/* GStreamer * Copyright (C) 2010 Marc-Andre Lureau * Copyright (C) 2010 Andoni Morales Alastruey * Copyright (C) 2011, Hewlett-Packard Development Company, L.P. * Author: Youness Alaoui , Collabora Ltd. * Author: Sebastian Dröge , Collabora Ltd. * Copyright (C) 2014 Sebastian Dröge * Copyright (C) 2015 Tim-Philipp Müller * * Copyright (C) 2021-2022 Centricular Ltd * Author: Edward Hervey * Author: Jan Schmidt * * gsthlsdemux-stream.c: * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include #include "gsthlsdemux.h" #include "gsthlsdemux-stream.h" GST_DEBUG_CATEGORY_EXTERN (gst_hls_demux2_debug); #define GST_CAT_DEFAULT gst_hls_demux2_debug /* Maximum values for mpeg-ts DTS values */ #define MPEG_TS_MAX_PTS (((((guint64)1) << 33) * (guint64)100000) / 9) static GstBuffer *gst_hls_demux_decrypt_fragment (GstHLSDemux * demux, GstHLSDemuxStream * stream, GstBuffer * encrypted_buffer, GError ** err); static gboolean gst_hls_demux_stream_decrypt_start (GstHLSDemuxStream * stream, const guint8 * key_data, const guint8 * iv_data); static void gst_hls_demux_stream_decrypt_end (GstHLSDemuxStream * stream); static gboolean gst_hls_demux_stream_start_fragment (GstAdaptiveDemux2Stream * stream); static GstFlowReturn gst_hls_demux_stream_finish_fragment (GstAdaptiveDemux2Stream * stream); static GstFlowReturn gst_hls_demux_stream_data_received (GstAdaptiveDemux2Stream * stream, GstBuffer * buffer); static gboolean gst_hls_demux_stream_has_next_fragment (GstAdaptiveDemux2Stream * stream); static GstFlowReturn gst_hls_demux_stream_advance_fragment (GstAdaptiveDemux2Stream * stream); static GstFlowReturn gst_hls_demux_stream_update_fragment_info (GstAdaptiveDemux2Stream * stream); static GstFlowReturn gst_hls_demux_stream_submit_request (GstAdaptiveDemux2Stream * stream, DownloadRequest * download_req); static void gst_hls_demux_stream_start (GstAdaptiveDemux2Stream * stream); static void gst_hls_demux_stream_stop (GstAdaptiveDemux2Stream * stream); static void gst_hls_demux_stream_create_tracks (GstAdaptiveDemux2Stream * stream); static gboolean gst_hls_demux_stream_select_bitrate (GstAdaptiveDemux2Stream * stream, guint64 bitrate); static GstClockTime gst_hls_demux_stream_get_presentation_offset (GstAdaptiveDemux2Stream * stream); static void gst_hls_demux_stream_finalize (GObject * object); #define gst_hls_demux_stream_parent_class stream_parent_class G_DEFINE_TYPE (GstHLSDemuxStream, gst_hls_demux_stream, GST_TYPE_ADAPTIVE_DEMUX2_STREAM); static void gst_hls_demux_stream_class_init (GstHLSDemuxStreamClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstAdaptiveDemux2StreamClass *adaptivedemux2stream_class = GST_ADAPTIVE_DEMUX2_STREAM_CLASS (klass); gobject_class->finalize = gst_hls_demux_stream_finalize; adaptivedemux2stream_class->update_fragment_info = gst_hls_demux_stream_update_fragment_info; adaptivedemux2stream_class->submit_request = gst_hls_demux_stream_submit_request; adaptivedemux2stream_class->has_next_fragment = gst_hls_demux_stream_has_next_fragment; adaptivedemux2stream_class->stream_seek = gst_hls_demux_stream_seek; adaptivedemux2stream_class->advance_fragment = gst_hls_demux_stream_advance_fragment; adaptivedemux2stream_class->select_bitrate = gst_hls_demux_stream_select_bitrate; adaptivedemux2stream_class->start = gst_hls_demux_stream_start; adaptivedemux2stream_class->stop = gst_hls_demux_stream_stop; adaptivedemux2stream_class->create_tracks = gst_hls_demux_stream_create_tracks; adaptivedemux2stream_class->start_fragment = gst_hls_demux_stream_start_fragment; adaptivedemux2stream_class->finish_fragment = gst_hls_demux_stream_finish_fragment; adaptivedemux2stream_class->data_received = gst_hls_demux_stream_data_received; adaptivedemux2stream_class->get_presentation_offset = gst_hls_demux_stream_get_presentation_offset; } static void gst_hls_demux_stream_init (GstHLSDemuxStream * stream) { stream->parser_type = GST_HLS_PARSER_NONE; stream->do_typefind = TRUE; stream->reset_pts = TRUE; stream->presentation_offset = 60 * GST_SECOND; stream->pdt_tag_sent = FALSE; } void gst_hls_demux_stream_clear_pending_data (GstHLSDemuxStream * hls_stream, gboolean force) { GST_DEBUG_OBJECT (hls_stream, "force : %d", force); if (hls_stream->pending_encrypted_data) gst_adapter_clear (hls_stream->pending_encrypted_data); gst_buffer_replace (&hls_stream->pending_decrypted_buffer, NULL); gst_buffer_replace (&hls_stream->pending_typefind_buffer, NULL); if (force || !hls_stream->pending_data_is_header) { gst_buffer_replace (&hls_stream->pending_segment_data, NULL); hls_stream->pending_data_is_header = FALSE; } hls_stream->current_offset = -1; hls_stream->process_buffer_content = TRUE; gst_hls_demux_stream_decrypt_end (hls_stream); } GstFlowReturn gst_hls_demux_stream_seek (GstAdaptiveDemux2Stream * stream, gboolean forward, GstSeekFlags flags, GstClockTimeDiff ts, GstClockTimeDiff * final_ts) { GstFlowReturn ret = GST_FLOW_OK; GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (stream); GstHLSDemux *hlsdemux = (GstHLSDemux *) stream->demux; GST_DEBUG_OBJECT (stream, "is_variant:%d media:%p current_variant:%p forward:%d ts:%" GST_TIME_FORMAT, hls_stream->is_variant, hls_stream->current_rendition, hlsdemux->current_variant, forward, GST_TIME_ARGS (ts)); /* If this stream doesn't have a playlist yet, we can't seek on it */ if (!hls_stream->playlist_fetched) { return GST_ADAPTIVE_DEMUX_FLOW_BUSY; } /* Allow jumping to partial segments in the last 2 segments in LL-HLS */ if (GST_HLS_MEDIA_PLAYLIST_IS_LIVE (hls_stream->playlist)) flags |= GST_HLS_M3U8_SEEK_FLAG_ALLOW_PARTIAL; GstM3U8SeekResult seek_result; if (gst_hls_media_playlist_seek (hls_stream->playlist, forward, flags, ts, &seek_result)) { if (hls_stream->current_segment) 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; hls_stream->reset_pts = TRUE; if (final_ts) *final_ts = seek_result.stream_time; } else { GST_WARNING_OBJECT (stream, "Seeking failed"); ret = GST_FLOW_ERROR; } return ret; } static GstCaps * get_caps_of_stream_type (GstCaps * full_caps, GstStreamType streamtype) { GstCaps *ret = NULL; guint i; for (i = 0; i < gst_caps_get_size (full_caps); i++) { GstStructure *st = gst_caps_get_structure (full_caps, i); if (gst_hls_get_stream_type_from_structure (st) == streamtype) { ret = gst_caps_new_empty (); gst_caps_append_structure (ret, gst_structure_copy (st)); break; } } return ret; } static GstHLSRenditionStream * find_uriless_rendition (GstHLSDemux * demux, GstStreamType stream_type) { GList *tmp; for (tmp = demux->master->renditions; tmp; tmp = tmp->next) { GstHLSRenditionStream *media = tmp->data; if (media->uri == NULL && gst_stream_type_from_hls_type (media->mtype) == stream_type) return media; } return NULL; } static void gst_hls_demux_stream_create_tracks (GstAdaptiveDemux2Stream * stream) { GstHLSDemux *hlsdemux = (GstHLSDemux *) stream->demux; GstHLSDemuxStream *hlsdemux_stream = (GstHLSDemuxStream *) stream; guint i; GstStreamType uriless_types = 0; GstCaps *variant_caps = NULL; GST_DEBUG_OBJECT (stream, "Update tracks of variant stream"); if (hlsdemux->master->have_codecs) { variant_caps = gst_hls_master_playlist_get_common_caps (hlsdemux->master); } /* Use the stream->stream_collection and manifest to create the appropriate tracks */ for (i = 0; i < gst_stream_collection_get_size (stream->stream_collection); i++) { GstStream *gst_stream = gst_stream_collection_get_stream (stream->stream_collection, i); GstStreamType stream_type = gst_stream_get_stream_type (gst_stream); GstAdaptiveDemuxTrack *track; GstHLSRenditionStream *embedded_media = NULL; /* tracks from the variant streams should be prefered over those provided by renditions */ GstStreamFlags flags = gst_stream_get_stream_flags (gst_stream) | GST_STREAM_FLAG_SELECT; GstCaps *manifest_caps = NULL; if (stream_type == GST_STREAM_TYPE_UNKNOWN) continue; if (variant_caps) manifest_caps = get_caps_of_stream_type (variant_caps, stream_type); hlsdemux_stream->rendition_type |= stream_type; if ((uriless_types & stream_type) == 0) { /* Do we have a uriless media for this stream type */ /* Find if there is a rendition without URI, it will be provided by this variant */ embedded_media = find_uriless_rendition (hlsdemux, stream_type); /* Remember we used this type for a embedded media */ uriless_types |= stream_type; } if (embedded_media) { GstTagList *tags = gst_stream_get_tags (gst_stream); GST_DEBUG_OBJECT (stream, "Adding track '%s' to main variant stream", embedded_media->name); track = gst_hls_demux_new_track_for_rendition (hlsdemux, embedded_media, manifest_caps, flags, tags ? gst_tag_list_make_writable (tags) : tags); } else { gchar *stream_id; stream_id = g_strdup_printf ("main-%s-%d", gst_stream_type_get_name (stream_type), i); GST_DEBUG_OBJECT (stream, "Adding track '%s' to main variant stream", stream_id); track = gst_adaptive_demux_track_new (stream->demux, stream_type, flags, stream_id, manifest_caps, NULL); g_free (stream_id); } track->upstream_stream_id = g_strdup (gst_stream_get_stream_id (gst_stream)); gst_adaptive_demux2_stream_add_track (stream, track); gst_adaptive_demux_track_unref (track); } if (variant_caps) gst_caps_unref (variant_caps); /* Update the stream object with rendition types. * FIXME: rendition_type could be removed */ stream->stream_type = hlsdemux_stream->rendition_type; } static gboolean gst_hls_demux_stream_start_fragment (GstAdaptiveDemux2Stream * stream) { GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (stream); GstHLSDemux *hlsdemux = GST_HLS_DEMUX_CAST (stream->demux); const GstHLSKey *key; GstHLSMediaPlaylist *m3u8; GST_DEBUG_OBJECT (stream, "Fragment starting"); gst_hls_demux_stream_clear_pending_data (hls_stream, FALSE); /* If no decryption is needed, there's nothing to be done here */ if (hls_stream->current_key == NULL) return TRUE; m3u8 = hls_stream->playlist; key = gst_hls_demux_get_key (hlsdemux, hls_stream->current_key, m3u8->uri, m3u8->allowcache); if (key == NULL) goto key_failed; if (!gst_hls_demux_stream_decrypt_start (hls_stream, key->data, hls_stream->current_iv)) goto decrypt_start_failed; return TRUE; key_failed: { GST_ELEMENT_ERROR (hlsdemux, STREAM, DECRYPT_NOKEY, ("Couldn't retrieve key for decryption"), (NULL)); GST_WARNING_OBJECT (hlsdemux, "Failed to decrypt data"); return FALSE; } decrypt_start_failed: { GST_ELEMENT_ERROR (hlsdemux, STREAM, DECRYPT, ("Failed to start decrypt"), ("Couldn't set key and IV or plugin was built without crypto library")); return FALSE; } } static GstHLSParserType caps_to_parser_type (const GstCaps * caps) { const GstStructure *s = gst_caps_get_structure (caps, 0); if (gst_structure_has_name (s, "video/mpegts")) return GST_HLS_PARSER_MPEGTS; if (gst_structure_has_name (s, "application/x-id3")) return GST_HLS_PARSER_ID3; if (gst_structure_has_name (s, "application/x-subtitle-vtt")) return GST_HLS_PARSER_WEBVTT; if (gst_structure_has_name (s, "video/quicktime")) return GST_HLS_PARSER_ISOBMFF; return GST_HLS_PARSER_NONE; } /* Identify the nature of data for this stream * * Will also setup the appropriate parser (tsreader) if needed * * Consumes the input buffer when it returns FALSE, but * replaces / returns the input buffer in the `buffer` parameter * when it returns TRUE. * * Returns TRUE if we are done with typefinding */ static gboolean gst_hls_demux_typefind_stream (GstHLSDemux * hlsdemux, GstAdaptiveDemux2Stream * stream, GstBuffer ** out_buffer, gboolean at_eos, GstFlowReturn * ret) { GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (stream); // FIXME: pass HlsStream into function GstCaps *caps = NULL; guint buffer_size; GstTypeFindProbability prob = GST_TYPE_FIND_NONE; GstMapInfo info; GstBuffer *buffer = *out_buffer; if (hls_stream->pending_typefind_buffer) { /* Append to the existing typefind buffer and create a new one that * we'll return (or consume below) */ buffer = *out_buffer = gst_buffer_append (hls_stream->pending_typefind_buffer, buffer); hls_stream->pending_typefind_buffer = NULL; } gst_buffer_map (buffer, &info, GST_MAP_READ); buffer_size = info.size; /* Typefind could miss if buffer is too small. In this case we * will retry later */ if (buffer_size >= (2 * 1024) || at_eos) { caps = gst_type_find_helper_for_data (GST_OBJECT_CAST (hlsdemux), info.data, info.size, &prob); } if (G_UNLIKELY (!caps)) { /* Won't need this mapping any more all paths return inside this if() */ gst_buffer_unmap (buffer, &info); /* 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, ("Could not determine type of stream"), (NULL)); gst_buffer_unref (buffer); *ret = GST_FLOW_NOT_NEGOTIATED; } else { GST_LOG_OBJECT (stream, "Not enough data to typefind"); 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; gst_time_map_set_values (map, current_stream_time, internal_time, current_segment->datetime); 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)); gst_time_map_set_values (map, current_stream_time, internal_time, current_segment->datetime); 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)); /* We allow a tolerance of 3-4 frames between the estimated and observed * stream time. */ if (ABS (difference) > 100 * 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 = current_stream_time; gst_time_map_set_values (map, current_stream_time, internal_time, hls_stream->current_segment->datetime); } } 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); gboolean found_segment_discont = FALSE; /* 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) { if (!gst_hls_media_playlist_sync_to_playlist (new_playlist, stream->playlist, &found_segment_discont)) { /* 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, &found_segment_discont)) { GST_DEBUG_OBJECT (stream, "Could not do fallback synchronization of rendition stream to variant stream"); goto lost_sync; } } } else { found_segment_discont = TRUE; 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, NULL); } } GST_DEBUG_OBJECT (stream, "Synchronized playlist. Update is discont : %d", found_segment_discont); if (found_segment_discont) { stream->pending_discont = TRUE; } 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; stream->pending_discont = 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 ((hlsdemux_stream->pending_discont || 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 || hlsdemux_stream->pending_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; hlsdemux_stream->pending_discont = FALSE; 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; }