTime
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gst_hls_demux_stream_get_presentation_offset (GstAdaptiveDemux2Stream * stream);
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static void gst_hls_demux_stream_finalize (GObject * object);
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#define gst_hls_demux_stream_parent_class stream_parent_class
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G_DEFINE_TYPE (GstHLSDemuxStream, gst_hls_demux_stream,
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GST_TYPE_ADAPTIVE_DEMUX2_STREAM);
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static void
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gst_hls_demux_stream_class_init (GstHLSDemuxStreamClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstAdaptiveDemux2StreamClass *adaptivedemux2stream_class =
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GST_ADAPTIVE_DEMUX2_STREAM_CLASS (klass);
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gobject_class->finalize = gst_hls_demux_stream_finalize;
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adaptivedemux2stream_class->update_fragment_info =
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gst_hls_demux_stream_update_fragment_info;
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adaptivedemux2stream_class->submit_request =
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gst_hls_demux_stream_submit_request;
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adaptivedemux2stream_class->has_next_fragment =
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gst_hls_demux_stream_has_next_fragment;
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adaptivedemux2stream_class->stream_seek = gst_hls_demux_stream_seek;
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adaptivedemux2stream_class->advance_fragment =
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gst_hls_demux_stream_advance_fragment;
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adaptivedemux2stream_class->select_bitrate =
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gst_hls_demux_stream_select_bitrate;
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adaptivedemux2stream_class->start = gst_hls_demux_stream_start;
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adaptivedemux2stream_class->stop = gst_hls_demux_stream_stop;
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adaptivedemux2stream_class->create_tracks =
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gst_hls_demux_stream_create_tracks;
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adaptivedemux2stream_class->start_fragment =
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gst_hls_demux_stream_start_fragment;
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adaptivedemux2stream_class->finish_fragment =
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gst_hls_demux_stream_finish_fragment;
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adaptivedemux2stream_class->data_received =
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gst_hls_demux_stream_data_received;
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adaptivedemux2stream_class->get_presentation_offset =
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gst_hls_demux_stream_get_presentation_offset;
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}
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static void
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gst_hls_demux_stream_init (GstHLSDemuxStream * stream)
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{
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stream->parser_type = GST_HLS_PARSER_NONE;
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stream->do_typefind = TRUE;
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stream->reset_pts = TRUE;
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stream->presentation_offset = 60 * GST_SECOND;
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stream->pdt_tag_sent = FALSE;
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}
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void
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gst_hls_demux_stream_clear_pending_data (GstHLSDemuxStream * hls_stream,
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gboolean force)
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{
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GST_DEBUG_OBJECT (hls_stream, "force : %d", force);
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if (hls_stream->pending_encrypted_data)
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gst_adapter_clear (hls_stream->pending_encrypted_data);
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gst_buffer_replace (&hls_stream->pending_decrypted_buffer, NULL);
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gst_buffer_replace (&hls_stream->pending_typefind_buffer, NULL);
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if (force || !hls_stream->pending_data_is_header) {
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gst_buffer_replace (&hls_stream->pending_segment_data, NULL);
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hls_stream->pending_data_is_header = FALSE;
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}
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hls_stream->current_offset = -1;
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hls_stream->process_buffer_content = TRUE;
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gst_hls_demux_stream_decrypt_end (hls_stream);
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}
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GstFlowReturn
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gst_hls_demux_stream_seek (GstAdaptiveDemux2Stream * stream, gboolean forward,
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GstSeekFlags flags, GstClockTimeDiff ts, GstClockTimeDiff * final_ts)
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{
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GstFlowReturn ret = GST_FLOW_OK;
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GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (stream);
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GstHLSDemux *hlsdemux = (GstHLSDemux *) stream->demux;
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GST_DEBUG_OBJECT (stream,
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"is_variant:%d media:%p current_variant:%p forward:%d ts:%"
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GST_TIME_FORMAT, hls_stream->is_variant, hls_stream->current_rendition,
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hlsdemux->current_variant, forward, GST_TIME_ARGS (ts));
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/* If this stream doesn't have a playlist yet, we can't seek on it */
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if (!hls_stream->playlist_fetched) {
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return GST_ADAPTIVE_DEMUX_FLOW_BUSY;
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}
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/* Allow jumping to partial segments in the last 2 segments in LL-HLS */
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if (GST_HLS_MEDIA_PLAYLIST_IS_LIVE (hls_stream->playlist))
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flags |= GST_HLS_M3U8_SEEK_FLAG_ALLOW_PARTIAL;
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GstM3U8SeekResult seek_result;
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if (gst_hls_media_playlist_seek (hls_stream->playlist, forward, flags, ts,
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&seek_result)) {
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if (hls_stream->current_segment)
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gst_m3u8_media_segment_unref (hls_stream->current_segment);
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hls_stream->current_segment = seek_result.segment;
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hls_stream->in_partial_segments = seek_result.found_partial_segment;
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hls_stream->part_idx = seek_result.part_idx;
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hls_stream->reset_pts = TRUE;
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if (final_ts)
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*final_ts = seek_result.stream_time;
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} else {
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GST_WARNING_OBJECT (stream, "Seeking failed");
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ret = GST_FLOW_ERROR;
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}
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return ret;
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}
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static GstCaps *
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get_caps_of_stream_type (GstCaps * full_caps, GstStreamType streamtype)
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{
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GstCaps *ret = NULL;
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guint i;
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for (i = 0; i < gst_caps_get_size (full_caps); i++) {
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GstStructure *st = gst_caps_get_structure (full_caps, i);
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if (gst_hls_get_stream_type_from_structure (st) == streamtype) {
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ret = gst_caps_new_empty ();
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gst_caps_append_structure (ret, gst_structure_copy (st));
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break;
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}
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}
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return ret;
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}
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static GstHLSRenditionStream *
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find_uriless_rendition (GstHLSDemux * demux, GstStreamType stream_type)
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{
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GList *tmp;
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for (tmp = demux->master->renditions; tmp; tmp = tmp->next) {
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GstHLSRenditionStream *media = tmp->data;
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if (media->uri == NULL &&
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gst_stream_type_from_hls_type (media->mtype) == stream_type)
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return media;
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}
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return NULL;
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}
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static void
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gst_hls_demux_stream_create_tracks (GstAdaptiveDemux2Stream * stream)
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{
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GstHLSDemux *hlsdemux = (GstHLSDemux *) stream->demux;
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GstHLSDemuxStream *hlsdemux_stream = (GstHLSDemuxStream *) stream;
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guint i;
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GstStreamType uriless_types = 0;
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GstCaps *variant_caps = NULL;
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GST_DEBUG_OBJECT (stream, "Update tracks of variant stream");
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if (hlsdemux->master->have_codecs) {
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variant_caps = gst_hls_master_playlist_get_common_caps (hlsdemux->master);
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}
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/* Use the stream->stream_collection and manifest to create the appropriate tracks */
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for (i = 0; i < gst_stream_collection_get_size (stream->stream_collection);
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i++) {
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GstStream *gst_stream =
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gst_stream_collection_get_stream (stream->stream_collection, i);
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GstStreamType stream_type = gst_stream_get_stream_type (gst_stream);
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GstAdaptiveDemuxTrack *track;
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GstHLSRenditionStream *embedded_media = NULL;
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/* tracks from the variant streams should be prefered over those provided by renditions */
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GstStreamFlags flags =
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gst_stream_get_stream_flags (gst_stream) | GST_STREAM_FLAG_SELECT;
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GstCaps *manifest_caps = NULL;
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if (stream_type == GST_STREAM_TYPE_UNKNOWN)
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continue;
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if (variant_caps)
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manifest_caps = get_caps_of_stream_type (variant_caps, stream_type);
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hlsdemux_stream->rendition_type |= stream_type;
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if ((uriless_types & stream_type) == 0) {
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/* Do we have a uriless media for this stream type */
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/* Find if there is a rendition without URI, it will be provided by this variant */
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embedded_media = find_uriless_rendition (hlsdemux, stream_type);
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/* Remember we used this type for a embedded media */
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uriless_types |= stream_type;
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}
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if (embedded_media) {
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GstTagList *tags = gst_stream_get_tags (gst_stream);
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GST_DEBUG_OBJECT (stream, "Adding track '%s' to main variant stream",
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embedded_media->name);
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track =
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gst_hls_demux_new_track_for_rendition (hlsdemux, embedded_media,
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manifest_caps, flags,
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tags ? gst_tag_list_make_writable (tags) : tags);
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} else {
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gchar *stream_id;
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stream_id =
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g_strdup_printf ("main-%s-%d", gst_stream_type_get_name (stream_type),
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i);
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GST_DEBUG_OBJECT (stream, "Adding track '%s' to main variant stream",
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stream_id);
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track =
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gst_adaptive_demux_track_new (stream->demux, stream_type,
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flags, stream_id, manifest_caps, NULL);
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g_free (stream_id);
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}
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track->upstream_stream_id =
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g_strdup (gst_stream_get_stream_id (gst_stream));
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gst_adaptive_demux2_stream_add_track (stream, track);
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gst_adaptive_demux_track_unref (track);
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}
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if (variant_caps)
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gst_caps_unref (variant_caps);
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/* Update the stream object with rendition types.
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* FIXME: rendition_type could be removed */
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stream->stream_type = hlsdemux_stream->rendition_type;
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}
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static gboolean
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gst_hls_demux_stream_start_fragment (GstAdaptiveDemux2Stream * stream)
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{
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GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (stream);
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GstHLSDemux *hlsdemux = GST_HLS_DEMUX_CAST (stream->demux);
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const GstHLSKey *key;
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GstHLSMediaPlaylist *m3u8;
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GST_DEBUG_OBJECT (stream, "Fragment starting");
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gst_hls_demux_stream_clear_pending_data (hls_stream, FALSE);
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/* If no decryption is needed, there's nothing to be done here */
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if (hls_stream->current_key == NULL)
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return TRUE;
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m3u8 = hls_stream->playlist;
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key = gst_hls_demux_get_key (hlsdemux, hls_stream->current_key,
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m3u8->uri, m3u8->allowcache);
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if (key == NULL)
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goto key_failed;
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if (!gst_hls_demux_stream_decrypt_start (hls_stream, key->data,
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hls_stream->current_iv))
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goto decrypt_start_failed;
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return TRUE;
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key_failed:
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{
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GST_ELEMENT_ERROR (hlsdemux, STREAM, DECRYPT_NOKEY,
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("Couldn't retrieve key for decryption"), (NULL));
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GST_WARNING_OBJECT (hlsdemux, "Failed to decrypt data");
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return FALSE;
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}
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decrypt_start_failed:
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{
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GST_ELEMENT_ERROR (hlsdemux, STREAM, DECRYPT, ("Failed to start decrypt"),
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("Couldn't set key and IV or plugin was built without crypto library"));
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return FALSE;
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}
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}
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static GstHLSParserType
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caps_to_parser_type (const GstCaps * caps)
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{
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const GstStructure *s = gst_caps_get_structure (caps, 0);
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if (gst_structure_has_name (s, "video/mpegts"))
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return GST_HLS_PARSER_MPEGTS;
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if (gst_structure_has_name (s, "application/x-id3"))
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return GST_HLS_PARSER_ID3;
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if (gst_structure_has_name (s, "application/x-subtitle-vtt"))
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return GST_HLS_PARSER_WEBVTT;
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if (gst_structure_has_name (s, "video/quicktime"))
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return GST_HLS_PARSER_ISOBMFF;
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|
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return GST_HLS_PARSER_NONE;
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}
|
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|
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/* Identify the nature of data for this stream
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*
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* Will also setup the appropriate parser (tsreader) if needed
|
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*
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* Consumes the input buffer when it returns FALSE, but
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* replaces / returns the input buffer in the `buffer` parameter
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* when it returns TRUE.
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*
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* Returns TRUE if we are done with typefinding */
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static gboolean
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gst_hls_demux_typefind_stream (GstHLSDemux * hlsdemux,
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GstAdaptiveDemux2Stream * stream, GstBuffer ** out_buffer, gboolean at_eos,
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GstFlowReturn * ret)
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{
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GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (stream); // FIXME: pass HlsStream into function
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GstCaps *caps = NULL;
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guint buffer_size;
|
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GstTypeFindProbability prob = GST_TYPE_FIND_NONE;
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GstMapInfo info;
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GstBuffer *buffer = *out_buffer;
|
|
|
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if (hls_stream->pending_typefind_buffer) {
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/* Append to the existing typefind buffer and create a new one that
|
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* we'll return (or consume below) */
|
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buffer = *out_buffer =
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gst_buffer_append (hls_stream->pending_typefind_buffer, buffer);
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hls_stream->pending_typefind_buffer = NULL;
|
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}
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|
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gst_buffer_map (buffer, &info, GST_MAP_READ);
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buffer_size = info.size;
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/* Typefind could miss if buffer is too small. In this case we
|
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* will retry later */
|
|
if (buffer_size >= (2 * 1024) || at_eos) {
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caps =
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gst_type_find_helper_for_data (GST_OBJECT_CAST (hlsdemux), info.data,
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info.size, &prob);
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}
|
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|
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if (G_UNLIKELY (!caps)) {
|
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/* Won't need this mapping any more all paths return inside this if() */
|
|
gst_buffer_unmap (buffer, &info);
|
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|
|
/* Only fail typefinding if we already a good amount of data
|
|
* and we still don't know the type */
|
|
if (buffer_size > (2 * 1024 * 1024) || at_eos) {
|
|
GST_ELEMENT_ERROR (hlsdemux, STREAM, TYPE_NOT_FOUND,
|
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("Could not determine type of stream"), (NULL));
|
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gst_buffer_unref (buffer);
|
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*ret = GST_FLOW_NOT_NEGOTIATED;
|
|
} else {
|
|
GST_LOG_OBJECT (stream, "Not enough data to typefind");
|
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hls_stream->pending_typefind_buffer = buffer; /* Transfer the ref */
|
|
*ret = GST_FLOW_OK;
|
|
}
|
|
*out_buffer = NULL;
|
|
return FALSE;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (stream,
|
|
"Typefind result: %" GST_PTR_FORMAT " prob:%d", caps, prob);
|
|
|
|
if (hls_stream->parser_type == GST_HLS_PARSER_NONE) {
|
|
hls_stream->parser_type = caps_to_parser_type (caps);
|
|
if (hls_stream->parser_type == GST_HLS_PARSER_NONE) {
|
|
GST_WARNING_OBJECT (stream,
|
|
"Unsupported stream type %" GST_PTR_FORMAT, caps);
|
|
GST_MEMDUMP_OBJECT (stream, "unknown data", info.data,
|
|
MIN (info.size, 128));
|
|
gst_buffer_unref (buffer);
|
|
*ret = GST_FLOW_ERROR;
|
|
return FALSE;
|
|
}
|
|
if (hls_stream->parser_type == GST_HLS_PARSER_ISOBMFF)
|
|
hls_stream->presentation_offset = 0;
|
|
}
|
|
|
|
gst_adaptive_demux2_stream_set_caps (stream, caps);
|
|
|
|
hls_stream->do_typefind = FALSE;
|
|
|
|
gst_buffer_unmap (buffer, &info);
|
|
|
|
/* We are done with typefinding. Doesn't consume the input buffer */
|
|
*ret = GST_FLOW_OK;
|
|
return TRUE;
|
|
}
|
|
|
|
/* Compute the stream time for the given internal time, based on the provided
|
|
* time map.
|
|
*
|
|
* Will handle mpeg-ts wraparound. */
|
|
GstClockTimeDiff
|
|
gst_hls_internal_to_stream_time (GstHLSTimeMap * map,
|
|
GstClockTime internal_time)
|
|
{
|
|
if (map->internal_time == GST_CLOCK_TIME_NONE)
|
|
return GST_CLOCK_STIME_NONE;
|
|
|
|
/* Handle MPEG-TS Wraparound */
|
|
if (internal_time < map->internal_time &&
|
|
map->internal_time - internal_time > (MPEG_TS_MAX_PTS / 2))
|
|
internal_time += MPEG_TS_MAX_PTS;
|
|
|
|
return (map->stream_time + internal_time - map->internal_time);
|
|
}
|
|
|
|
/* Handle the internal time discovered on a segment.
|
|
*
|
|
* This function is called by the individual buffer parsers once they have
|
|
* extracted that internal time (which is most of the time based on mpegts time,
|
|
* but can also be ISOBMFF pts).
|
|
*
|
|
* This will update the time map when appropriate.
|
|
*
|
|
* If a synchronization issue is detected, the appropriate steps will be taken
|
|
* and the RESYNC return value will be returned
|
|
*/
|
|
GstHLSParserResult
|
|
gst_hlsdemux_stream_handle_internal_time (GstHLSDemuxStream * hls_stream,
|
|
GstClockTime internal_time)
|
|
{
|
|
GstM3U8MediaSegment *current_segment = hls_stream->current_segment;
|
|
GstHLSTimeMap *map;
|
|
GstClockTimeDiff current_stream_time;
|
|
GstClockTimeDiff real_stream_time, difference;
|
|
|
|
g_return_val_if_fail (current_segment != NULL, GST_HLS_PARSER_RESULT_ERROR);
|
|
|
|
current_stream_time = current_segment->stream_time;
|
|
if (hls_stream->in_partial_segments) {
|
|
/* If the current partial segment is valid, update the stream current position to the partial
|
|
* segment stream_time, otherwise leave it alone and fix it up later when we resync */
|
|
if (current_segment->partial_segments
|
|
&& hls_stream->part_idx < current_segment->partial_segments->len) {
|
|
GstM3U8PartialSegment *part =
|
|
g_ptr_array_index (current_segment->partial_segments,
|
|
hls_stream->part_idx);
|
|
current_stream_time = part->stream_time;
|
|
}
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (hls_stream,
|
|
"Got internal time %" GST_TIME_FORMAT " for current segment stream time %"
|
|
GST_STIME_FORMAT, GST_TIME_ARGS (internal_time),
|
|
GST_STIME_ARGS (current_stream_time));
|
|
|
|
GstHLSDemux *demux =
|
|
GST_HLS_DEMUX_CAST (GST_ADAPTIVE_DEMUX2_STREAM_CAST (hls_stream)->demux);
|
|
map = gst_hls_demux_find_time_map (demux, current_segment->discont_sequence);
|
|
|
|
/* Time mappings will always be created upon initial parsing and when advancing */
|
|
g_assert (map);
|
|
|
|
/* Handle the first internal time of a discont sequence. We can only store/use
|
|
* those values for variant streams. */
|
|
if (!GST_CLOCK_TIME_IS_VALID (map->internal_time)) {
|
|
if (!hls_stream->is_variant) {
|
|
GST_WARNING_OBJECT (hls_stream,
|
|
"Got data from a new discont sequence on a rendition stream, can't validate stream time");
|
|
return GST_HLS_PARSER_RESULT_DONE;
|
|
}
|
|
GST_DEBUG_OBJECT (hls_stream,
|
|
"Updating time map dsn:%" G_GINT64_FORMAT " stream_time:%"
|
|
GST_STIME_FORMAT " internal_time:%" GST_TIME_FORMAT, map->dsn,
|
|
GST_STIME_ARGS (current_stream_time), GST_TIME_ARGS (internal_time));
|
|
/* The stream time for a mapping should always be positive ! */
|
|
g_assert (current_stream_time >= 0);
|
|
|
|
if (hls_stream->parser_type == GST_HLS_PARSER_ISOBMFF)
|
|
hls_stream->presentation_offset = internal_time - current_stream_time;
|
|
|
|
map->stream_time = current_stream_time;
|
|
map->internal_time = internal_time;
|
|
|
|
gst_hls_demux_start_rendition_streams (demux);
|
|
return GST_HLS_PARSER_RESULT_DONE;
|
|
}
|
|
|
|
/* The information in a discont is always valid */
|
|
if (current_segment->discont) {
|
|
GST_DEBUG_OBJECT (hls_stream,
|
|
"DISCONT segment, Updating time map to stream_time:%" GST_STIME_FORMAT
|
|
" internal_time:%" GST_TIME_FORMAT, GST_STIME_ARGS (internal_time),
|
|
GST_TIME_ARGS (current_stream_time));
|
|
map->stream_time = current_stream_time;
|
|
map->internal_time = internal_time;
|
|
return GST_HLS_PARSER_RESULT_DONE;
|
|
}
|
|
|
|
/* Check if the segment is the expected one */
|
|
real_stream_time = gst_hls_internal_to_stream_time (map, internal_time);
|
|
difference = current_stream_time - real_stream_time;
|
|
GST_DEBUG_OBJECT (hls_stream,
|
|
"Segment contains stream time %" GST_STIME_FORMAT
|
|
" difference against expected : %" GST_STIME_FORMAT,
|
|
GST_STIME_ARGS (real_stream_time), GST_STIME_ARGS (difference));
|
|
|
|
if (ABS (difference) > 10 * GST_MSECOND) {
|
|
GstClockTimeDiff wrong_position_threshold =
|
|
hls_stream->current_segment->duration / 2;
|
|
|
|
/* Update the value */
|
|
GST_DEBUG_OBJECT (hls_stream,
|
|
"Updating current stream time to %" GST_STIME_FORMAT,
|
|
GST_STIME_ARGS (real_stream_time));
|
|
|
|
/* For LL-HLS, make sure to update and recalculate stream time from
|
|
* the right partial segment if playing one */
|
|
if (hls_stream->in_partial_segments && hls_stream->part_idx != 0) {
|
|
if (current_segment->partial_segments
|
|
&& hls_stream->part_idx < current_segment->partial_segments->len) {
|
|
GstM3U8PartialSegment *part =
|
|
g_ptr_array_index (current_segment->partial_segments,
|
|
hls_stream->part_idx);
|
|
part->stream_time = real_stream_time;
|
|
|
|
gst_hls_media_playlist_recalculate_stream_time_from_part
|
|
(hls_stream->playlist, hls_stream->current_segment,
|
|
hls_stream->part_idx);
|
|
|
|
/* When playing partial segments, the "Wrong position" threshold should be
|
|
* half the part duration */
|
|
wrong_position_threshold = part->duration / 2;
|
|
}
|
|
} else {
|
|
/* Aligned to the start of the segment, update there */
|
|
current_segment->stream_time = real_stream_time;
|
|
gst_hls_media_playlist_recalculate_stream_time (hls_stream->playlist,
|
|
hls_stream->current_segment);
|
|
}
|
|
gst_hls_media_playlist_dump (hls_stream->playlist);
|
|
|
|
if (ABS (difference) > wrong_position_threshold) {
|
|
GstAdaptiveDemux2Stream *stream = (GstAdaptiveDemux2Stream *) hls_stream;
|
|
GstM3U8SeekResult seek_result;
|
|
|
|
/* We are at the wrong segment, try to figure out the *actual* segment */
|
|
GST_DEBUG_OBJECT (hls_stream,
|
|
"Trying to find the correct segment in the playlist for %"
|
|
GST_STIME_FORMAT, GST_STIME_ARGS (current_stream_time));
|
|
if (gst_hls_media_playlist_find_position (hls_stream->playlist,
|
|
current_stream_time, hls_stream->in_partial_segments,
|
|
&seek_result)) {
|
|
|
|
GST_DEBUG_OBJECT (hls_stream, "Synced to position %" GST_STIME_FORMAT,
|
|
GST_STIME_ARGS (seek_result.stream_time));
|
|
|
|
gst_m3u8_media_segment_unref (hls_stream->current_segment);
|
|
hls_stream->current_segment = seek_result.segment;
|
|
hls_stream->in_partial_segments = seek_result.found_partial_segment;
|
|
hls_stream->part_idx = seek_result.part_idx;
|
|
|
|
/* Ask parent class to restart this fragment */
|
|
return GST_HLS_PARSER_RESULT_RESYNC;
|
|
}
|
|
|
|
GST_WARNING_OBJECT (hls_stream,
|
|
"Could not find a replacement stream, carrying on with segment");
|
|
stream->discont = TRUE;
|
|
stream->fragment.stream_time = real_stream_time;
|
|
}
|
|
}
|
|
|
|
return GST_HLS_PARSER_RESULT_DONE;
|
|
}
|
|
|
|
static GstHLSParserResult
|
|
gst_hls_demux_handle_buffer_content (GstHLSDemux * demux,
|
|
GstHLSDemuxStream * hls_stream, gboolean draining, GstBuffer ** buffer)
|
|
{
|
|
GstHLSTimeMap *map;
|
|
GstAdaptiveDemux2Stream *stream = (GstAdaptiveDemux2Stream *) hls_stream;
|
|
GstClockTimeDiff current_stream_time =
|
|
hls_stream->current_segment->stream_time;
|
|
GstClockTime current_duration = hls_stream->current_segment->duration;
|
|
GstHLSParserResult parser_ret;
|
|
|
|
GST_LOG_OBJECT (stream,
|
|
"stream_time:%" GST_STIME_FORMAT " duration:%" GST_TIME_FORMAT
|
|
" discont:%d draining:%d header:%d index:%d",
|
|
GST_STIME_ARGS (current_stream_time), GST_TIME_ARGS (current_duration),
|
|
hls_stream->current_segment->discont, draining,
|
|
stream->downloading_header, stream->downloading_index);
|
|
|
|
/* FIXME : Replace the boolean parser return value (and this function's return
|
|
* value) by an enum which clearly specifies whether:
|
|
*
|
|
* * The content parsing happened succesfully and it no longer needs to be
|
|
* called for the remainder of this fragment
|
|
* * More data is needed in order to parse the data
|
|
* * There was a fatal error parsing the contents (ex: invalid/incompatible
|
|
* content)
|
|
* * The computed fragment stream time is out of sync
|
|
*/
|
|
|
|
g_assert (demux->mappings);
|
|
map =
|
|
gst_hls_demux_find_time_map (demux,
|
|
hls_stream->current_segment->discont_sequence);
|
|
if (!map) {
|
|
/* For rendition streams, we can't do anything without time mapping */
|
|
if (!hls_stream->is_variant) {
|
|
GST_DEBUG_OBJECT (stream,
|
|
"No available time mapping for dsn:%" G_GINT64_FORMAT
|
|
" using estimated stream time",
|
|
hls_stream->current_segment->discont_sequence);
|
|
goto out_done;
|
|
}
|
|
|
|
/* Variants will be able to fill in the the time mapping, so we can carry on without a time mapping */
|
|
} else {
|
|
GST_DEBUG_OBJECT (stream,
|
|
"Using mapping dsn:%" G_GINT64_FORMAT " stream_time:%" GST_TIME_FORMAT
|
|
" internal_time:%" GST_TIME_FORMAT, map->dsn,
|
|
GST_TIME_ARGS (map->stream_time), GST_TIME_ARGS (map->internal_time));
|
|
}
|
|
|
|
switch (hls_stream->parser_type) {
|
|
case GST_HLS_PARSER_MPEGTS:
|
|
parser_ret =
|
|
gst_hlsdemux_handle_content_mpegts (demux, hls_stream, draining,
|
|
buffer);
|
|
break;
|
|
case GST_HLS_PARSER_ID3:
|
|
parser_ret =
|
|
gst_hlsdemux_handle_content_id3 (demux, hls_stream, draining, buffer);
|
|
break;
|
|
case GST_HLS_PARSER_WEBVTT:
|
|
{
|
|
/* Furthermore it will handle timeshifting itself */
|
|
parser_ret =
|
|
gst_hlsdemux_handle_content_webvtt (demux, hls_stream, draining,
|
|
buffer);
|
|
break;
|
|
}
|
|
case GST_HLS_PARSER_ISOBMFF:
|
|
parser_ret =
|
|
gst_hlsdemux_handle_content_isobmff (demux, hls_stream, draining,
|
|
buffer);
|
|
break;
|
|
case GST_HLS_PARSER_NONE:
|
|
default:
|
|
{
|
|
GST_ERROR_OBJECT (stream, "Unknown stream type");
|
|
goto out_error;
|
|
}
|
|
}
|
|
|
|
if (parser_ret == GST_HLS_PARSER_RESULT_NEED_MORE_DATA) {
|
|
if (stream->downloading_index || stream->downloading_header)
|
|
goto out_need_more;
|
|
/* Else if we're draining, it's an error */
|
|
if (draining)
|
|
goto out_error;
|
|
/* Else we just need more data */
|
|
goto out_need_more;
|
|
}
|
|
|
|
if (parser_ret == GST_HLS_PARSER_RESULT_ERROR)
|
|
goto out_error;
|
|
|
|
if (parser_ret == GST_HLS_PARSER_RESULT_RESYNC)
|
|
goto out_resync;
|
|
|
|
out_done:
|
|
GST_DEBUG_OBJECT (stream, "Done. Finished parsing");
|
|
return GST_HLS_PARSER_RESULT_DONE;
|
|
|
|
out_error:
|
|
GST_DEBUG_OBJECT (stream, "Done. Error while parsing");
|
|
return GST_HLS_PARSER_RESULT_ERROR;
|
|
|
|
out_need_more:
|
|
GST_DEBUG_OBJECT (stream, "Done. Need more data");
|
|
return GST_HLS_PARSER_RESULT_NEED_MORE_DATA;
|
|
|
|
out_resync:
|
|
GST_DEBUG_OBJECT (stream, "Done. Resync required");
|
|
return GST_HLS_PARSER_RESULT_RESYNC;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_hls_demux_stream_handle_buffer (GstAdaptiveDemux2Stream * stream,
|
|
GstBuffer * buffer, gboolean at_eos)
|
|
{
|
|
GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (stream); // FIXME: pass HlsStream into function
|
|
GstHLSDemux *hlsdemux = GST_HLS_DEMUX_CAST (stream->demux);
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstBuffer *pending_header_data = NULL;
|
|
|
|
/* If current segment is not present, this means that a playlist update
|
|
* happened between the moment ::update_fragment_info() was called and the
|
|
* moment we received data. And that playlist update couldn't match the
|
|
* current position. This will happen in live playback when we are downloading
|
|
* too slowly, therefore we try to "catch up" back to live
|
|
*/
|
|
if (hls_stream->current_segment == NULL) {
|
|
GST_WARNING_OBJECT (stream, "Lost sync");
|
|
/* Drop the buffer */
|
|
gst_buffer_unref (buffer);
|
|
return GST_ADAPTIVE_DEMUX_FLOW_LOST_SYNC;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (stream,
|
|
"buffer:%p at_eos:%d do_typefind:%d uri:%s", buffer, at_eos,
|
|
hls_stream->do_typefind, GST_STR_NULL (stream->fragment.uri));
|
|
|
|
if (buffer == NULL)
|
|
goto out;
|
|
|
|
/* If we need to do typefind and we're not done with it (or we errored), return */
|
|
if (G_UNLIKELY (hls_stream->do_typefind) &&
|
|
!gst_hls_demux_typefind_stream (hlsdemux, stream, &buffer, at_eos,
|
|
&ret)) {
|
|
goto out;
|
|
}
|
|
g_assert (hls_stream->pending_typefind_buffer == NULL);
|
|
|
|
if (hls_stream->process_buffer_content) {
|
|
GstHLSParserResult parse_ret;
|
|
|
|
if (hls_stream->pending_segment_data) {
|
|
if (hls_stream->pending_data_is_header) {
|
|
/* Keep a copy of the header data in case we need to requeue it
|
|
* due to GST_ADAPTIVE_DEMUX_FLOW_RESTART_FRAGMENT below */
|
|
pending_header_data = gst_buffer_ref (hls_stream->pending_segment_data);
|
|
}
|
|
buffer = gst_buffer_append (hls_stream->pending_segment_data, buffer);
|
|
hls_stream->pending_segment_data = NULL;
|
|
}
|
|
|
|
/* Try to get the timing information */
|
|
parse_ret =
|
|
gst_hls_demux_handle_buffer_content (hlsdemux, hls_stream, at_eos,
|
|
&buffer);
|
|
|
|
switch (parse_ret) {
|
|
case GST_HLS_PARSER_RESULT_NEED_MORE_DATA:
|
|
/* If we don't have enough, store and return */
|
|
hls_stream->pending_segment_data = buffer;
|
|
hls_stream->pending_data_is_header =
|
|
(stream->downloading_header == TRUE);
|
|
if (hls_stream->pending_data_is_header)
|
|
stream->send_segment = TRUE;
|
|
goto out;
|
|
case GST_HLS_PARSER_RESULT_ERROR:
|
|
/* Error, drop buffer and return */
|
|
gst_buffer_unref (buffer);
|
|
ret = GST_FLOW_ERROR;
|
|
goto out;
|
|
case GST_HLS_PARSER_RESULT_RESYNC:
|
|
/* Resync, drop buffer and return */
|
|
gst_buffer_unref (buffer);
|
|
ret = GST_ADAPTIVE_DEMUX_FLOW_RESTART_FRAGMENT;
|
|
/* If we had a pending set of header data, requeue it */
|
|
if (pending_header_data != NULL) {
|
|
g_assert (hls_stream->pending_segment_data == NULL);
|
|
|
|
GST_DEBUG_OBJECT (hls_stream,
|
|
"Requeueing header data %" GST_PTR_FORMAT
|
|
" before returning RESTART_FRAGMENT", pending_header_data);
|
|
hls_stream->pending_segment_data = pending_header_data;
|
|
pending_header_data = NULL;
|
|
}
|
|
goto out;
|
|
case GST_HLS_PARSER_RESULT_DONE:
|
|
/* Done parsing, carry on */
|
|
hls_stream->process_buffer_content = FALSE;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (!buffer)
|
|
goto out;
|
|
|
|
buffer = gst_buffer_make_writable (buffer);
|
|
|
|
GST_BUFFER_OFFSET (buffer) = hls_stream->current_offset;
|
|
hls_stream->current_offset += gst_buffer_get_size (buffer);
|
|
GST_BUFFER_OFFSET_END (buffer) = hls_stream->current_offset;
|
|
|
|
GST_DEBUG_OBJECT (stream, "We have a buffer, pushing: %" GST_PTR_FORMAT,
|
|
buffer);
|
|
|
|
ret = gst_adaptive_demux2_stream_push_buffer (stream, buffer);
|
|
|
|
out:
|
|
if (pending_header_data != NULL) {
|
|
/* Throw away the pending header data now. If it wasn't consumed above,
|
|
* we won't need it */
|
|
gst_buffer_unref (pending_header_data);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (stream, "Returning %s", gst_flow_get_name (ret));
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_hls_demux_stream_finish_fragment (GstAdaptiveDemux2Stream * stream)
|
|
{
|
|
GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (stream); // FIXME: pass HlsStream into function
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
|
|
GST_DEBUG_OBJECT (stream, "Finishing %ssegment uri:%s",
|
|
hls_stream->in_partial_segments ? "partial " : "",
|
|
GST_STR_NULL (stream->fragment.uri));
|
|
|
|
/* Drain all pending data */
|
|
if (hls_stream->current_key)
|
|
gst_hls_demux_stream_decrypt_end (hls_stream);
|
|
|
|
if (hls_stream->current_segment && stream->last_ret == GST_FLOW_OK) {
|
|
if (hls_stream->pending_decrypted_buffer) {
|
|
if (hls_stream->current_key) {
|
|
GstMapInfo info;
|
|
gssize unpadded_size;
|
|
|
|
/* Handle pkcs7 unpadding here */
|
|
gst_buffer_map (hls_stream->pending_decrypted_buffer, &info,
|
|
GST_MAP_READ);
|
|
unpadded_size = info.size - info.data[info.size - 1];
|
|
gst_buffer_unmap (hls_stream->pending_decrypted_buffer, &info);
|
|
|
|
gst_buffer_resize (hls_stream->pending_decrypted_buffer, 0,
|
|
unpadded_size);
|
|
}
|
|
|
|
ret =
|
|
gst_hls_demux_stream_handle_buffer (stream,
|
|
hls_stream->pending_decrypted_buffer, TRUE);
|
|
hls_stream->pending_decrypted_buffer = NULL;
|
|
}
|
|
|
|
if (ret == GST_FLOW_OK || ret == GST_FLOW_NOT_LINKED) {
|
|
if (G_UNLIKELY (hls_stream->pending_typefind_buffer)) {
|
|
GstBuffer *buf = hls_stream->pending_typefind_buffer;
|
|
hls_stream->pending_typefind_buffer = NULL;
|
|
|
|
gst_hls_demux_stream_handle_buffer (stream, buf, TRUE);
|
|
}
|
|
|
|
if (hls_stream->pending_segment_data) {
|
|
GstBuffer *buf = hls_stream->pending_segment_data;
|
|
hls_stream->pending_segment_data = NULL;
|
|
|
|
ret = gst_hls_demux_stream_handle_buffer (stream, buf, TRUE);
|
|
}
|
|
}
|
|
}
|
|
|
|
gst_hls_demux_stream_clear_pending_data (hls_stream, FALSE);
|
|
|
|
if (G_UNLIKELY (stream->downloading_header || stream->downloading_index))
|
|
return GST_FLOW_OK;
|
|
|
|
if (hls_stream->current_segment == NULL) {
|
|
/* We can't advance, we just return OK for now and let the base class
|
|
* trigger a new download (or fail and resync itself) */
|
|
GST_DEBUG_OBJECT (stream, "Can't advance - current_segment is NULL");
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
if (ret == GST_FLOW_OK || ret == GST_FLOW_NOT_LINKED) {
|
|
GstClockTime duration = hls_stream->current_segment->duration;
|
|
|
|
/* We can update the stream current position with a more accurate value
|
|
* before advancing. Note that we don't have any period so we can set the
|
|
* stream_time as-is on the stream current position */
|
|
if (hls_stream->in_partial_segments) {
|
|
GstM3U8MediaSegment *cur_segment = hls_stream->current_segment;
|
|
|
|
/* If the current partial segment is valid, update the stream current position, otherwise
|
|
* leave it alone and fix it up later when we resync */
|
|
if (cur_segment->partial_segments
|
|
&& hls_stream->part_idx < cur_segment->partial_segments->len) {
|
|
GstM3U8PartialSegment *part =
|
|
g_ptr_array_index (cur_segment->partial_segments,
|
|
hls_stream->part_idx);
|
|
stream->current_position = part->stream_time;
|
|
duration = part->duration;
|
|
}
|
|
} else {
|
|
stream->current_position = hls_stream->current_segment->stream_time;
|
|
}
|
|
|
|
return gst_adaptive_demux2_stream_advance_fragment (stream, duration);
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_hls_demux_stream_data_received (GstAdaptiveDemux2Stream * stream,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (stream);
|
|
GstHLSDemux *hlsdemux = GST_HLS_DEMUX_CAST (stream->demux);
|
|
GstM3U8MediaSegment *file = hls_stream->current_segment;
|
|
|
|
if (hls_stream->current_segment == NULL)
|
|
return GST_ADAPTIVE_DEMUX_FLOW_LOST_SYNC;
|
|
|
|
if (hls_stream->current_offset == -1)
|
|
hls_stream->current_offset = 0;
|
|
|
|
/* Is it encrypted? */
|
|
if (hls_stream->current_key) {
|
|
GError *err = NULL;
|
|
gsize size;
|
|
GstBuffer *decrypted_buffer;
|
|
GstBuffer *tmp_buffer;
|
|
|
|
if (hls_stream->pending_encrypted_data == NULL)
|
|
hls_stream->pending_encrypted_data = gst_adapter_new ();
|
|
|
|
gst_adapter_push (hls_stream->pending_encrypted_data, buffer);
|
|
size = gst_adapter_available (hls_stream->pending_encrypted_data);
|
|
|
|
/* must be a multiple of 16 */
|
|
size &= (~0xF);
|
|
|
|
if (size == 0) {
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
buffer = gst_adapter_take_buffer (hls_stream->pending_encrypted_data, size);
|
|
decrypted_buffer =
|
|
gst_hls_demux_decrypt_fragment (hlsdemux, hls_stream, buffer, &err);
|
|
if (err) {
|
|
GST_ELEMENT_ERROR (hlsdemux, STREAM, DECODE, ("Failed to decrypt buffer"),
|
|
("decryption failed %s", err->message));
|
|
g_error_free (err);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
|
|
tmp_buffer = hls_stream->pending_decrypted_buffer;
|
|
hls_stream->pending_decrypted_buffer = decrypted_buffer;
|
|
buffer = tmp_buffer;
|
|
if (!buffer)
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
if (!hls_stream->pdt_tag_sent && file != NULL && file->datetime != NULL) {
|
|
GstDateTime *pdt_time = gst_date_time_new_from_g_date_time (g_date_time_ref
|
|
(file->datetime));
|
|
gst_adaptive_demux2_stream_set_tags (stream,
|
|
gst_tag_list_new (GST_TAG_DATE_TIME, pdt_time, NULL));
|
|
gst_date_time_unref (pdt_time);
|
|
hls_stream->pdt_tag_sent = TRUE;
|
|
}
|
|
|
|
|
|
return gst_hls_demux_stream_handle_buffer (stream, buffer, FALSE);
|
|
}
|
|
|
|
static void
|
|
gst_hls_demux_stream_finalize (GObject * object)
|
|
{
|
|
GstAdaptiveDemux2Stream *stream = (GstAdaptiveDemux2Stream *) object;
|
|
GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (object);
|
|
GstHLSDemux *hlsdemux = (GstHLSDemux *) stream->demux;
|
|
|
|
if (hls_stream == hlsdemux->main_stream)
|
|
hlsdemux->main_stream = NULL;
|
|
|
|
g_free (hls_stream->lang);
|
|
g_free (hls_stream->name);
|
|
|
|
if (hls_stream->playlist) {
|
|
gst_hls_media_playlist_unref (hls_stream->playlist);
|
|
hls_stream->playlist = NULL;
|
|
}
|
|
|
|
if (hls_stream->init_file) {
|
|
gst_m3u8_init_file_unref (hls_stream->init_file);
|
|
hls_stream->init_file = NULL;
|
|
}
|
|
|
|
if (hls_stream->pending_encrypted_data)
|
|
g_object_unref (hls_stream->pending_encrypted_data);
|
|
|
|
gst_buffer_replace (&hls_stream->pending_decrypted_buffer, NULL);
|
|
gst_buffer_replace (&hls_stream->pending_typefind_buffer, NULL);
|
|
gst_buffer_replace (&hls_stream->pending_segment_data, NULL);
|
|
|
|
if (hls_stream->playlistloader) {
|
|
gst_hls_demux_playlist_loader_stop (hls_stream->playlistloader);
|
|
gst_object_unparent (GST_OBJECT (hls_stream->playlistloader));
|
|
gst_object_unref (hls_stream->playlistloader);
|
|
}
|
|
|
|
if (hls_stream->preloader) {
|
|
gst_hls_demux_preloader_free (hls_stream->preloader);
|
|
hls_stream->preloader = NULL;
|
|
}
|
|
|
|
if (hls_stream->moov)
|
|
gst_isoff_moov_box_free (hls_stream->moov);
|
|
|
|
if (hls_stream->current_key) {
|
|
g_free (hls_stream->current_key);
|
|
hls_stream->current_key = NULL;
|
|
}
|
|
if (hls_stream->current_iv) {
|
|
g_free (hls_stream->current_iv);
|
|
hls_stream->current_iv = NULL;
|
|
}
|
|
if (hls_stream->current_rendition) {
|
|
gst_hls_rendition_stream_unref (hls_stream->current_rendition);
|
|
hls_stream->current_rendition = NULL;
|
|
}
|
|
if (hls_stream->pending_rendition) {
|
|
gst_hls_rendition_stream_unref (hls_stream->pending_rendition);
|
|
hls_stream->pending_rendition = NULL;
|
|
}
|
|
|
|
if (hls_stream->current_segment) {
|
|
gst_m3u8_media_segment_unref (hls_stream->current_segment);
|
|
hls_stream->current_segment = NULL;
|
|
}
|
|
gst_hls_demux_stream_decrypt_end (hls_stream);
|
|
|
|
G_OBJECT_CLASS (stream_parent_class)->finalize (object);
|
|
}
|
|
|
|
static gboolean
|
|
gst_hls_demux_stream_has_next_fragment (GstAdaptiveDemux2Stream * stream)
|
|
{
|
|
GstHLSDemuxStream *hls_stream = (GstHLSDemuxStream *) stream;
|
|
|
|
GST_DEBUG_OBJECT (stream, "has next ?");
|
|
|
|
if (hls_stream->current_segment == NULL)
|
|
return FALSE;
|
|
|
|
return gst_hls_media_playlist_has_next_fragment (hls_stream->playlist,
|
|
hls_stream->current_segment, stream->demux->segment.rate > 0);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_hls_demux_stream_advance_fragment (GstAdaptiveDemux2Stream * stream)
|
|
{
|
|
GstHLSDemuxStream *hlsdemux_stream = GST_HLS_DEMUX_STREAM_CAST (stream);
|
|
GstHLSDemux *hlsdemux = (GstHLSDemux *) stream->demux;
|
|
GstM3U8MediaSegment *new_segment = NULL;
|
|
|
|
/* If we're playing partial segments, we need to continue
|
|
* doing that. We can only swap back to a full segment on a
|
|
* segment boundary */
|
|
if (hlsdemux_stream->in_partial_segments) {
|
|
/* Check if there's another partial segment in this fragment */
|
|
GstM3U8MediaSegment *cur_segment = hlsdemux_stream->current_segment;
|
|
guint avail_segments =
|
|
cur_segment->partial_segments !=
|
|
NULL ? cur_segment->partial_segments->len : 0;
|
|
|
|
if (hlsdemux_stream->part_idx + 1 < avail_segments) {
|
|
/* Advance to the next partial segment */
|
|
hlsdemux_stream->part_idx += 1;
|
|
|
|
GstM3U8PartialSegment *part =
|
|
g_ptr_array_index (cur_segment->partial_segments,
|
|
hlsdemux_stream->part_idx);
|
|
|
|
GST_DEBUG_OBJECT (stream,
|
|
"Advanced to partial segment sn:%" G_GINT64_FORMAT
|
|
" part %d stream_time:%" GST_STIME_FORMAT " uri:%s",
|
|
hlsdemux_stream->current_segment->sequence, hlsdemux_stream->part_idx,
|
|
GST_STIME_ARGS (part->stream_time), GST_STR_NULL (part->uri));
|
|
|
|
return GST_FLOW_OK;
|
|
} else if (cur_segment->partial_only) {
|
|
/* There's no partial segment available, because we're at the live edge */
|
|
GST_DEBUG_OBJECT (stream,
|
|
"Hit live edge playing partial segments. Will wait for playlist update.");
|
|
hlsdemux_stream->part_idx += 1;
|
|
return GST_FLOW_OK;
|
|
} else {
|
|
/* At the end of the partial segments for this full segment. Advance to the next full segment */
|
|
hlsdemux_stream->in_partial_segments = FALSE;
|
|
GST_DEBUG_OBJECT (stream,
|
|
"No more partial segments in current segment. Advancing");
|
|
}
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (stream,
|
|
"Current segment sn:%" G_GINT64_FORMAT " stream_time:%" GST_STIME_FORMAT
|
|
" uri:%s", hlsdemux_stream->current_segment->sequence,
|
|
GST_STIME_ARGS (hlsdemux_stream->current_segment->stream_time),
|
|
GST_STR_NULL (hlsdemux_stream->current_segment->uri));
|
|
|
|
new_segment =
|
|
gst_hls_media_playlist_advance_fragment (hlsdemux_stream->playlist,
|
|
hlsdemux_stream->current_segment, stream->demux->segment.rate > 0);
|
|
|
|
if (new_segment) {
|
|
hlsdemux_stream->reset_pts = FALSE;
|
|
if (new_segment->discont_sequence !=
|
|
hlsdemux_stream->current_segment->discont_sequence)
|
|
gst_hls_demux_add_time_mapping (hlsdemux, new_segment->discont_sequence,
|
|
new_segment->stream_time, new_segment->datetime);
|
|
gst_m3u8_media_segment_unref (hlsdemux_stream->current_segment);
|
|
hlsdemux_stream->current_segment = new_segment;
|
|
|
|
/* In LL-HLS, handle advancing into the partial-only segment */
|
|
if (GST_HLS_MEDIA_PLAYLIST_IS_LIVE (hlsdemux_stream->playlist)
|
|
&& new_segment->partial_only) {
|
|
hlsdemux_stream->in_partial_segments = TRUE;
|
|
hlsdemux_stream->part_idx = 0;
|
|
|
|
GstM3U8PartialSegment *new_part =
|
|
g_ptr_array_index (new_segment->partial_segments,
|
|
hlsdemux_stream->part_idx);
|
|
|
|
GST_DEBUG_OBJECT (stream,
|
|
"Advanced to partial segment sn:%" G_GINT64_FORMAT
|
|
" part %u stream_time:%" GST_STIME_FORMAT " uri:%s",
|
|
new_segment->sequence, hlsdemux_stream->part_idx,
|
|
GST_STIME_ARGS (new_part->stream_time), GST_STR_NULL (new_part->uri));
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (stream,
|
|
"Advanced to segment sn:%" G_GINT64_FORMAT " stream_time:%"
|
|
GST_STIME_FORMAT " uri:%s", hlsdemux_stream->current_segment->sequence,
|
|
GST_STIME_ARGS (hlsdemux_stream->current_segment->stream_time),
|
|
GST_STR_NULL (hlsdemux_stream->current_segment->uri));
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
GST_LOG_OBJECT (stream, "Could not advance to next fragment");
|
|
if (GST_HLS_MEDIA_PLAYLIST_IS_LIVE (hlsdemux_stream->playlist)) {
|
|
gst_m3u8_media_segment_unref (hlsdemux_stream->current_segment);
|
|
hlsdemux_stream->current_segment = NULL;
|
|
hlsdemux_stream->in_partial_segments = FALSE;
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
return GST_FLOW_EOS;
|
|
}
|
|
|
|
static void
|
|
gst_hls_demux_stream_update_preloads (GstHLSDemuxStream * hlsdemux_stream)
|
|
{
|
|
GstHLSMediaPlaylist *playlist = hlsdemux_stream->playlist;
|
|
gboolean preloads_allowed = GST_HLS_MEDIA_PLAYLIST_IS_LIVE (playlist);
|
|
|
|
if (playlist->preload_hints == NULL || !preloads_allowed) {
|
|
if (hlsdemux_stream->preloader != NULL) {
|
|
/* Cancel any preloads, the new playlist doesn't have them */
|
|
gst_hls_demux_preloader_cancel (hlsdemux_stream->preloader,
|
|
M3U8_PRELOAD_HINT_ALL);
|
|
}
|
|
/* Nothing to preload */
|
|
return;
|
|
}
|
|
|
|
if (hlsdemux_stream->preloader == NULL) {
|
|
GstAdaptiveDemux *demux =
|
|
GST_ADAPTIVE_DEMUX2_STREAM (hlsdemux_stream)->demux;
|
|
hlsdemux_stream->preloader =
|
|
gst_hls_demux_preloader_new (demux->download_helper);
|
|
if (hlsdemux_stream->preloader == NULL) {
|
|
GST_WARNING_OBJECT (hlsdemux_stream, "Failed to create preload handler");
|
|
return;
|
|
}
|
|
}
|
|
|
|
/* The HLS spec says any extra preload hint of each type should be ignored */
|
|
GstM3U8PreloadHintType seen_types = 0;
|
|
guint idx;
|
|
for (idx = 0; idx < playlist->preload_hints->len; idx++) {
|
|
GstM3U8PreloadHint *hint = g_ptr_array_index (playlist->preload_hints, idx);
|
|
switch (hint->hint_type) {
|
|
case M3U8_PRELOAD_HINT_MAP:
|
|
case M3U8_PRELOAD_HINT_PART:
|
|
if (seen_types & hint->hint_type) {
|
|
continue; /* Ignore preload hint type we've already seen */
|
|
}
|
|
seen_types |= hint->hint_type;
|
|
break;
|
|
default:
|
|
GST_FIXME_OBJECT (hlsdemux_stream, "Ignoring unknown preload type %d",
|
|
hint->hint_type);
|
|
continue; /* Unknown hint type, ignore it */
|
|
}
|
|
gst_hls_demux_preloader_load (hlsdemux_stream->preloader, hint,
|
|
playlist->uri);
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_hls_demux_stream_submit_request (GstAdaptiveDemux2Stream * stream,
|
|
DownloadRequest * download_req)
|
|
{
|
|
GstHLSDemuxStream *hlsdemux_stream = GST_HLS_DEMUX_STREAM_CAST (stream);
|
|
|
|
/* See if the request can be satisfied from a preload */
|
|
if (hlsdemux_stream->preloader != NULL) {
|
|
if (gst_hls_demux_preloader_provide_request (hlsdemux_stream->preloader,
|
|
download_req))
|
|
return GST_FLOW_OK;
|
|
|
|
/* We're about to request something, but it wasn't the active preload,
|
|
* so make sure that's been stopped / cancelled so we're not downloading
|
|
* two things in parallel. This usually means the playlist refresh
|
|
* took too long and the preload became obsolete */
|
|
if (stream->downloading_header) {
|
|
gst_hls_demux_preloader_cancel (hlsdemux_stream->preloader,
|
|
M3U8_PRELOAD_HINT_MAP);
|
|
} else {
|
|
gst_hls_demux_preloader_cancel (hlsdemux_stream->preloader,
|
|
M3U8_PRELOAD_HINT_PART);
|
|
}
|
|
}
|
|
|
|
return
|
|
GST_ADAPTIVE_DEMUX2_STREAM_CLASS (stream_parent_class)->submit_request
|
|
(stream, download_req);
|
|
}
|
|
|
|
static void
|
|
gst_hls_demux_stream_handle_playlist_update (GstHLSDemuxStream * stream,
|
|
const gchar * new_playlist_uri, GstHLSMediaPlaylist * new_playlist)
|
|
{
|
|
GstHLSDemux *demux = GST_HLS_DEMUX_STREAM_GET_DEMUX (stream);
|
|
|
|
/* Synchronize playlist with previous one. If we can't update the playlist
|
|
* timing and inform the base class that we lost sync */
|
|
if (stream->playlist
|
|
&& !gst_hls_media_playlist_sync_to_playlist (new_playlist,
|
|
stream->playlist)) {
|
|
/* Failure to synchronize with the previous media playlist is only fatal for
|
|
* variant streams. */
|
|
if (stream->is_variant) {
|
|
GST_DEBUG_OBJECT (stream,
|
|
"Could not synchronize new variant playlist with previous one !");
|
|
goto lost_sync;
|
|
}
|
|
|
|
/* For rendition streams, we can attempt synchronization against the
|
|
* variant playlist which is constantly updated */
|
|
if (demux->main_stream->playlist
|
|
&& !gst_hls_media_playlist_sync_to_playlist (new_playlist,
|
|
demux->main_stream->playlist)) {
|
|
GST_DEBUG_OBJECT (stream,
|
|
"Could not do fallback synchronization of rendition stream to variant stream");
|
|
goto lost_sync;
|
|
}
|
|
} else if (!stream->is_variant && demux->main_stream->playlist) {
|
|
/* For initial rendition media playlist, attempt to synchronize the playlist
|
|
* against the variant stream. This is non-fatal if it fails. */
|
|
GST_DEBUG_OBJECT (stream,
|
|
"Attempting to synchronize initial rendition stream with variant stream");
|
|
gst_hls_media_playlist_sync_to_playlist (new_playlist,
|
|
demux->main_stream->playlist);
|
|
}
|
|
|
|
if (stream->current_segment) {
|
|
GstM3U8MediaSegment *new_segment;
|
|
GST_DEBUG_OBJECT (stream,
|
|
"Current segment sn:%" G_GINT64_FORMAT " stream_time:%" GST_STIME_FORMAT
|
|
" uri:%s", stream->current_segment->sequence,
|
|
GST_STIME_ARGS (stream->current_segment->stream_time),
|
|
GST_STR_NULL (stream->current_segment->uri));
|
|
|
|
/* Use best-effort techniques to find the corresponding current media segment
|
|
* in the new playlist. This might be off in some cases, but it doesn't matter
|
|
* since we will be checking the embedded timestamp later */
|
|
new_segment =
|
|
gst_hls_media_playlist_sync_to_segment (new_playlist,
|
|
stream->current_segment);
|
|
|
|
/* Handle LL-HLS partial segment sync by checking our partial segment
|
|
* still makes sense */
|
|
if (stream->in_partial_segments && new_segment) {
|
|
/* We must be either playing the trailing open-ended partial segment,
|
|
* or if we're playing partials from a complete segment, check that we
|
|
* still have a) partial segments attached (didn't get too old and
|
|
* the server removed them from the playlist) and b) we didn't advance
|
|
* beyond the end of that partial segment (when we advance past the live
|
|
* edge and increment part_idx, then the segment completes without
|
|
* adding any more partial segments) */
|
|
if (!new_segment->partial_only) {
|
|
if (new_segment->partial_segments == NULL) {
|
|
GST_DEBUG_OBJECT (stream,
|
|
"Partial segments we were playing became unavailable. Will try and resync");
|
|
stream->in_partial_segments = FALSE;
|
|
gst_m3u8_media_segment_unref (new_segment);
|
|
new_segment = NULL;
|
|
} else if (stream->part_idx >= new_segment->partial_segments->len) {
|
|
GST_DEBUG_OBJECT (stream,
|
|
"After playlist reload, there are no more partial segments to play in the current segment. Resyncing");
|
|
stream->in_partial_segments = FALSE;
|
|
gst_m3u8_media_segment_unref (new_segment);
|
|
new_segment = NULL;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (new_segment) {
|
|
if (new_segment->discont_sequence !=
|
|
stream->current_segment->discont_sequence)
|
|
gst_hls_demux_add_time_mapping (demux, new_segment->discont_sequence,
|
|
new_segment->stream_time, new_segment->datetime);
|
|
/* This can happen in case of misaligned variants/renditions. Only warn about it */
|
|
if (new_segment->stream_time != stream->current_segment->stream_time)
|
|
GST_WARNING_OBJECT (stream,
|
|
"Returned segment stream time %" GST_STIME_FORMAT
|
|
" differs from current stream time %" GST_STIME_FORMAT,
|
|
GST_STIME_ARGS (new_segment->stream_time),
|
|
GST_STIME_ARGS (stream->current_segment->stream_time));
|
|
} else {
|
|
/* Not finding a matching segment only happens in live (otherwise we would
|
|
* have found a match by stream time) when we are at the live edge. This is normal*/
|
|
GST_DEBUG_OBJECT (stream, "Could not find a matching segment");
|
|
}
|
|
gst_m3u8_media_segment_unref (stream->current_segment);
|
|
stream->current_segment = new_segment;
|
|
} else {
|
|
GST_DEBUG_OBJECT (stream, "No current segment");
|
|
}
|
|
|
|
if (stream->is_variant) {
|
|
/* Updates on the variant playlist have some special requirements to
|
|
* set up the time mapping and initial stream config */
|
|
gst_hls_demux_handle_variant_playlist_update (demux, new_playlist_uri,
|
|
new_playlist);
|
|
} else if (stream->pending_rendition) {
|
|
/* Switching rendition configures a new playlist on the loader,
|
|
* and we should never get a callback for a stale download URI */
|
|
g_assert (!g_strcmp0 (stream->pending_rendition->uri, new_playlist_uri));
|
|
|
|
gst_hls_rendition_stream_unref (stream->current_rendition);
|
|
/* Stealing ref */
|
|
stream->current_rendition = stream->pending_rendition;
|
|
stream->pending_rendition = NULL;
|
|
}
|
|
|
|
if (stream->playlist)
|
|
gst_hls_media_playlist_unref (stream->playlist);
|
|
stream->playlist = gst_hls_media_playlist_ref (new_playlist);
|
|
stream->playlist_fetched = TRUE;
|
|
|
|
if (!GST_HLS_MEDIA_PLAYLIST_IS_LIVE (stream->playlist)) {
|
|
/* Make sure to cancel any preloads if a playlist isn't live after reload */
|
|
gst_hls_demux_stream_update_preloads (stream);
|
|
}
|
|
|
|
if (stream->current_segment) {
|
|
GST_DEBUG_OBJECT (stream,
|
|
"After update, current segment now sn:%" G_GINT64_FORMAT
|
|
" stream_time:%" GST_STIME_FORMAT " uri:%s",
|
|
stream->current_segment->sequence,
|
|
GST_STIME_ARGS (stream->current_segment->stream_time),
|
|
GST_STR_NULL (stream->current_segment->uri));
|
|
} else {
|
|
GST_DEBUG_OBJECT (stream, "No current segment selected");
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (stream, "done");
|
|
return;
|
|
|
|
/* ERRORS */
|
|
lost_sync:
|
|
{
|
|
/* Set new playlist, lost sync handler will know what to do with it */
|
|
if (stream->playlist)
|
|
gst_hls_media_playlist_unref (stream->playlist);
|
|
stream->playlist = new_playlist;
|
|
stream->playlist = gst_hls_media_playlist_ref (new_playlist);
|
|
stream->playlist_fetched = TRUE;
|
|
|
|
gst_hls_demux_reset_for_lost_sync (demux);
|
|
}
|
|
}
|
|
|
|
static void
|
|
on_playlist_update_success (GstHLSDemuxPlaylistLoader * pl,
|
|
const gchar * new_playlist_uri, GstHLSMediaPlaylist * new_playlist,
|
|
gpointer userdata)
|
|
{
|
|
GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (userdata);
|
|
|
|
gst_hls_demux_stream_handle_playlist_update (hls_stream,
|
|
new_playlist_uri, new_playlist);
|
|
gst_adaptive_demux2_stream_mark_prepared (GST_ADAPTIVE_DEMUX2_STREAM_CAST
|
|
(hls_stream));
|
|
}
|
|
|
|
static void
|
|
on_playlist_update_error (GstHLSDemuxPlaylistLoader * pl,
|
|
const gchar * playlist_uri, gpointer userdata)
|
|
{
|
|
GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (userdata);
|
|
|
|
/* FIXME: How to handle rendition playlist update errors? There's
|
|
* not much we can do about it except throw an error */
|
|
if (hls_stream->is_variant) {
|
|
GstHLSDemux *demux = GST_HLS_DEMUX_STREAM_GET_DEMUX (hls_stream);
|
|
gst_hls_demux_handle_variant_playlist_update_error (demux, playlist_uri);
|
|
} else {
|
|
GstHLSDemux *demux = GST_HLS_DEMUX_STREAM_GET_DEMUX (hls_stream);
|
|
GST_ELEMENT_ERROR (demux, STREAM, FAILED,
|
|
(_("Internal data stream error.")),
|
|
("Could not update rendition playlist"));
|
|
}
|
|
}
|
|
|
|
static GstHLSDemuxPlaylistLoader *
|
|
gst_hls_demux_stream_get_playlist_loader (GstHLSDemuxStream * hls_stream)
|
|
{
|
|
GstAdaptiveDemux *demux = GST_ADAPTIVE_DEMUX2_STREAM_CAST (hls_stream)->demux;
|
|
if (hls_stream->playlistloader == NULL) {
|
|
hls_stream->playlistloader =
|
|
gst_hls_demux_playlist_loader_new (demux, demux->download_helper);
|
|
gst_hls_demux_playlist_loader_set_callbacks (hls_stream->playlistloader,
|
|
on_playlist_update_success, on_playlist_update_error, hls_stream);
|
|
}
|
|
|
|
return hls_stream->playlistloader;
|
|
}
|
|
|
|
void
|
|
gst_hls_demux_stream_set_playlist_uri (GstHLSDemuxStream * hls_stream,
|
|
gchar * uri)
|
|
{
|
|
GstAdaptiveDemux *demux = GST_ADAPTIVE_DEMUX2_STREAM_CAST (hls_stream)->demux;
|
|
GstHLSDemuxPlaylistLoader *pl =
|
|
gst_hls_demux_stream_get_playlist_loader (hls_stream);
|
|
|
|
const gchar *main_uri = gst_adaptive_demux_get_manifest_ref_uri (demux);
|
|
gst_hls_demux_playlist_loader_set_playlist_uri (pl, main_uri, uri);
|
|
}
|
|
|
|
void
|
|
gst_hls_demux_stream_start_playlist_loading (GstHLSDemuxStream * hls_stream)
|
|
{
|
|
GstHLSDemuxPlaylistLoader *pl =
|
|
gst_hls_demux_stream_get_playlist_loader (hls_stream);
|
|
gst_hls_demux_playlist_loader_start (pl);
|
|
}
|
|
|
|
GstFlowReturn
|
|
gst_hls_demux_stream_check_current_playlist_uri (GstHLSDemuxStream * stream,
|
|
gchar * uri)
|
|
{
|
|
GstHLSDemuxPlaylistLoader *pl =
|
|
gst_hls_demux_stream_get_playlist_loader (stream);
|
|
|
|
if (!gst_hls_demux_playlist_loader_has_current_uri (pl, uri)) {
|
|
GST_LOG_OBJECT (stream, "Target playlist not available yet");
|
|
return GST_ADAPTIVE_DEMUX_FLOW_BUSY;
|
|
}
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
#if 0
|
|
/* Check if a redirect happened */
|
|
if (g_strcmp0 (*uri, new_playlist->uri)) {
|
|
GST_DEBUG_OBJECT (stream, "Playlist URI update : '%s' => '%s'", *uri,
|
|
new_playlist->uri);
|
|
g_free (*uri);
|
|
*uri = g_strdup (new_playlist->uri);
|
|
}
|
|
#endif
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_hls_demux_stream_update_fragment_info (GstAdaptiveDemux2Stream * stream)
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstHLSDemuxStream *hlsdemux_stream = GST_HLS_DEMUX_STREAM_CAST (stream);
|
|
GstAdaptiveDemux *demux = stream->demux;
|
|
GstHLSDemux *hlsdemux = GST_HLS_DEMUX_CAST (demux);
|
|
GstM3U8MediaSegment *file;
|
|
GstM3U8PartialSegment *part = NULL;
|
|
gboolean discont;
|
|
|
|
/* Return BUSY if no playlist is loaded yet. Even if
|
|
* we switched an another playlist is loading, we'll keep*/
|
|
if (!hlsdemux_stream->playlist_fetched) {
|
|
gst_hls_demux_stream_start_playlist_loading (hlsdemux_stream);
|
|
return GST_ADAPTIVE_DEMUX_FLOW_BUSY;
|
|
}
|
|
g_assert (hlsdemux_stream->playlist != NULL);
|
|
if ((ret =
|
|
gst_hls_demux_stream_check_current_playlist_uri (hlsdemux_stream,
|
|
NULL)) != GST_FLOW_OK) {
|
|
/* The URI of the playlist we have is not the target URI due
|
|
* to a bitrate switch - wait for it to load */
|
|
GST_DEBUG_OBJECT (hlsdemux_stream,
|
|
"Playlist is stale. Waiting for new playlist");
|
|
gst_hls_demux_stream_start_playlist_loading (hlsdemux_stream);
|
|
return ret;
|
|
}
|
|
#ifndef GST_DISABLE_GST_DEBUG
|
|
GstClockTimeDiff live_edge_dist =
|
|
GST_CLOCK_TIME_IS_VALID (stream->current_position) ?
|
|
gst_hls_media_playlist_get_end_stream_time (hlsdemux_stream->playlist) -
|
|
stream->current_position : GST_CLOCK_TIME_NONE;
|
|
GstClockTime playlist_age =
|
|
gst_adaptive_demux2_get_monotonic_time (GST_ADAPTIVE_DEMUX (demux)) -
|
|
hlsdemux_stream->playlist->playlist_ts;
|
|
GST_DEBUG_OBJECT (stream,
|
|
"Updating fragment information, current_position:%" GST_TIME_FORMAT
|
|
" which is %" GST_STIME_FORMAT " from live edge. Playlist age %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (stream->current_position),
|
|
GST_STIME_ARGS (live_edge_dist), GST_TIME_ARGS (playlist_age));
|
|
#endif
|
|
|
|
/* Find the current segment if we don't already have it */
|
|
if (hlsdemux_stream->current_segment == NULL) {
|
|
GST_LOG_OBJECT (stream, "No current segment");
|
|
if (stream->current_position == GST_CLOCK_TIME_NONE) {
|
|
GstM3U8SeekResult seek_result;
|
|
|
|
GST_DEBUG_OBJECT (stream, "Setting up initial segment");
|
|
|
|
if (gst_hls_media_playlist_get_starting_segment
|
|
(hlsdemux_stream->playlist, &seek_result)) {
|
|
hlsdemux_stream->current_segment = seek_result.segment;
|
|
hlsdemux_stream->in_partial_segments =
|
|
seek_result.found_partial_segment;
|
|
hlsdemux_stream->part_idx = seek_result.part_idx;
|
|
}
|
|
} else {
|
|
if (gst_hls_media_playlist_has_lost_sync (hlsdemux_stream->playlist,
|
|
stream->current_position)) {
|
|
GST_WARNING_OBJECT (stream, "Lost SYNC !");
|
|
return GST_ADAPTIVE_DEMUX_FLOW_LOST_SYNC;
|
|
}
|
|
GST_DEBUG_OBJECT (stream,
|
|
"Looking up segment for position %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (stream->current_position));
|
|
|
|
GstM3U8SeekResult seek_result;
|
|
if (!gst_hls_media_playlist_find_position (hlsdemux_stream->playlist,
|
|
stream->current_position, hlsdemux_stream->in_partial_segments,
|
|
&seek_result)) {
|
|
GST_INFO_OBJECT (stream, "At the end of the current media playlist");
|
|
gst_hls_demux_stream_update_preloads (hlsdemux_stream);
|
|
return GST_FLOW_EOS;
|
|
}
|
|
|
|
hlsdemux_stream->current_segment = seek_result.segment;
|
|
hlsdemux_stream->in_partial_segments = seek_result.found_partial_segment;
|
|
hlsdemux_stream->part_idx = seek_result.part_idx;
|
|
|
|
/* If on a full segment, update time mapping. If it already exists it will be ignored.
|
|
* Don't add time mappings for partial segments, wait for a full segment boundary */
|
|
if (!hlsdemux_stream->in_partial_segments
|
|
|| hlsdemux_stream->part_idx == 0) {
|
|
gst_hls_demux_add_time_mapping (hlsdemux,
|
|
hlsdemux_stream->current_segment->discont_sequence,
|
|
hlsdemux_stream->current_segment->stream_time,
|
|
hlsdemux_stream->current_segment->datetime);
|
|
}
|
|
}
|
|
}
|
|
|
|
file = hlsdemux_stream->current_segment;
|
|
|
|
if (hlsdemux_stream->in_partial_segments) {
|
|
if (file->partial_segments == NULL) {
|
|
/* I think this can only happen if we reloaded the playlist
|
|
* and the segment we were in the middle of playing from
|
|
* removed its partial segments because we were playing
|
|
* too slowly */
|
|
GST_DEBUG_OBJECT (stream,
|
|
"Partial segment idx %d is not available in current playlist",
|
|
hlsdemux_stream->part_idx);
|
|
return GST_ADAPTIVE_DEMUX_FLOW_LOST_SYNC;
|
|
}
|
|
|
|
if (hlsdemux_stream->part_idx >= file->partial_segments->len) {
|
|
/* Being beyond the available partial segments in the partial_only
|
|
* segment at the end of the playlist in LL-HLS means we've
|
|
* hit the live edge and need to wait for a playlist update */
|
|
if (file->partial_only) {
|
|
GST_INFO_OBJECT (stream, "At the end of the current media playlist");
|
|
gst_hls_demux_stream_update_preloads (hlsdemux_stream);
|
|
return GST_FLOW_EOS;
|
|
}
|
|
|
|
/* Otherwise, we reloaded the playlist and found that the partial_only segment we
|
|
* were playing from became a real segment and we overstepped the end of
|
|
* the parts. Reloading the playlist should have synced that up properly,
|
|
* so we should never get here. */
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
part =
|
|
g_ptr_array_index (file->partial_segments, hlsdemux_stream->part_idx);
|
|
|
|
GST_DEBUG_OBJECT (stream,
|
|
"Current partial segment %d stream_time %" GST_STIME_FORMAT,
|
|
hlsdemux_stream->part_idx, GST_STIME_ARGS (part->stream_time));
|
|
discont = stream->discont;
|
|
/* Use the segment discont flag only on the first partial segment */
|
|
if (file->discont && hlsdemux_stream->part_idx == 0)
|
|
discont = TRUE;
|
|
} else {
|
|
GST_DEBUG_OBJECT (stream, "Current segment stream_time %" GST_STIME_FORMAT,
|
|
GST_STIME_ARGS (file->stream_time));
|
|
discont = file->discont || stream->discont;
|
|
}
|
|
|
|
gboolean need_header = GST_ADAPTIVE_DEMUX2_STREAM_NEED_HEADER (stream);
|
|
|
|
/* Check if the MAP header file changed and update it */
|
|
if (file->init_file != NULL
|
|
&& !gst_m3u8_init_file_equal (hlsdemux_stream->init_file,
|
|
file->init_file)) {
|
|
GST_DEBUG_OBJECT (stream, "MAP header info changed. Updating");
|
|
if (hlsdemux_stream->init_file != NULL)
|
|
gst_m3u8_init_file_unref (hlsdemux_stream->init_file);
|
|
hlsdemux_stream->init_file = gst_m3u8_init_file_ref (file->init_file);
|
|
need_header = TRUE;
|
|
}
|
|
|
|
if (file->init_file && need_header) {
|
|
GstM3U8InitFile *header_file = file->init_file;
|
|
g_free (stream->fragment.header_uri);
|
|
stream->fragment.header_uri = g_strdup (header_file->uri);
|
|
stream->fragment.header_range_start = header_file->offset;
|
|
if (header_file->size != -1) {
|
|
stream->fragment.header_range_end =
|
|
header_file->offset + header_file->size - 1;
|
|
} else {
|
|
stream->fragment.header_range_end = -1;
|
|
}
|
|
|
|
stream->need_header = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (stream, "Need header uri: %s %" G_GUINT64_FORMAT " %"
|
|
G_GINT64_FORMAT, stream->fragment.header_uri,
|
|
stream->fragment.header_range_start, stream->fragment.header_range_end);
|
|
}
|
|
|
|
/* set up our source for download */
|
|
stream->fragment.stream_time = GST_CLOCK_STIME_NONE;
|
|
g_free (stream->fragment.uri);
|
|
stream->fragment.range_start = 0;
|
|
stream->fragment.range_end = -1;
|
|
|
|
/* Encryption params always come from the parent segment */
|
|
g_free (hlsdemux_stream->current_key);
|
|
hlsdemux_stream->current_key = g_strdup (file->key);
|
|
g_free (hlsdemux_stream->current_iv);
|
|
hlsdemux_stream->current_iv = g_memdup2 (file->iv, sizeof (file->iv));
|
|
|
|
/* Other info could come from the part when playing partial segments */
|
|
|
|
if (part == NULL) {
|
|
if (hlsdemux_stream->reset_pts || discont || demux->segment.rate < 0.0) {
|
|
stream->fragment.stream_time = file->stream_time;
|
|
}
|
|
stream->fragment.uri = g_strdup (file->uri);
|
|
stream->fragment.range_start = file->offset;
|
|
if (file->size != -1)
|
|
stream->fragment.range_end = file->offset + file->size - 1;
|
|
stream->fragment.duration = file->duration;
|
|
} else {
|
|
if (hlsdemux_stream->reset_pts || discont || demux->segment.rate < 0.0) {
|
|
stream->fragment.stream_time = part->stream_time;
|
|
}
|
|
stream->fragment.uri = g_strdup (part->uri);
|
|
stream->fragment.range_start = part->offset;
|
|
if (part->size != -1)
|
|
stream->fragment.range_end = part->offset + part->size - 1;
|
|
stream->fragment.duration = part->duration;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (stream, "Stream URI now %s", stream->fragment.uri);
|
|
|
|
stream->recommended_buffering_threshold =
|
|
gst_hls_media_playlist_recommended_buffering_threshold
|
|
(hlsdemux_stream->playlist);
|
|
|
|
if (discont)
|
|
stream->discont = TRUE;
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_hls_demux_stream_can_start (GstAdaptiveDemux2Stream * stream)
|
|
{
|
|
GstHLSDemux *hlsdemux = (GstHLSDemux *) stream->demux;
|
|
GstHLSDemuxStream *hls_stream = (GstHLSDemuxStream *) stream;
|
|
GList *tmp;
|
|
|
|
GST_DEBUG_OBJECT (stream, "is_variant:%d mappings:%p", hls_stream->is_variant,
|
|
hlsdemux->mappings);
|
|
|
|
/* Variant streams can always start straight away */
|
|
if (hls_stream->is_variant)
|
|
return TRUE;
|
|
|
|
/* Renditions of the exact same type as the variant are pure alternatives,
|
|
* they must be started. This can happen for example with audio-only manifests
|
|
* where the initial stream selected is a rendition and not a variant */
|
|
if (hls_stream->rendition_type == hlsdemux->main_stream->rendition_type)
|
|
return TRUE;
|
|
|
|
/* Rendition streams only require delaying if we don't have time mappings yet */
|
|
if (!hlsdemux->mappings)
|
|
return FALSE;
|
|
|
|
/* We can start if we have at least one internal time observation */
|
|
for (tmp = hlsdemux->mappings; tmp; tmp = tmp->next) {
|
|
GstHLSTimeMap *map = tmp->data;
|
|
if (map->internal_time != GST_CLOCK_TIME_NONE)
|
|
return TRUE;
|
|
}
|
|
|
|
/* Otherwise we have to wait */
|
|
return FALSE;
|
|
}
|
|
|
|
static void
|
|
gst_hls_demux_stream_start (GstAdaptiveDemux2Stream * stream)
|
|
{
|
|
if (!gst_hls_demux_stream_can_start (stream))
|
|
return;
|
|
|
|
/* Start the playlist loader */
|
|
GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (stream);
|
|
|
|
gst_hls_demux_stream_start_playlist_loading (hls_stream);
|
|
|
|
/* Chain up, to start the downloading */
|
|
GST_ADAPTIVE_DEMUX2_STREAM_CLASS (stream_parent_class)->start (stream);
|
|
}
|
|
|
|
static void
|
|
gst_hls_demux_stream_stop (GstAdaptiveDemux2Stream * stream)
|
|
{
|
|
GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (stream);
|
|
|
|
if (hls_stream->playlistloader && !hls_stream->is_variant) {
|
|
/* Don't stop the loader for the variant stream, keep it running
|
|
* until the scheduler itself is stopped so we keep updating
|
|
* the live playlist timeline */
|
|
gst_hls_demux_playlist_loader_stop (hls_stream->playlistloader);
|
|
}
|
|
|
|
/* Chain up, to stop the downloading */
|
|
GST_ADAPTIVE_DEMUX2_STREAM_CLASS (stream_parent_class)->stop (stream);
|
|
}
|
|
|
|
/* Called when the variant is changed, to set a new rendition
|
|
* for this stream to download. Returns TRUE if the rendition
|
|
* stream switched group-id */
|
|
static gboolean
|
|
gst_hls_demux_update_rendition_stream_uri (GstHLSDemux * hlsdemux,
|
|
GstHLSDemuxStream * hls_stream, GError ** err)
|
|
{
|
|
gchar *current_group_id, *requested_group_id;
|
|
GstHLSRenditionStream *replacement_media = NULL;
|
|
GList *tmp;
|
|
|
|
/* There always should be a current variant set */
|
|
g_assert (hlsdemux->current_variant);
|
|
/* There always is a GstHLSRenditionStream set for rendition streams */
|
|
g_assert (hls_stream->current_rendition);
|
|
|
|
requested_group_id =
|
|
hlsdemux->current_variant->media_groups[hls_stream->
|
|
current_rendition->mtype];
|
|
current_group_id = hls_stream->current_rendition->group_id;
|
|
|
|
GST_DEBUG_OBJECT (hlsdemux,
|
|
"Checking playlist change for variant stream %s lang: %s current group-id: %s / requested group-id: %s",
|
|
gst_stream_type_get_name (hls_stream->rendition_type), hls_stream->lang,
|
|
current_group_id, requested_group_id);
|
|
|
|
|
|
if (!g_strcmp0 (requested_group_id, current_group_id)) {
|
|
GST_DEBUG_OBJECT (hlsdemux, "No change needed");
|
|
return FALSE;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (hlsdemux,
|
|
"group-id changed, looking for replacement playlist");
|
|
|
|
/* Need to switch/update */
|
|
for (tmp = hlsdemux->master->renditions; tmp; tmp = tmp->next) {
|
|
GstHLSRenditionStream *cand = tmp->data;
|
|
|
|
if (cand->mtype == hls_stream->current_rendition->mtype
|
|
&& !g_strcmp0 (cand->lang, hls_stream->lang)
|
|
&& !g_strcmp0 (cand->group_id, requested_group_id)) {
|
|
replacement_media = cand;
|
|
break;
|
|
}
|
|
}
|
|
if (!replacement_media) {
|
|
GST_ERROR_OBJECT (hlsdemux,
|
|
"Could not find a replacement playlist. Staying with previous one");
|
|
return FALSE;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (hlsdemux, "Use replacement playlist %s",
|
|
replacement_media->name);
|
|
if (hls_stream->pending_rendition) {
|
|
GST_ERROR_OBJECT (hlsdemux,
|
|
"Already had a pending rendition switch to '%s'",
|
|
hls_stream->pending_rendition->name);
|
|
gst_hls_rendition_stream_unref (hls_stream->pending_rendition);
|
|
}
|
|
hls_stream->pending_rendition =
|
|
gst_hls_rendition_stream_ref (replacement_media);
|
|
|
|
gst_hls_demux_stream_set_playlist_uri (hls_stream, replacement_media->uri);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_hls_demux_stream_select_bitrate (GstAdaptiveDemux2Stream * stream,
|
|
guint64 bitrate)
|
|
{
|
|
GstAdaptiveDemux *demux = GST_ADAPTIVE_DEMUX_CAST (stream->demux);
|
|
GstHLSDemux *hlsdemux = GST_HLS_DEMUX_CAST (stream->demux);
|
|
GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (stream);
|
|
|
|
/* Fast-Path, no changes possible */
|
|
if (hlsdemux->master == NULL || hlsdemux->master->is_simple)
|
|
return FALSE;
|
|
|
|
/* Currently playing partial segments, disallow bitrate
|
|
* switches and rendition playlist changes - except exactly
|
|
* at the first partial segment in a full segment (implying
|
|
* we are about to play a partial segment but didn't yet) */
|
|
if (hls_stream->in_partial_segments && hls_stream->part_idx > 0)
|
|
return FALSE;
|
|
|
|
if (hls_stream->is_variant) {
|
|
gdouble play_rate = gst_adaptive_demux_play_rate (demux);
|
|
gboolean changed = FALSE;
|
|
|
|
/* If not calculated yet, continue using start bitrate */
|
|
if (bitrate == 0)
|
|
bitrate = hlsdemux->start_bitrate;
|
|
|
|
/* Handle variant streams */
|
|
GST_DEBUG_OBJECT (hlsdemux,
|
|
"Checking playlist change for main variant stream");
|
|
if (!gst_hls_demux_change_variant_playlist (hlsdemux,
|
|
hlsdemux->current_variant->iframe,
|
|
bitrate / MAX (1.0, ABS (play_rate)), &changed)) {
|
|
GST_ERROR_OBJECT (hlsdemux, "Failed to choose a new variant to play");
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (hlsdemux, "Returning changed: %d", changed);
|
|
return changed;
|
|
}
|
|
|
|
/* Handle rendition streams */
|
|
return gst_hls_demux_update_rendition_stream_uri (hlsdemux, hls_stream, NULL);
|
|
}
|
|
|
|
#if defined(HAVE_OPENSSL)
|
|
static gboolean
|
|
gst_hls_demux_stream_decrypt_start (GstHLSDemuxStream * stream,
|
|
const guint8 * key_data, const guint8 * iv_data)
|
|
{
|
|
EVP_CIPHER_CTX *ctx;
|
|
#if OPENSSL_VERSION_NUMBER < 0x10100000L
|
|
EVP_CIPHER_CTX_init (&stream->aes_ctx);
|
|
ctx = &stream->aes_ctx;
|
|
#else
|
|
stream->aes_ctx = EVP_CIPHER_CTX_new ();
|
|
ctx = stream->aes_ctx;
|
|
#endif
|
|
if (!EVP_DecryptInit_ex (ctx, EVP_aes_128_cbc (), NULL, key_data, iv_data))
|
|
return FALSE;
|
|
EVP_CIPHER_CTX_set_padding (ctx, 0);
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
decrypt_fragment (GstHLSDemuxStream * stream, gsize length,
|
|
const guint8 * encrypted_data, guint8 * decrypted_data)
|
|
{
|
|
int len, flen = 0;
|
|
EVP_CIPHER_CTX *ctx;
|
|
|
|
#if OPENSSL_VERSION_NUMBER < 0x10100000L
|
|
ctx = &stream->aes_ctx;
|
|
#else
|
|
ctx = stream->aes_ctx;
|
|
#endif
|
|
|
|
if (G_UNLIKELY (length > G_MAXINT || length % 16 != 0))
|
|
return FALSE;
|
|
|
|
len = (int) length;
|
|
if (!EVP_DecryptUpdate (ctx, decrypted_data, &len, encrypted_data, len))
|
|
return FALSE;
|
|
EVP_DecryptFinal_ex (ctx, decrypted_data + len, &flen);
|
|
g_return_val_if_fail (len + flen == length, FALSE);
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_hls_demux_stream_decrypt_end (GstHLSDemuxStream * stream)
|
|
{
|
|
#if OPENSSL_VERSION_NUMBER < 0x10100000L
|
|
EVP_CIPHER_CTX_cleanup (&stream->aes_ctx);
|
|
#else
|
|
EVP_CIPHER_CTX_free (stream->aes_ctx);
|
|
stream->aes_ctx = NULL;
|
|
#endif
|
|
}
|
|
|
|
#elif defined(HAVE_NETTLE)
|
|
static gboolean
|
|
gst_hls_demux_stream_decrypt_start (GstHLSDemuxStream * stream,
|
|
const guint8 * key_data, const guint8 * iv_data)
|
|
{
|
|
aes128_set_decrypt_key (&stream->aes_ctx.ctx, key_data);
|
|
CBC_SET_IV (&stream->aes_ctx, iv_data);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
decrypt_fragment (GstHLSDemuxStream * stream, gsize length,
|
|
const guint8 * encrypted_data, guint8 * decrypted_data)
|
|
{
|
|
if (length % 16 != 0)
|
|
return FALSE;
|
|
|
|
CBC_DECRYPT (&stream->aes_ctx, aes128_decrypt, length, decrypted_data,
|
|
encrypted_data);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_hls_demux_stream_decrypt_end (GstHLSDemuxStream * stream)
|
|
{
|
|
/* NOP */
|
|
}
|
|
|
|
#elif defined(HAVE_LIBGCRYPT)
|
|
static gboolean
|
|
gst_hls_demux_stream_decrypt_start (GstHLSDemuxStream * stream,
|
|
const guint8 * key_data, const guint8 * iv_data)
|
|
{
|
|
gcry_error_t err = 0;
|
|
gboolean ret = FALSE;
|
|
|
|
err =
|
|
gcry_cipher_open (&stream->aes_ctx, GCRY_CIPHER_AES128,
|
|
GCRY_CIPHER_MODE_CBC, 0);
|
|
if (err)
|
|
goto out;
|
|
err = gcry_cipher_setkey (stream->aes_ctx, key_data, 16);
|
|
if (err)
|
|
goto out;
|
|
err = gcry_cipher_setiv (stream->aes_ctx, iv_data, 16);
|
|
if (!err)
|
|
ret = TRUE;
|
|
|
|
out:
|
|
if (!ret)
|
|
if (stream->aes_ctx)
|
|
gcry_cipher_close (stream->aes_ctx);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
decrypt_fragment (GstHLSDemuxStream * stream, gsize length,
|
|
const guint8 * encrypted_data, guint8 * decrypted_data)
|
|
{
|
|
gcry_error_t err = 0;
|
|
|
|
err = gcry_cipher_decrypt (stream->aes_ctx, decrypted_data, length,
|
|
encrypted_data, length);
|
|
|
|
return err == 0;
|
|
}
|
|
|
|
static void
|
|
gst_hls_demux_stream_decrypt_end (GstHLSDemuxStream * stream)
|
|
{
|
|
if (stream->aes_ctx) {
|
|
gcry_cipher_close (stream->aes_ctx);
|
|
stream->aes_ctx = NULL;
|
|
}
|
|
}
|
|
|
|
#else
|
|
/* NO crypto available */
|
|
static gboolean
|
|
gst_hls_demux_stream_decrypt_start (GstHLSDemuxStream * stream,
|
|
const guint8 * key_data, const guint8 * iv_data)
|
|
{
|
|
GST_ERROR ("No crypto available");
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
decrypt_fragment (GstHLSDemuxStream * stream, gsize length,
|
|
const guint8 * encrypted_data, guint8 * decrypted_data)
|
|
{
|
|
GST_ERROR ("Cannot decrypt fragment, no crypto available");
|
|
return FALSE;
|
|
}
|
|
|
|
static void
|
|
gst_hls_demux_stream_decrypt_end (GstHLSDemuxStream * stream)
|
|
{
|
|
return;
|
|
}
|
|
#endif
|
|
|
|
static GstBuffer *
|
|
gst_hls_demux_decrypt_fragment (GstHLSDemux * demux, GstHLSDemuxStream * stream,
|
|
GstBuffer * encrypted_buffer, GError ** err)
|
|
{
|
|
GstBuffer *decrypted_buffer = NULL;
|
|
GstMapInfo encrypted_info, decrypted_info;
|
|
|
|
decrypted_buffer =
|
|
gst_buffer_new_allocate (NULL, gst_buffer_get_size (encrypted_buffer),
|
|
NULL);
|
|
|
|
gst_buffer_map (encrypted_buffer, &encrypted_info, GST_MAP_READ);
|
|
gst_buffer_map (decrypted_buffer, &decrypted_info, GST_MAP_WRITE);
|
|
|
|
if (!decrypt_fragment (stream, encrypted_info.size,
|
|
encrypted_info.data, decrypted_info.data))
|
|
goto decrypt_error;
|
|
|
|
|
|
gst_buffer_unmap (decrypted_buffer, &decrypted_info);
|
|
gst_buffer_unmap (encrypted_buffer, &encrypted_info);
|
|
|
|
gst_buffer_unref (encrypted_buffer);
|
|
|
|
return decrypted_buffer;
|
|
|
|
decrypt_error:
|
|
GST_ERROR_OBJECT (demux, "Failed to decrypt fragment");
|
|
g_set_error (err, GST_STREAM_ERROR, GST_STREAM_ERROR_DECRYPT,
|
|
"Failed to decrypt fragment");
|
|
|
|
gst_buffer_unmap (decrypted_buffer, &decrypted_info);
|
|
gst_buffer_unmap (encrypted_buffer, &encrypted_info);
|
|
|
|
gst_buffer_unref (encrypted_buffer);
|
|
gst_buffer_unref (decrypted_buffer);
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static GstClockTime
|
|
gst_hls_demux_stream_get_presentation_offset (GstAdaptiveDemux2Stream * stream)
|
|
{
|
|
GstHLSDemux *hlsdemux = (GstHLSDemux *) stream->demux;
|
|
GstHLSDemuxStream *hls_stream = (GstHLSDemuxStream *) stream;
|
|
|
|
GST_DEBUG_OBJECT (stream, "presentation_offset %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (hls_stream->presentation_offset));
|
|
|
|
/* If this stream and the variant stream are ISOBMFF, returns the presentation
|
|
* offset of the variant stream */
|
|
if (hls_stream->parser_type == GST_HLS_PARSER_ISOBMFF
|
|
&& hlsdemux->main_stream->parser_type == GST_HLS_PARSER_ISOBMFF)
|
|
return hlsdemux->main_stream->presentation_offset;
|
|
return hls_stream->presentation_offset;
|
|
}
|