gstreamer/subprojects/gst-plugins-good/ext/adaptivedemux2/hls/gsthlsdemux.c
Edward Hervey 5d0b112c0c adaptivedemux2/hlsdemux2: Handle loss of sync when dowloading.
Media playlist updates and fragment downloads happen in an interleaved
fashion. When a media playlist update fails *while* a segment is being
downloaded, this means we lost synchronization.

Properly propagate and handle this

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2839>
2022-08-04 14:49:27 +01:00

2923 lines
94 KiB
C

/* GStreamer
* Copyright (C) 2010 Marc-Andre Lureau <marcandre.lureau@gmail.com>
* Copyright (C) 2010 Andoni Morales Alastruey <ylatuya@gmail.com>
* Copyright (C) 2011, Hewlett-Packard Development Company, L.P.
* Author: Youness Alaoui <youness.alaoui@collabora.co.uk>, Collabora Ltd.
* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>, Collabora Ltd.
* Copyright (C) 2014 Sebastian Dröge <sebastian@centricular.com>
* Copyright (C) 2015 Tim-Philipp Müller <tim@centricular.com>
*
* Copyright (C) 2021-2022 Centricular Ltd
* Author: Edward Hervey <edward@centricular.com>
* Author: Jan Schmidt <jan@centricular.com>
*
* Gsthlsdemux.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.
*/
/**
* SECTION:element-hlsdemux2
* @title: hlsdemux2
*
* HTTP Live Streaming demuxer element.
*
* ## Example launch line
* |[
* gst-launch-1.0 playbin3 uri=http://devimages.apple.com/iphone/samples/bipbop/gear4/prog_index.m3u8
* ]|
*
* Since: 1.22
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/base/gsttypefindhelper.h>
#include <gst/tag/tag.h>
#include "gsthlselements.h"
#include "gstadaptivedemuxelements.h"
#include "gsthlsdemux.h"
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-hls"));
GST_DEBUG_CATEGORY (gst_hls_demux2_debug);
#define GST_CAT_DEFAULT gst_hls_demux2_debug
enum
{
PROP_0,
PROP_START_BITRATE,
};
#define DEFAULT_START_BITRATE 0
/* Maximum values for mpeg-ts DTS values */
#define MPEG_TS_MAX_PTS (((((guint64)1) << 33) * (guint64)100000) / 9)
/* GObject */
static void gst_hls_demux_finalize (GObject * obj);
/* GstElement */
static GstStateChangeReturn
gst_hls_demux_change_state (GstElement * element, GstStateChange transition);
/* GstHLSDemux */
static GstFlowReturn gst_hls_demux_update_playlist (GstHLSDemux * demux,
gboolean update, GError ** err);
/* FIXME: the return value is never used? */
static gboolean gst_hls_demux_change_playlist (GstHLSDemux * demux,
guint max_bitrate, gboolean * changed);
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_is_live (GstAdaptiveDemux * demux);
static GstClockTime gst_hls_demux_get_duration (GstAdaptiveDemux * demux);
static gint64 gst_hls_demux_get_manifest_update_interval (GstAdaptiveDemux *
demux);
static gboolean gst_hls_demux_process_manifest (GstAdaptiveDemux * demux,
GstBuffer * buf);
static GstFlowReturn gst_hls_demux_stream_update_rendition_playlist (GstHLSDemux
* demux, GstHLSDemuxStream * stream);
static GstFlowReturn gst_hls_demux_update_manifest (GstAdaptiveDemux * demux);
static void setup_initial_playlist (GstHLSDemux * demux,
GstHLSMediaPlaylist * playlist);
static void gst_hls_demux_add_time_mapping (GstHLSDemux * demux,
gint64 dsn, GstClockTimeDiff stream_time, GDateTime * pdt);
static void
gst_hls_update_time_mappings (GstHLSDemux * demux,
GstHLSMediaPlaylist * playlist);
static void gst_hls_prune_time_mappings (GstHLSDemux * demux);
static gboolean gst_hls_demux_seek (GstAdaptiveDemux * demux, GstEvent * seek);
static GstFlowReturn gst_hls_demux_stream_seek (GstAdaptiveDemux2Stream *
stream, gboolean forward, GstSeekFlags flags, GstClockTimeDiff ts,
GstClockTimeDiff * final_ts);
static gboolean
gst_hls_demux_start_fragment (GstAdaptiveDemux * demux,
GstAdaptiveDemux2Stream * stream);
static GstFlowReturn gst_hls_demux_finish_fragment (GstAdaptiveDemux * demux,
GstAdaptiveDemux2Stream * stream);
static GstFlowReturn gst_hls_demux_data_received (GstAdaptiveDemux * demux,
GstAdaptiveDemux2Stream * stream, GstBuffer * buffer);
static gboolean gst_hls_demux_stream_has_next_fragment (GstAdaptiveDemux2Stream
* stream);
static GstFlowReturn gst_hls_demux_advance_fragment (GstAdaptiveDemux2Stream *
stream);
static GstFlowReturn gst_hls_demux_update_fragment_info (GstAdaptiveDemux2Stream
* stream);
static gboolean gst_hls_demux_stream_can_start (GstAdaptiveDemux * demux,
GstAdaptiveDemux2Stream * stream);
static void gst_hls_demux_stream_update_tracks (GstAdaptiveDemux * demux,
GstAdaptiveDemux2Stream * stream);
static gboolean gst_hls_demux_select_bitrate (GstAdaptiveDemux2Stream * stream,
guint64 bitrate);
static void gst_hls_demux_reset (GstAdaptiveDemux * demux);
static gboolean gst_hls_demux_get_live_seek_range (GstAdaptiveDemux * demux,
gint64 * start, gint64 * stop);
static GstClockTime gst_hls_demux_get_presentation_offset (GstAdaptiveDemux *
demux, GstAdaptiveDemux2Stream * stream);
static void gst_hls_demux_set_current_variant (GstHLSDemux * hlsdemux,
GstHLSVariantStream * variant);
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 gboolean hlsdemux2_element_init (GstPlugin * plugin);
GST_ELEMENT_REGISTER_DEFINE_CUSTOM (hlsdemux2, hlsdemux2_element_init);
static void
gst_hls_demux_stream_class_init (GstHLSDemuxStreamClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->finalize = gst_hls_demux_stream_finalize;
}
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;
}
typedef struct _GstHLSDemux2 GstHLSDemux2;
typedef struct _GstHLSDemux2Class GstHLSDemux2Class;
#define gst_hls_demux2_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstHLSDemux2, gst_hls_demux2, GST_TYPE_ADAPTIVE_DEMUX,
hls2_element_init ());
static void
gst_hls_demux_finalize (GObject * obj)
{
GstHLSDemux *demux = GST_HLS_DEMUX (obj);
gst_hls_demux_reset (GST_ADAPTIVE_DEMUX_CAST (demux));
g_mutex_clear (&demux->keys_lock);
if (demux->keys) {
g_hash_table_unref (demux->keys);
demux->keys = NULL;
}
G_OBJECT_CLASS (parent_class)->finalize (obj);
}
static void
gst_hls_demux_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstHLSDemux *demux = GST_HLS_DEMUX (object);
switch (prop_id) {
case PROP_START_BITRATE:
demux->start_bitrate = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_hls_demux_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstHLSDemux *demux = GST_HLS_DEMUX (object);
switch (prop_id) {
case PROP_START_BITRATE:
g_value_set_uint (value, demux->start_bitrate);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_hls_demux2_class_init (GstHLSDemux2Class * klass)
{
GObjectClass *gobject_class;
GstElementClass *element_class;
GstAdaptiveDemuxClass *adaptivedemux_class;
gobject_class = (GObjectClass *) klass;
element_class = (GstElementClass *) klass;
adaptivedemux_class = (GstAdaptiveDemuxClass *) klass;
gobject_class->set_property = gst_hls_demux_set_property;
gobject_class->get_property = gst_hls_demux_get_property;
gobject_class->finalize = gst_hls_demux_finalize;
g_object_class_install_property (gobject_class, PROP_START_BITRATE,
g_param_spec_uint ("start-bitrate", "Starting Bitrate",
"Initial bitrate to use to choose first alternate (0 = automatic) (bits/s)",
0, G_MAXUINT, DEFAULT_START_BITRATE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
element_class->change_state = GST_DEBUG_FUNCPTR (gst_hls_demux_change_state);
gst_element_class_add_static_pad_template (element_class, &sinktemplate);
gst_element_class_set_static_metadata (element_class,
"HLS Demuxer",
"Codec/Demuxer/Adaptive",
"HTTP Live Streaming demuxer",
"Edward Hervey <edward@centricular.com>\n"
"Jan Schmidt <jan@centricular.com>");
adaptivedemux_class->is_live = gst_hls_demux_is_live;
adaptivedemux_class->get_live_seek_range = gst_hls_demux_get_live_seek_range;
adaptivedemux_class->get_presentation_offset =
gst_hls_demux_get_presentation_offset;
adaptivedemux_class->get_duration = gst_hls_demux_get_duration;
adaptivedemux_class->get_manifest_update_interval =
gst_hls_demux_get_manifest_update_interval;
adaptivedemux_class->process_manifest = gst_hls_demux_process_manifest;
adaptivedemux_class->update_manifest = gst_hls_demux_update_manifest;
adaptivedemux_class->reset = gst_hls_demux_reset;
adaptivedemux_class->seek = gst_hls_demux_seek;
adaptivedemux_class->stream_seek = gst_hls_demux_stream_seek;
adaptivedemux_class->stream_has_next_fragment =
gst_hls_demux_stream_has_next_fragment;
adaptivedemux_class->stream_advance_fragment = gst_hls_demux_advance_fragment;
adaptivedemux_class->stream_update_fragment_info =
gst_hls_demux_update_fragment_info;
adaptivedemux_class->stream_select_bitrate = gst_hls_demux_select_bitrate;
adaptivedemux_class->stream_can_start = gst_hls_demux_stream_can_start;
adaptivedemux_class->stream_update_tracks =
gst_hls_demux_stream_update_tracks;
adaptivedemux_class->start_fragment = gst_hls_demux_start_fragment;
adaptivedemux_class->finish_fragment = gst_hls_demux_finish_fragment;
adaptivedemux_class->data_received = gst_hls_demux_data_received;
}
static void
gst_hls_demux2_init (GstHLSDemux * demux)
{
demux->keys = g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_free);
g_mutex_init (&demux->keys_lock);
}
static GstStateChangeReturn
gst_hls_demux_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstHLSDemux *demux = GST_HLS_DEMUX (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_hls_demux_reset (GST_ADAPTIVE_DEMUX_CAST (demux));
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_hls_demux_reset (GST_ADAPTIVE_DEMUX_CAST (demux));
g_hash_table_remove_all (demux->keys);
break;
default:
break;
}
return ret;
}
static guint64
gst_hls_demux_get_bitrate (GstHLSDemux * hlsdemux)
{
GstAdaptiveDemux *demux = GST_ADAPTIVE_DEMUX_CAST (hlsdemux);
/* FIXME !!!
*
* No, there isn't a single output :D */
/* Valid because hlsdemux only has a single output */
if (demux->input_period->streams) {
GstAdaptiveDemux2Stream *stream = demux->input_period->streams->data;
return stream->current_download_rate;
}
return 0;
}
static 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);
}
static void
gst_hls_demux_clear_all_pending_data (GstHLSDemux * hlsdemux)
{
GstAdaptiveDemux *demux = (GstAdaptiveDemux *) hlsdemux;
GList *walk;
if (!demux->input_period)
return;
for (walk = demux->input_period->streams; walk != NULL; walk = walk->next) {
GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (walk->data);
gst_hls_demux_stream_clear_pending_data (hls_stream, TRUE);
}
}
#define SEEK_UPDATES_PLAY_POSITION(r, start_type, stop_type) \
((r >= 0 && start_type != GST_SEEK_TYPE_NONE) || \
(r < 0 && stop_type != GST_SEEK_TYPE_NONE))
#define IS_SNAP_SEEK(f) (f & (GST_SEEK_FLAG_SNAP_BEFORE | \
GST_SEEK_FLAG_SNAP_AFTER | \
GST_SEEK_FLAG_SNAP_NEAREST | \
GST_SEEK_FLAG_TRICKMODE_KEY_UNITS | \
GST_SEEK_FLAG_KEY_UNIT))
static gboolean
gst_hls_demux_seek (GstAdaptiveDemux * demux, GstEvent * seek)
{
GstHLSDemux *hlsdemux = GST_HLS_DEMUX_CAST (demux);
GstFormat format;
GstSeekFlags flags;
GstSeekType start_type, stop_type;
gint64 start, stop;
gdouble rate, old_rate;
GList *walk;
gint64 current_pos, target_pos, final_pos;
guint64 bitrate;
gst_event_parse_seek (seek, &rate, &format, &flags, &start_type, &start,
&stop_type, &stop);
if (!SEEK_UPDATES_PLAY_POSITION (rate, start_type, stop_type)) {
/* nothing to do if we don't have to update the current position */
return TRUE;
}
old_rate = demux->segment.rate;
bitrate = gst_hls_demux_get_bitrate (hlsdemux);
/* Use I-frame variants for trick modes */
if (hlsdemux->master->iframe_variants != NULL
&& rate < -1.0 && old_rate >= -1.0 && old_rate <= 1.0) {
GError *err = NULL;
/* Switch to I-frame variant */
gst_hls_demux_set_current_variant (hlsdemux,
hlsdemux->master->iframe_variants->data);
if (gst_hls_demux_update_playlist (hlsdemux, FALSE, &err) != GST_FLOW_OK) {
GST_ELEMENT_ERROR_FROM_ERROR (hlsdemux, "Could not switch playlist", err);
return FALSE;
}
//hlsdemux->discont = TRUE;
gst_hls_demux_change_playlist (hlsdemux, bitrate / ABS (rate), NULL);
} else if (rate > -1.0 && rate <= 1.0 && (old_rate < -1.0 || old_rate > 1.0)) {
GError *err = NULL;
/* Switch to normal variant */
gst_hls_demux_set_current_variant (hlsdemux,
hlsdemux->master->variants->data);
if (gst_hls_demux_update_playlist (hlsdemux, FALSE, &err) != GST_FLOW_OK) {
GST_ELEMENT_ERROR_FROM_ERROR (hlsdemux, "Could not switch playlist", err);
return FALSE;
}
//hlsdemux->discont = TRUE;
/* TODO why not continue using the same? that was being used up to now? */
gst_hls_demux_change_playlist (hlsdemux, bitrate, NULL);
}
target_pos = rate < 0 ? stop : start;
final_pos = target_pos;
/* properly cleanup pending decryption status */
if (flags & GST_SEEK_FLAG_FLUSH) {
gst_hls_demux_clear_all_pending_data (hlsdemux);
gst_hls_prune_time_mappings (hlsdemux);
}
for (walk = demux->input_period->streams; walk; walk = g_list_next (walk)) {
GstAdaptiveDemux2Stream *stream =
GST_ADAPTIVE_DEMUX2_STREAM_CAST (walk->data);
/* Only seek on selected streams */
if (!gst_adaptive_demux2_stream_is_selected (stream))
continue;
if (gst_hls_demux_stream_seek (stream, rate >= 0, flags, target_pos,
&current_pos) != GST_FLOW_OK) {
GST_ERROR_OBJECT (stream, "Failed to seek on stream");
return FALSE;
}
/* FIXME: use minimum position always ? */
if (final_pos > current_pos)
final_pos = current_pos;
}
if (IS_SNAP_SEEK (flags)) {
if (rate >= 0)
gst_segment_do_seek (&demux->segment, rate, format, flags, start_type,
final_pos, stop_type, stop, NULL);
else
gst_segment_do_seek (&demux->segment, rate, format, flags, start_type,
start, stop_type, final_pos, NULL);
}
return TRUE;
}
static 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;
GstM3U8MediaSegment *new_position;
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 the rendition playlist needs to be updated, do it now */
if (!hls_stream->is_variant && !hls_stream->playlist_fetched) {
ret = gst_hls_demux_stream_update_rendition_playlist (hlsdemux, hls_stream);
if (ret != GST_FLOW_OK) {
GST_WARNING_OBJECT (stream,
"Failed to update the rendition playlist before seeking");
return ret;
}
}
new_position =
gst_hls_media_playlist_seek (hls_stream->playlist, forward, flags, ts);
if (new_position) {
if (hls_stream->current_segment)
gst_m3u8_media_segment_unref (hls_stream->current_segment);
hls_stream->current_segment = new_position;
hls_stream->reset_pts = TRUE;
if (final_ts)
*final_ts = new_position->stream_time;
} else {
GST_WARNING_OBJECT (stream, "Seeking failed");
ret = GST_FLOW_ERROR;
}
return ret;
}
static GstFlowReturn
gst_hls_demux_update_manifest (GstAdaptiveDemux * demux)
{
GstHLSDemux *hlsdemux = GST_HLS_DEMUX_CAST (demux);
return gst_hls_demux_update_playlist (hlsdemux, TRUE, NULL);
}
static GstAdaptiveDemux2Stream *
create_common_hls_stream (GstHLSDemux * demux, const gchar * name)
{
GstAdaptiveDemux2Stream *stream;
stream = g_object_new (GST_TYPE_HLS_DEMUX_STREAM, "name", name, NULL);
gst_adaptive_demux2_add_stream ((GstAdaptiveDemux *) demux, stream);
return stream;
}
static GstAdaptiveDemuxTrack *
new_track_for_rendition (GstHLSDemux * demux, GstHLSRenditionStream * rendition,
GstCaps * caps, GstStreamFlags flags, GstTagList * tags)
{
GstAdaptiveDemuxTrack *track;
gchar *stream_id;
GstStreamType stream_type = gst_stream_type_from_hls_type (rendition->mtype);
if (rendition->name)
stream_id =
g_strdup_printf ("%s-%s", gst_stream_type_get_name (stream_type),
rendition->name);
else if (rendition->lang)
stream_id =
g_strdup_printf ("%s-%s", gst_stream_type_get_name (stream_type),
rendition->lang);
else
stream_id = g_strdup (gst_stream_type_get_name (stream_type));
if (rendition->lang) {
if (tags == NULL)
tags = gst_tag_list_new_empty ();
if (gst_tag_check_language_code (rendition->lang))
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE, GST_TAG_LANGUAGE_CODE,
rendition->lang, NULL);
else
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE, GST_TAG_LANGUAGE_NAME,
rendition->lang, NULL);
}
if (stream_type == GST_STREAM_TYPE_TEXT)
flags |= GST_STREAM_FLAG_SPARSE;
if (rendition->is_default)
flags |= GST_STREAM_FLAG_SELECT;
track =
gst_adaptive_demux_track_new ((GstAdaptiveDemux *) demux, stream_type,
flags, stream_id, caps, tags);
g_free (stream_id);
return track;
}
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 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 void
gst_hls_demux_stream_update_tracks (GstAdaptiveDemux * demux,
GstAdaptiveDemux2Stream * stream)
{
GstHLSDemux *hlsdemux = (GstHLSDemux *) demux;
GstHLSDemuxStream *hlsdemux_stream = (GstHLSDemuxStream *) stream;
guint i;
GstStreamType uriless_types = 0;
GstCaps *variant_caps = NULL;
GST_DEBUG_OBJECT (demux, "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 (demux, "Adding track '%s' to main variant stream",
embedded_media->name);
track =
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 (demux, "Adding track '%s' to main variant stream",
stream_id);
track =
gst_adaptive_demux_track_new (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 void
create_main_variant_stream (GstHLSDemux * demux)
{
GstAdaptiveDemux2Stream *stream;
GstHLSDemuxStream *hlsdemux_stream;
GST_DEBUG_OBJECT (demux, "Creating main variant stream");
stream = create_common_hls_stream (demux, "hlsstream-variant");
demux->main_stream = hlsdemux_stream = (GstHLSDemuxStream *) stream;
hlsdemux_stream->is_variant = TRUE;
hlsdemux_stream->playlist_fetched = TRUE;
/* Due to HLS manifest information being so unreliable/inconsistent, we will
* create the actual tracks once we have information about the streams present
* in the variant data stream */
stream->pending_tracks = TRUE;
}
static GstHLSDemuxStream *
create_rendition_stream (GstHLSDemux * demux, GstHLSRenditionStream * media)
{
GstAdaptiveDemux2Stream *stream;
GstAdaptiveDemuxTrack *track;
GstHLSDemuxStream *hlsdemux_stream;
gchar *stream_name;
GST_DEBUG_OBJECT (demux,
"Creating stream for media %s lang:%s (%" GST_PTR_FORMAT ")", media->name,
media->lang, media->caps);
/* We can't reliably provide caps for HLS target tracks since they might
* change at any point in time */
track = new_track_for_rendition (demux, media, NULL, 0, NULL);
stream_name = g_strdup_printf ("hlsstream-%s", track->stream_id);
stream = create_common_hls_stream (demux, stream_name);
g_free (stream_name);
hlsdemux_stream = (GstHLSDemuxStream *) stream;
hlsdemux_stream->is_variant = FALSE;
hlsdemux_stream->playlist_fetched = FALSE;
stream->stream_type = hlsdemux_stream->rendition_type =
gst_stream_type_from_hls_type (media->mtype);
if (media->lang)
hlsdemux_stream->lang = g_strdup (media->lang);
if (media->name)
hlsdemux_stream->name = g_strdup (media->name);
gst_adaptive_demux2_stream_add_track (stream, track);
gst_adaptive_demux_track_unref (track);
return hlsdemux_stream;
}
static GstHLSDemuxStream *
existing_rendition_stream (GList * streams, GstHLSRenditionStream * media)
{
GList *tmp;
GstStreamType stream_type = gst_stream_type_from_hls_type (media->mtype);
for (tmp = streams; tmp; tmp = tmp->next) {
GstHLSDemuxStream *demux_stream = tmp->data;
if (demux_stream->is_variant)
continue;
if (demux_stream->rendition_type == stream_type) {
if (!g_strcmp0 (demux_stream->name, media->name))
return demux_stream;
if (media->lang && !g_strcmp0 (demux_stream->lang, media->lang))
return demux_stream;
}
}
return NULL;
}
static gboolean
gst_hls_demux_setup_streams (GstAdaptiveDemux * demux)
{
GstHLSDemux *hlsdemux = GST_HLS_DEMUX_CAST (demux);
GstHLSVariantStream *playlist = hlsdemux->current_variant;
GList *tmp;
GList *streams = NULL;
if (playlist == NULL) {
GST_WARNING_OBJECT (demux, "Can't configure streams - no variant selected");
return FALSE;
}
GST_DEBUG_OBJECT (demux, "Setting up streams");
/* If there are alternate renditions, we will produce a GstAdaptiveDemux2Stream
* and GstAdaptiveDemuxTrack for each combination of GstStreamType and other
* unique identifier (for now just language)
*
* Which actual GstHLSMedia to use for each stream will be determined based on
* the `group-id` (if present and more than one) selected on the main variant
* stream */
for (tmp = hlsdemux->master->renditions; tmp; tmp = tmp->next) {
GstHLSRenditionStream *media = tmp->data;
GstHLSDemuxStream *media_stream, *previous_media_stream;
GST_LOG_OBJECT (demux, "Rendition %s name:'%s' lang:'%s' uri:%s",
gst_stream_type_get_name (gst_stream_type_from_hls_type (media->mtype)),
media->name, media->lang, media->uri);
if (media->uri == NULL) {
GST_DEBUG_OBJECT (demux,
"Skipping media '%s' , it's provided by the variant stream",
media->name);
continue;
}
media_stream = previous_media_stream =
existing_rendition_stream (streams, media);
if (!media_stream) {
media_stream = create_rendition_stream (hlsdemux, tmp->data);
} else
GST_DEBUG_OBJECT (demux, "Re-using existing GstHLSDemuxStream %s %s",
media_stream->name, media_stream->lang);
/* Is this rendition active in the current variant ? */
if (!g_strcmp0 (playlist->media_groups[media->mtype], media->group_id)) {
GST_DEBUG_OBJECT (demux, "Enabling rendition");
if (media_stream->current_rendition)
gst_hls_rendition_stream_unref (media_stream->current_rendition);
media_stream->current_rendition = gst_hls_rendition_stream_ref (media);
}
if (!previous_media_stream)
streams = g_list_append (streams, media_stream);
}
/* Free the list (but not the contents, which are stored
* elsewhere */
if (streams)
g_list_free (streams);
create_main_variant_stream (hlsdemux);
return TRUE;
}
static const gchar *
gst_adaptive_demux_get_manifest_ref_uri (GstAdaptiveDemux * d)
{
return d->manifest_base_uri ? d->manifest_base_uri : d->manifest_uri;
}
static void
gst_hls_demux_set_current_variant (GstHLSDemux * hlsdemux,
GstHLSVariantStream * variant)
{
if (hlsdemux->current_variant == variant || variant == NULL)
return;
if (hlsdemux->current_variant != NULL) {
GST_DEBUG_OBJECT (hlsdemux, "Will switch from variant '%s' to '%s'",
hlsdemux->current_variant->name, variant->name);
if (hlsdemux->pending_variant) {
GST_ERROR_OBJECT (hlsdemux, "Already waiting for pending variant '%s'",
hlsdemux->pending_variant->name);
gst_hls_variant_stream_unref (hlsdemux->pending_variant);
}
hlsdemux->pending_variant = gst_hls_variant_stream_ref (variant);
} else {
GST_DEBUG_OBJECT (hlsdemux, "Setting variant '%s'", variant->name);
hlsdemux->current_variant = gst_hls_variant_stream_ref (variant);
}
}
static gboolean
gst_hls_demux_process_manifest (GstAdaptiveDemux * demux, GstBuffer * buf)
{
GstHLSVariantStream *variant;
GstHLSDemux *hlsdemux = GST_HLS_DEMUX_CAST (demux);
gchar *playlist = NULL;
gboolean ret;
GstHLSMediaPlaylist *simple_media_playlist = NULL;
GST_INFO_OBJECT (demux, "Initial playlist location: %s (base uri: %s)",
demux->manifest_uri, demux->manifest_base_uri);
playlist = gst_hls_buf_to_utf8_text (buf);
if (playlist == NULL) {
GST_WARNING_OBJECT (demux, "Error validating initial playlist");
return FALSE;
}
if (hlsdemux->master) {
gst_hls_master_playlist_unref (hlsdemux->master);
hlsdemux->master = NULL;
}
hlsdemux->master = gst_hls_master_playlist_new_from_data (playlist,
gst_adaptive_demux_get_manifest_ref_uri (demux));
if (hlsdemux->master == NULL) {
/* In most cases, this will happen if we set a wrong url in the
* source element and we have received the 404 HTML response instead of
* the playlist */
GST_ELEMENT_ERROR (demux, STREAM, DECODE, ("Invalid playlist."),
("Could not parse playlist. Check if the URL is correct."));
return FALSE;
}
if (hlsdemux->master->is_simple) {
simple_media_playlist =
gst_hls_media_playlist_parse (playlist,
gst_adaptive_demux_get_manifest_ref_uri (demux), NULL);
}
/* select the initial variant stream */
if (demux->connection_speed == 0) {
variant = hlsdemux->master->default_variant;
} else if (hlsdemux->start_bitrate > 0) {
variant =
gst_hls_master_playlist_get_variant_for_bitrate (hlsdemux->master,
NULL, hlsdemux->start_bitrate, demux->min_bitrate);
} else {
variant =
gst_hls_master_playlist_get_variant_for_bitrate (hlsdemux->master,
NULL, demux->connection_speed, demux->min_bitrate);
}
if (variant) {
GST_INFO_OBJECT (hlsdemux,
"Manifest processed, initial variant selected : `%s`", variant->name);
gst_hls_demux_set_current_variant (hlsdemux, variant); // FIXME: inline?
}
GST_DEBUG_OBJECT (hlsdemux, "Manifest handled, now setting up streams");
ret = gst_hls_demux_setup_streams (demux);
if (simple_media_playlist) {
hlsdemux->main_stream->playlist = simple_media_playlist;
hlsdemux->main_stream->current_segment =
gst_hls_media_playlist_get_starting_segment (simple_media_playlist);
setup_initial_playlist (hlsdemux, simple_media_playlist);
gst_hls_update_time_mappings (hlsdemux, simple_media_playlist);
gst_hls_media_playlist_dump (simple_media_playlist);
}
/* get the selected media playlist (unless the initial list was one already) */
if (!hlsdemux->master->is_simple) {
GError *err = NULL;
if (gst_hls_demux_update_playlist (hlsdemux, FALSE, &err) != GST_FLOW_OK) {
GST_ELEMENT_ERROR_FROM_ERROR (demux, "Could not fetch media playlist",
err);
return FALSE;
}
}
return ret;
}
static GstClockTime
gst_hls_demux_get_duration (GstAdaptiveDemux * demux)
{
GstHLSDemux *hlsdemux = GST_HLS_DEMUX_CAST (demux);
GstClockTime duration = GST_CLOCK_TIME_NONE;
if (hlsdemux->main_stream)
duration =
gst_hls_media_playlist_get_duration (hlsdemux->main_stream->playlist);
return duration;
}
static gboolean
gst_hls_demux_is_live (GstAdaptiveDemux * demux)
{
GstHLSDemux *hlsdemux = GST_HLS_DEMUX_CAST (demux);
gboolean is_live = FALSE;
if (hlsdemux->main_stream)
is_live = gst_hls_media_playlist_is_live (hlsdemux->main_stream->playlist);
return is_live;
}
static const GstHLSKey *
gst_hls_demux_get_key (GstHLSDemux * demux, const gchar * key_url,
const gchar * referer, gboolean allow_cache)
{
GstAdaptiveDemux *adaptive_demux = GST_ADAPTIVE_DEMUX (demux);
DownloadRequest *key_request;
DownloadFlags dl_flags = DOWNLOAD_FLAG_NONE;
GstBuffer *key_buffer;
GstHLSKey *key;
GError *err = NULL;
GST_LOG_OBJECT (demux, "Looking up key for key url %s", key_url);
g_mutex_lock (&demux->keys_lock);
key = g_hash_table_lookup (demux->keys, key_url);
if (key != NULL) {
GST_LOG_OBJECT (demux, "Found key for key url %s in key cache", key_url);
goto out;
}
GST_INFO_OBJECT (demux, "Fetching key %s", key_url);
if (!allow_cache)
dl_flags |= DOWNLOAD_FLAG_FORCE_REFRESH;
key_request =
downloadhelper_fetch_uri (adaptive_demux->download_helper,
key_url, referer, dl_flags, &err);
if (key_request == NULL) {
GST_WARNING_OBJECT (demux, "Failed to download key to decrypt data: %s",
err ? err->message : "error");
g_clear_error (&err);
goto out;
}
key_buffer = download_request_take_buffer (key_request);
download_request_unref (key_request);
key = g_new0 (GstHLSKey, 1);
if (gst_buffer_extract (key_buffer, 0, key->data, 16) < 16)
GST_WARNING_OBJECT (demux, "Download decryption key is too short!");
g_hash_table_insert (demux->keys, g_strdup (key_url), key);
gst_buffer_unref (key_buffer);
out:
g_mutex_unlock (&demux->keys_lock);
if (key != NULL)
GST_MEMDUMP_OBJECT (demux, "Key", key->data, 16);
return key;
}
static gboolean
gst_hls_demux_start_fragment (GstAdaptiveDemux * demux,
GstAdaptiveDemux2Stream * stream)
{
GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (stream);
GstHLSDemux *hlsdemux = GST_HLS_DEMUX_CAST (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 (demux, STREAM, DECRYPT_NOKEY,
("Couldn't retrieve key for decryption"), (NULL));
GST_WARNING_OBJECT (demux, "Failed to decrypt data");
return FALSE;
}
decrypt_start_failed:
{
GST_ELEMENT_ERROR (demux, STREAM, DECRYPT, ("Failed to start decrypt"),
("Couldn't set key and IV or plugin was built without crypto library"));
return FALSE;
}
}
static void
gst_hls_demux_start_rendition_streams (GstHLSDemux * hlsdemux)
{
GstAdaptiveDemux *demux = (GstAdaptiveDemux *) hlsdemux;
GList *tmp;
for (tmp = demux->input_period->streams; tmp; tmp = tmp->next) {
GstAdaptiveDemux2Stream *stream = (GstAdaptiveDemux2Stream *) tmp->data;
GstHLSDemuxStream *hls_stream = (GstHLSDemuxStream *) stream;
if (!hls_stream->is_variant
&& gst_adaptive_demux2_stream_is_selected (stream))
gst_adaptive_demux2_stream_start (stream);
}
}
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
*
* Returns TRUE if we are done with typefinding */
static gboolean
gst_hls_demux_typefind_stream (GstHLSDemux * hlsdemux,
GstAdaptiveDemux2Stream * stream, GstBuffer * 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;
if (hls_stream->pending_typefind_buffer)
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;
*ret = GST_FLOW_OK;
}
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));
*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 */
*ret = GST_FLOW_OK;
return TRUE;
}
static GstHLSTimeMap *
time_map_in_list (GList * list, gint64 dsn)
{
GList *iter;
for (iter = list; iter; iter = iter->next) {
GstHLSTimeMap *map = iter->data;
if (map->dsn == dsn)
return map;
}
return NULL;
}
GstHLSTimeMap *
gst_hls_find_time_map (GstHLSDemux * demux, gint64 dsn)
{
return time_map_in_list (demux->mappings, dsn);
}
/* 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_handle_internal_time (GstHLSDemux * demux,
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;
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));
map = gst_hls_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;
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) {
/* Update the value */
GST_DEBUG_OBJECT (hls_stream,
"Updating current stream time to %" GST_STIME_FORMAT,
GST_STIME_ARGS (real_stream_time));
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) > (hls_stream->current_segment->duration / 2)) {
GstAdaptiveDemux2Stream *stream = (GstAdaptiveDemux2Stream *) hls_stream;
GstM3U8MediaSegment *actual_segment;
/* We are at the wrong segment, try to figure out the *actual* segment */
GST_DEBUG_OBJECT (hls_stream,
"Trying to seek to the correct segment for %" GST_STIME_FORMAT,
GST_STIME_ARGS (current_stream_time));
actual_segment =
gst_hls_media_playlist_seek (hls_stream->playlist, TRUE,
GST_SEEK_FLAG_SNAP_NEAREST, current_stream_time);
if (actual_segment) {
GST_DEBUG_OBJECT (hls_stream, "Synced to position %" GST_STIME_FORMAT,
GST_STIME_ARGS (actual_segment->stream_time));
gst_m3u8_media_segment_unref (hls_stream->current_segment);
hls_stream->current_segment = actual_segment;
/* 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_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_handle_buffer (GstAdaptiveDemux * demux,
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 (demux);
GstFlowReturn ret = GST_FLOW_OK;
/* 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");
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, hls_stream->current_segment->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) {
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;
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:
GST_DEBUG_OBJECT (stream, "Returning %s", gst_flow_get_name (ret));
return ret;
}
static GstFlowReturn
gst_hls_demux_finish_fragment (GstAdaptiveDemux * demux,
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 fragment uri:%s",
hls_stream->current_segment->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_handle_buffer (demux, 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_handle_buffer (demux, 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_handle_buffer (demux, 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) */
return GST_FLOW_OK;
}
if (ret == GST_FLOW_OK || ret == GST_FLOW_NOT_LINKED) {
/* 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 */
stream->current_position = hls_stream->current_segment->stream_time;
return gst_adaptive_demux2_stream_advance_fragment (demux, stream,
hls_stream->current_segment->duration);
}
return ret;
}
static GstFlowReturn
gst_hls_demux_data_received (GstAdaptiveDemux * demux,
GstAdaptiveDemux2Stream * stream, GstBuffer * buffer)
{
GstHLSDemuxStream *hls_stream = GST_HLS_DEMUX_STREAM_CAST (stream);
GstHLSDemux *hlsdemux = GST_HLS_DEMUX_CAST (demux);
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 (demux, 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;
}
return gst_hls_demux_handle_buffer (demux, 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->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->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 ?");
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_advance_fragment (GstAdaptiveDemux2Stream * stream)
{
GstHLSDemuxStream *hlsdemux_stream = GST_HLS_DEMUX_STREAM_CAST (stream);
GstHLSDemux *hlsdemux = (GstHLSDemux *) stream->demux;
GstM3U8MediaSegment *new_segment = NULL;
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),
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;
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),
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;
return GST_FLOW_OK;
}
return GST_FLOW_EOS;
}
static GstHLSMediaPlaylist *
download_media_playlist (GstHLSDemux * demux, gchar * uri, GError ** err,
GstHLSMediaPlaylist * current)
{
GstAdaptiveDemux *adaptive_demux;
const gchar *main_uri;
DownloadRequest *download;
GstBuffer *buf;
gchar *playlist_data;
GstHLSMediaPlaylist *playlist = NULL;
gchar *base_uri;
gboolean playlist_uri_change = FALSE;
adaptive_demux = GST_ADAPTIVE_DEMUX (demux);
main_uri = gst_adaptive_demux_get_manifest_ref_uri (adaptive_demux);
/* If there's no previous playlist, or the URI changed this
* is not a refresh/update but a switch to a new playlist */
playlist_uri_change = (current == NULL || g_strcmp0 (uri, current->uri) != 0);
if (!playlist_uri_change) {
GST_LOG_OBJECT (demux, "Updating the playlist");
}
download =
downloadhelper_fetch_uri (adaptive_demux->download_helper,
uri, main_uri, DOWNLOAD_FLAG_COMPRESS | DOWNLOAD_FLAG_FORCE_REFRESH, err);
if (download == NULL)
return NULL;
/* Set the base URI of the playlist to the redirect target if any */
if (download->redirect_permanent && download->redirect_uri) {
uri = g_strdup (download->redirect_uri);
base_uri = NULL;
} else {
uri = g_strdup (download->uri);
base_uri = g_strdup (download->redirect_uri);
}
if (download->state == DOWNLOAD_REQUEST_STATE_ERROR) {
GST_WARNING_OBJECT (demux,
"Couldn't get the playlist, got HTTP status code %d",
download->status_code);
download_request_unref (download);
if (err)
g_set_error (err, GST_STREAM_ERROR, GST_STREAM_ERROR_WRONG_TYPE,
"Couldn't download the playlist");
goto out;
}
buf = download_request_take_buffer (download);
download_request_unref (download);
/* there should be a buf if there wasn't an error (handled above) */
g_assert (buf);
playlist_data = gst_hls_buf_to_utf8_text (buf);
gst_buffer_unref (buf);
if (playlist_data == NULL) {
GST_WARNING_OBJECT (demux, "Couldn't validate playlist encoding");
if (err)
g_set_error (err, GST_STREAM_ERROR, GST_STREAM_ERROR_WRONG_TYPE,
"Couldn't validate playlist encoding");
goto out;
}
if (!playlist_uri_change && current
&& gst_hls_media_playlist_has_same_data (current, playlist_data)) {
GST_DEBUG_OBJECT (demux, "Same playlist data");
playlist = gst_hls_media_playlist_ref (current);
playlist->reloaded = TRUE;
g_free (playlist_data);
} else {
playlist = gst_hls_media_playlist_parse (playlist_data, uri, base_uri);
if (!playlist) {
GST_WARNING_OBJECT (demux, "Couldn't parse playlist");
if (err)
g_set_error (err, GST_STREAM_ERROR, GST_STREAM_ERROR_FAILED,
"Couldn't parse playlist");
}
}
out:
g_free (uri);
g_free (base_uri);
return playlist;
}
static GstHLSTimeMap *
gst_hls_time_map_new (void)
{
GstHLSTimeMap *map = g_new0 (GstHLSTimeMap, 1);
map->stream_time = GST_CLOCK_TIME_NONE;
map->internal_time = GST_CLOCK_TIME_NONE;
return map;
}
static void
gst_hls_demux_add_time_mapping (GstHLSDemux * demux, gint64 dsn,
GstClockTimeDiff stream_time, GDateTime * pdt)
{
#ifndef GST_DISABLE_GST_DEBUG
gchar *datestring = NULL;
#endif
GstHLSTimeMap *map;
GList *tmp;
g_assert (stream_time >= 0);
/* Check if we don't already have a mapping for the given dsn */
for (tmp = demux->mappings; tmp; tmp = tmp->next) {
GstHLSTimeMap *map = tmp->data;
if (map->dsn == dsn) {
#ifndef GST_DISABLE_GST_DEBUG
if (map->pdt)
datestring = g_date_time_format_iso8601 (map->pdt);
GST_DEBUG_OBJECT (demux,
"Already have mapping, dsn:%" G_GINT64_FORMAT " stream_time:%"
GST_TIME_FORMAT " internal_time:%" GST_TIME_FORMAT " pdt:%s",
map->dsn, GST_TIME_ARGS (map->stream_time),
GST_TIME_ARGS (map->internal_time), datestring);
g_free (datestring);
#endif
return;
}
}
#ifndef GST_DISABLE_GST_DEBUG
if (pdt)
datestring = g_date_time_format_iso8601 (pdt);
GST_DEBUG_OBJECT (demux,
"New mapping, dsn:%" G_GINT64_FORMAT " stream_time:%" GST_TIME_FORMAT
" pdt:%s", dsn, GST_TIME_ARGS (stream_time), datestring);
g_free (datestring);
#endif
map = gst_hls_time_map_new ();
map->dsn = dsn;
map->stream_time = stream_time;
if (pdt)
map->pdt = g_date_time_ref (pdt);
demux->mappings = g_list_append (demux->mappings, map);
}
/* Remove any time mapping which isn't currently used by any stream playlist */
static void
gst_hls_prune_time_mappings (GstHLSDemux * hlsdemux)
{
GstAdaptiveDemux *demux = (GstAdaptiveDemux *) hlsdemux;
GList *active = NULL;
GList *iterstream;
for (iterstream = demux->input_period->streams; iterstream;
iterstream = iterstream->next) {
GstAdaptiveDemux2Stream *stream = iterstream->data;
GstHLSDemuxStream *hls_stream = (GstHLSDemuxStream *) stream;
gint64 dsn = G_MAXINT64;
guint idx, len;
if (!hls_stream->playlist)
continue;
len = hls_stream->playlist->segments->len;
for (idx = 0; idx < len; idx++) {
GstM3U8MediaSegment *segment =
g_ptr_array_index (hls_stream->playlist->segments, idx);
if (dsn == G_MAXINT64 || segment->discont_sequence != dsn) {
dsn = segment->discont_sequence;
if (!time_map_in_list (active, dsn)) {
GstHLSTimeMap *map = gst_hls_find_time_map (hlsdemux, dsn);
if (map) {
GST_DEBUG_OBJECT (demux,
"Keeping active time map dsn:%" G_GINT64_FORMAT, map->dsn);
/* Move active dsn to active list */
hlsdemux->mappings = g_list_remove (hlsdemux->mappings, map);
active = g_list_append (active, map);
}
}
}
}
}
g_list_free_full (hlsdemux->mappings, g_free);
hlsdemux->mappings = active;
}
/* Go over the DSN from the playlist and add any missing time mapping */
static void
gst_hls_update_time_mappings (GstHLSDemux * demux,
GstHLSMediaPlaylist * playlist)
{
guint idx, len = playlist->segments->len;
gint64 dsn = G_MAXINT64;
for (idx = 0; idx < len; idx++) {
GstM3U8MediaSegment *segment = g_ptr_array_index (playlist->segments, idx);
if (dsn == G_MAXINT64 || segment->discont_sequence != dsn) {
dsn = segment->discont_sequence;
if (!gst_hls_find_time_map (demux, segment->discont_sequence))
gst_hls_demux_add_time_mapping (demux, segment->discont_sequence,
segment->stream_time, segment->datetime);
}
}
}
static void
setup_initial_playlist (GstHLSDemux * demux, GstHLSMediaPlaylist * playlist)
{
guint idx, len = playlist->segments->len;
GstM3U8MediaSegment *segment;
GstClockTimeDiff pos = 0;
GST_DEBUG_OBJECT (demux,
"Setting up initial variant segment and time mapping");
/* This is the initial variant playlist. We will use it to base all our timing
* from. */
for (idx = 0; idx < len; idx++) {
segment = g_ptr_array_index (playlist->segments, idx);
segment->stream_time = pos;
pos += segment->duration;
}
}
/* Reset hlsdemux in case of live synchronization loss (i.e. when a media
* playlist update doesn't match at all with the previous one) */
static void
gst_hls_demux_reset_for_lost_sync (GstHLSDemux * hlsdemux)
{
GstAdaptiveDemux *demux = (GstAdaptiveDemux *) hlsdemux;
GList *iter;
GST_DEBUG_OBJECT (hlsdemux, "Resetting for lost sync");
for (iter = demux->input_period->streams; iter; iter = iter->next) {
GstHLSDemuxStream *hls_stream = iter->data;
GstAdaptiveDemux2Stream *stream = (GstAdaptiveDemux2Stream *) hls_stream;
if (hls_stream->current_segment)
gst_m3u8_media_segment_unref (hls_stream->current_segment);
hls_stream->current_segment = NULL;
if (hls_stream->is_variant) {
GstHLSTimeMap *map;
/* Resynchronize the variant stream */
g_assert (stream->current_position != GST_CLOCK_STIME_NONE);
hls_stream->current_segment =
gst_hls_media_playlist_get_starting_segment (hls_stream->playlist);
hls_stream->current_segment->stream_time = stream->current_position;
gst_hls_media_playlist_recalculate_stream_time (hls_stream->playlist,
hls_stream->current_segment);
GST_DEBUG_OBJECT (stream,
"Resynced variant playlist to %" GST_STIME_FORMAT,
GST_STIME_ARGS (stream->current_position));
map =
gst_hls_find_time_map (hlsdemux,
hls_stream->current_segment->discont_sequence);
if (map)
map->internal_time = GST_CLOCK_TIME_NONE;
gst_hls_update_time_mappings (hlsdemux, hls_stream->playlist);
gst_hls_media_playlist_dump (hls_stream->playlist);
} else {
/* Force playlist update for the rendition streams, it will resync to the
* variant stream on the next round */
if (hls_stream->playlist)
gst_hls_media_playlist_unref (hls_stream->playlist);
hls_stream->playlist = NULL;
hls_stream->playlist_fetched = FALSE;
}
}
}
static GstFlowReturn
gst_hls_demux_stream_update_media_playlist (GstHLSDemux * demux,
GstHLSDemuxStream * stream, gchar ** uri, GError ** err)
{
GstHLSMediaPlaylist *new_playlist;
GST_DEBUG_OBJECT (stream, "Updating %s", *uri);
new_playlist = download_media_playlist (demux, *uri, err, stream->playlist);
if (new_playlist == NULL) {
GST_WARNING_OBJECT (stream, "Could not get playlist '%s'", *uri);
return GST_FLOW_ERROR;
}
/* 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);
}
/* 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),
stream->current_segment->uri);
/* Use best-effort techniques to find the correponding 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);
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->playlist) {
gst_hls_media_playlist_unref (stream->playlist);
stream->playlist = new_playlist;
} else {
if (stream->is_variant) {
GST_DEBUG_OBJECT (stream, "Setting up initial playlist");
setup_initial_playlist (demux, new_playlist);
}
stream->playlist = new_playlist;
}
if (stream->is_variant) {
/* Update time mappings. We only use the variant stream for collecting
* mappings since it is the reference on which rendition stream timing will
* be based. */
gst_hls_update_time_mappings (demux, stream->playlist);
}
gst_hls_media_playlist_dump (stream->playlist);
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),
stream->current_segment->uri);
} else {
GST_DEBUG_OBJECT (stream, "No current segment selected");
}
GST_DEBUG_OBJECT (stream, "done");
return GST_FLOW_OK;
/* 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;
gst_hls_demux_reset_for_lost_sync (demux);
return GST_ADAPTIVE_DEMUX_FLOW_LOST_SYNC;
}
}
static GstFlowReturn
gst_hls_demux_stream_update_rendition_playlist (GstHLSDemux * demux,
GstHLSDemuxStream * stream)
{
GstFlowReturn ret = GST_FLOW_OK;
GstHLSRenditionStream *target_rendition =
stream->pending_rendition ? stream->
pending_rendition : stream->current_rendition;
ret = gst_hls_demux_stream_update_media_playlist (demux, stream,
&target_rendition->uri, NULL);
if (ret != GST_FLOW_OK)
return ret;
if (stream->pending_rendition) {
gst_hls_rendition_stream_unref (stream->current_rendition);
/* Stealing ref */
stream->current_rendition = stream->pending_rendition;
stream->pending_rendition = NULL;
}
stream->playlist_fetched = TRUE;
return ret;
}
static GstFlowReturn
gst_hls_demux_stream_update_variant_playlist (GstHLSDemux * demux,
GstHLSDemuxStream * stream, GError ** err)
{
GstFlowReturn ret = GST_FLOW_OK;
GstHLSVariantStream *target_variant =
demux->pending_variant ? demux->pending_variant : demux->current_variant;
ret = gst_hls_demux_stream_update_media_playlist (demux, stream,
&target_variant->uri, err);
if (ret != GST_FLOW_OK)
return ret;
if (demux->pending_variant) {
gst_hls_variant_stream_unref (demux->current_variant);
/* Stealing ref */
demux->current_variant = demux->pending_variant;
demux->pending_variant = NULL;
}
stream->playlist_fetched = TRUE;
return ret;
}
static GstFlowReturn
gst_hls_demux_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;
gboolean discont;
/* If the rendition playlist needs to be updated, do it now */
if (!hlsdemux_stream->is_variant && !hlsdemux_stream->playlist_fetched) {
ret = gst_hls_demux_stream_update_rendition_playlist (hlsdemux,
hlsdemux_stream);
if (ret != GST_FLOW_OK)
return ret;
}
GST_DEBUG_OBJECT (stream,
"Updating fragment information, current_position:%" GST_TIME_FORMAT,
GST_TIME_ARGS (stream->current_position));
/* 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) {
GST_DEBUG_OBJECT (stream, "Setting up initial segment");
hlsdemux_stream->current_segment =
gst_hls_media_playlist_get_starting_segment
(hlsdemux_stream->playlist);
} 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));
hlsdemux_stream->current_segment =
gst_hls_media_playlist_seek (hlsdemux_stream->playlist, TRUE,
GST_SEEK_FLAG_SNAP_NEAREST, stream->current_position);
if (hlsdemux_stream->current_segment == NULL) {
GST_INFO_OBJECT (stream, "At the end of the current media playlist");
return GST_FLOW_EOS;
}
/* Update time mapping. If it already exists it will be ignored */
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;
GST_DEBUG_OBJECT (stream, "Current segment stream_time %" GST_STIME_FORMAT,
GST_STIME_ARGS (file->stream_time));
discont = file->discont || stream->discont;
if (GST_ADAPTIVE_DEMUX2_STREAM_NEED_HEADER (stream) && file->init_file) {
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;
}
}
/* set up our source for download */
if (hlsdemux_stream->reset_pts || discont || demux->segment.rate < 0.0) {
stream->fragment.stream_time = file->stream_time;
} else {
stream->fragment.stream_time = GST_CLOCK_STIME_NONE;
}
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));
g_free (stream->fragment.uri);
stream->fragment.uri = g_strdup (file->uri);
GST_DEBUG_OBJECT (stream, "Stream URI now %s", file->uri);
stream->fragment.range_start = file->offset;
if (file->size != -1)
stream->fragment.range_end = file->offset + file->size - 1;
else
stream->fragment.range_end = -1;
stream->fragment.duration = file->duration;
if (discont)
stream->discont = TRUE;
return ret;
}
static gboolean
gst_hls_demux_stream_can_start (GstAdaptiveDemux * demux,
GstAdaptiveDemux2Stream * stream)
{
GstHLSDemux *hlsdemux = (GstHLSDemux *) demux;
GstHLSDemuxStream *hls_stream = (GstHLSDemuxStream *) stream;
GList *tmp;
GST_DEBUG_OBJECT (demux, "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;
}
/* Returns TRUE if the rendition stream switched group-id */
static gboolean
gst_hls_demux_update_rendition_stream (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);
hls_stream->playlist_fetched = FALSE;
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);
return TRUE;
}
static gboolean
gst_hls_demux_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;
if (hls_stream->is_variant) {
gdouble play_rate = gst_adaptive_demux_play_rate (demux);
gboolean changed = FALSE;
/* Handle variant streams */
GST_DEBUG_OBJECT (hlsdemux,
"Checking playlist change for main variant stream");
gst_hls_demux_change_playlist (hlsdemux, bitrate / MAX (1.0,
ABS (play_rate)), &changed);
GST_DEBUG_OBJECT (hlsdemux, "Returning changed: %d", changed);
return changed;
}
/* Handle rendition streams */
return gst_hls_demux_update_rendition_stream (hlsdemux, hls_stream, NULL);
}
static void
gst_hls_demux_reset (GstAdaptiveDemux * ademux)
{
GstHLSDemux *demux = GST_HLS_DEMUX_CAST (ademux);
GST_DEBUG_OBJECT (demux, "resetting");
if (demux->master) {
gst_hls_master_playlist_unref (demux->master);
demux->master = NULL;
}
if (demux->current_variant != NULL) {
gst_hls_variant_stream_unref (demux->current_variant);
demux->current_variant = NULL;
}
if (demux->pending_variant != NULL) {
gst_hls_variant_stream_unref (demux->pending_variant);
demux->pending_variant = NULL;
}
g_list_free_full (demux->mappings, g_free);
demux->mappings = NULL;
gst_hls_demux_clear_all_pending_data (demux);
}
/*
* update: TRUE only when requested from parent class (via
* ::demux_update_manifest() or ::change_playlist() ).
*/
static GstFlowReturn
gst_hls_demux_update_playlist (GstHLSDemux * demux, gboolean update,
GError ** err)
{
GstFlowReturn ret = GST_FLOW_OK;
GstAdaptiveDemux *adaptive_demux = GST_ADAPTIVE_DEMUX (demux);
GST_DEBUG_OBJECT (demux, "update:%d", update);
/* Download and update the appropriate variant playlist (pending if any, else
* current) */
ret = gst_hls_demux_stream_update_variant_playlist (demux, demux->main_stream,
err);
if (ret != GST_FLOW_OK)
return ret;
if (update && gst_hls_demux_is_live (adaptive_demux)) {
GList *tmp;
GST_DEBUG_OBJECT (demux,
"LIVE, Marking rendition streams to be updated next");
/* We're live, instruct all rendition medias to be updated next */
for (tmp = adaptive_demux->input_period->streams; tmp; tmp = tmp->next) {
GstHLSDemuxStream *hls_stream = tmp->data;
if (!hls_stream->is_variant)
hls_stream->playlist_fetched = FALSE;
}
}
return GST_FLOW_OK;
}
static gboolean
gst_hls_demux_change_playlist (GstHLSDemux * demux, guint max_bitrate,
gboolean * changed)
{
GstHLSVariantStream *lowest_variant, *lowest_ivariant;
GstHLSVariantStream *previous_variant, *new_variant;
gint old_bandwidth, new_bandwidth;
GstAdaptiveDemux *adaptive_demux = GST_ADAPTIVE_DEMUX_CAST (demux);
GstAdaptiveDemux2Stream *stream;
g_return_val_if_fail (demux->main_stream != NULL, FALSE);
stream = (GstAdaptiveDemux2Stream *) demux->main_stream;
/* Make sure we keep a reference in case we need to switch back */
previous_variant = gst_hls_variant_stream_ref (demux->current_variant);
new_variant =
gst_hls_master_playlist_get_variant_for_bitrate (demux->master,
demux->current_variant, max_bitrate, adaptive_demux->min_bitrate);
retry_failover_protection:
old_bandwidth = previous_variant->bandwidth;
new_bandwidth = new_variant->bandwidth;
/* Don't do anything else if the playlist is the same */
if (new_bandwidth == old_bandwidth) {
gst_hls_variant_stream_unref (previous_variant);
return TRUE;
}
gst_hls_demux_set_current_variant (demux, new_variant);
GST_INFO_OBJECT (demux, "Client was on %dbps, max allowed is %dbps, switching"
" to bitrate %dbps", old_bandwidth, max_bitrate, new_bandwidth);
if (gst_hls_demux_update_playlist (demux, TRUE, NULL) == GST_FLOW_OK) {
const gchar *main_uri;
gchar *uri = new_variant->uri;
main_uri = gst_adaptive_demux_get_manifest_ref_uri (adaptive_demux);
gst_element_post_message (GST_ELEMENT_CAST (demux),
gst_message_new_element (GST_OBJECT_CAST (demux),
gst_structure_new (GST_ADAPTIVE_DEMUX_STATISTICS_MESSAGE_NAME,
"manifest-uri", G_TYPE_STRING,
main_uri, "uri", G_TYPE_STRING,
uri, "bitrate", G_TYPE_INT, new_bandwidth, NULL)));
if (changed)
*changed = TRUE;
stream->discont = TRUE;
} else if (gst_adaptive_demux2_is_running (GST_ADAPTIVE_DEMUX_CAST (demux))) {
GstHLSVariantStream *failover_variant = NULL;
GList *failover;
GST_INFO_OBJECT (demux, "Unable to update playlist. Switching back");
/* we find variants by bitrate by going from highest to lowest, so it's
* possible that there's another variant with the same bitrate before the
* one selected which we can use as failover */
failover = g_list_find (demux->master->variants, new_variant);
if (failover != NULL)
failover = failover->prev;
if (failover != NULL)
failover_variant = failover->data;
if (failover_variant && new_bandwidth == failover_variant->bandwidth) {
new_variant = failover_variant;
goto retry_failover_protection;
}
gst_hls_demux_set_current_variant (demux, previous_variant);
/* Try a lower bitrate (or stop if we just tried the lowest) */
if (previous_variant->iframe) {
lowest_ivariant = demux->master->iframe_variants->data;
if (new_bandwidth == lowest_ivariant->bandwidth)
return FALSE;
} else {
lowest_variant = demux->master->variants->data;
if (new_bandwidth == lowest_variant->bandwidth)
return FALSE;
}
return gst_hls_demux_change_playlist (demux, new_bandwidth - 1, changed);
}
gst_hls_variant_stream_unref (previous_variant);
return TRUE;
}
#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 gint64
gst_hls_demux_get_manifest_update_interval (GstAdaptiveDemux * demux)
{
GstHLSDemux *hlsdemux = GST_HLS_DEMUX_CAST (demux);
GstClockTime target_duration = 5 * GST_SECOND;
if (hlsdemux->main_stream && hlsdemux->main_stream->playlist) {
GstHLSMediaPlaylist *playlist = hlsdemux->main_stream->playlist;
if (playlist->version > 5) {
target_duration = hlsdemux->main_stream->playlist->targetduration;
} else if (playlist->segments->len) {
GstM3U8MediaSegment *last_seg =
g_ptr_array_index (playlist->segments, playlist->segments->len - 1);
target_duration = last_seg->duration;
}
if (playlist->reloaded && target_duration > (playlist->targetduration / 2)) {
GST_DEBUG_OBJECT (demux,
"Playlist didn't change previously, returning lower update interval");
target_duration /= 2;
}
}
GST_DEBUG_OBJECT (demux, "Returning update interval of %" GST_TIME_FORMAT,
GST_TIME_ARGS (target_duration));
return gst_util_uint64_scale (target_duration, G_USEC_PER_SEC, GST_SECOND);
}
static GstClockTime
gst_hls_demux_get_presentation_offset (GstAdaptiveDemux * demux,
GstAdaptiveDemux2Stream * stream)
{
GstHLSDemux *hlsdemux = (GstHLSDemux *) 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;
}
static gboolean
gst_hls_demux_get_live_seek_range (GstAdaptiveDemux * demux, gint64 * start,
gint64 * stop)
{
GstHLSDemux *hlsdemux = GST_HLS_DEMUX_CAST (demux);
gboolean ret = FALSE;
if (hlsdemux->main_stream && hlsdemux->main_stream->playlist)
ret =
gst_hls_media_playlist_get_seek_range (hlsdemux->main_stream->playlist,
start, stop);
return ret;
}
static gboolean
hlsdemux2_element_init (GstPlugin * plugin)
{
gboolean ret = TRUE;
GST_DEBUG_CATEGORY_INIT (gst_hls_demux2_debug, "hlsdemux2", 0,
"hlsdemux2 element");
if (!adaptivedemux2_base_element_init (plugin))
return TRUE;
ret = gst_element_register (plugin, "hlsdemux2",
GST_RANK_PRIMARY + 1, GST_TYPE_HLS_DEMUX2);
return ret;
}