gstreamer/ext/smoothstreaming/gstmssmanifest.c
Thiago Santos 628461b4d1 mssdemux: fix segfault when trying to access codec_data
Check if codec_data was obtained before trying to use it to
prevent a segfault
2013-12-18 18:52:19 -03:00

1227 lines
32 KiB
C

/* GStreamer
* Copyright (C) 2012 Smart TV Alliance
* Author: Thiago Sousa Santos <thiago.sousa.santos@collabora.com>, Collabora Ltd.
*
* gstmssmanifest.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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <glib.h>
#include <string.h>
#include <stdio.h>
#include <ctype.h>
#include <libxml/parser.h>
#include <libxml/tree.h>
/* for parsing h264 codec data */
#include <gst/codecparsers/gsth264parser.h>
#include "gstmssmanifest.h"
GST_DEBUG_CATEGORY_EXTERN (mssdemux_debug);
#define GST_CAT_DEFAULT mssdemux_debug
#define DEFAULT_TIMESCALE 10000000
#define MSS_NODE_STREAM_FRAGMENT "c"
#define MSS_NODE_STREAM_QUALITY "QualityLevel"
#define MSS_PROP_BITRATE "Bitrate"
#define MSS_PROP_DURATION "d"
#define MSS_PROP_LANGUAGE "Language"
#define MSS_PROP_NUMBER "n"
#define MSS_PROP_STREAM_DURATION "Duration"
#define MSS_PROP_TIME "t"
#define MSS_PROP_TIMESCALE "TimeScale"
#define MSS_PROP_URL "Url"
typedef struct _GstMssStreamFragment
{
guint number;
guint64 time;
guint64 duration;
} GstMssStreamFragment;
typedef struct _GstMssStreamQuality
{
xmlNodePtr xmlnode;
gchar *bitrate_str;
guint64 bitrate;
} GstMssStreamQuality;
struct _GstMssStream
{
xmlNodePtr xmlnode;
gboolean active; /* if the stream is currently being used */
gint selectedQualityIndex;
GList *fragments;
GList *qualities;
gchar *url;
gchar *lang;
GList *current_fragment;
GList *current_quality;
/* TODO move this to somewhere static */
GRegex *regex_bitrate;
GRegex *regex_position;
};
struct _GstMssManifest
{
xmlDocPtr xml;
xmlNodePtr xmlrootnode;
gboolean is_live;
GSList *streams;
};
static GstBuffer *gst_buffer_from_hex_string (const gchar * s);
static gboolean
node_has_type (xmlNodePtr node, const gchar * name)
{
return strcmp ((gchar *) node->name, name) == 0;
}
static GstMssStreamQuality *
gst_mss_stream_quality_new (xmlNodePtr node)
{
GstMssStreamQuality *q = g_slice_new (GstMssStreamQuality);
q->xmlnode = node;
q->bitrate_str = (gchar *) xmlGetProp (node, (xmlChar *) MSS_PROP_BITRATE);
if (q->bitrate_str != NULL)
q->bitrate = g_ascii_strtoull (q->bitrate_str, NULL, 10);
else
q->bitrate = 0;
return q;
}
static void
gst_mss_stream_quality_free (GstMssStreamQuality * quality)
{
g_return_if_fail (quality != NULL);
xmlFree (quality->bitrate_str);
g_slice_free (GstMssStreamQuality, quality);
}
static gint
compare_bitrate (GstMssStreamQuality * a, GstMssStreamQuality * b)
{
if (a->bitrate > b->bitrate)
return 1;
if (a->bitrate < b->bitrate)
return -1;
return 0;
}
static void
_gst_mss_stream_init (GstMssStream * stream, xmlNodePtr node)
{
xmlNodePtr iter;
GstMssStreamFragment *previous_fragment = NULL;
guint fragment_number = 0;
guint64 fragment_time_accum = 0;
stream->xmlnode = node;
/* get the base url path generator */
stream->url = (gchar *) xmlGetProp (node, (xmlChar *) MSS_PROP_URL);
stream->lang = (gchar *) xmlGetProp (node, (xmlChar *) MSS_PROP_LANGUAGE);
for (iter = node->children; iter; iter = iter->next) {
if (node_has_type (iter, MSS_NODE_STREAM_FRAGMENT)) {
gchar *duration_str;
gchar *time_str;
gchar *seqnum_str;
GstMssStreamFragment *fragment = g_new (GstMssStreamFragment, 1);
duration_str = (gchar *) xmlGetProp (iter, (xmlChar *) MSS_PROP_DURATION);
time_str = (gchar *) xmlGetProp (iter, (xmlChar *) MSS_PROP_TIME);
seqnum_str = (gchar *) xmlGetProp (iter, (xmlChar *) MSS_PROP_NUMBER);
/* use the node's seq number or use the previous + 1 */
if (seqnum_str) {
fragment->number = g_ascii_strtoull (seqnum_str, NULL, 10);
xmlFree (seqnum_str);
fragment_number = fragment->number;
} else {
fragment->number = fragment_number;
}
fragment_number = fragment->number + 1;
if (time_str) {
fragment->time = g_ascii_strtoull (time_str, NULL, 10);
xmlFree (time_str);
fragment_time_accum = fragment->time;
} else {
fragment->time = fragment_time_accum;
}
/* if we have a previous fragment, means we need to set its duration */
if (previous_fragment)
previous_fragment->duration = fragment->time - previous_fragment->time;
if (duration_str) {
fragment->duration = g_ascii_strtoull (duration_str, NULL, 10);
previous_fragment = NULL;
fragment_time_accum += fragment->duration;
xmlFree (duration_str);
} else {
/* store to set the duration at the next iteration */
previous_fragment = fragment;
}
/* we reverse it later */
stream->fragments = g_list_prepend (stream->fragments, fragment);
} else if (node_has_type (iter, MSS_NODE_STREAM_QUALITY)) {
GstMssStreamQuality *quality = gst_mss_stream_quality_new (iter);
stream->qualities = g_list_prepend (stream->qualities, quality);
} else {
/* TODO gst log this */
}
}
stream->fragments = g_list_reverse (stream->fragments);
/* order them from smaller to bigger based on bitrates */
stream->qualities =
g_list_sort (stream->qualities, (GCompareFunc) compare_bitrate);
stream->current_fragment = stream->fragments;
stream->current_quality = stream->qualities;
stream->regex_bitrate = g_regex_new ("\\{[Bb]itrate\\}", 0, 0, NULL);
stream->regex_position = g_regex_new ("\\{start[ _]time\\}", 0, 0, NULL);
}
GstMssManifest *
gst_mss_manifest_new (GstBuffer * data)
{
GstMssManifest *manifest;
xmlNodePtr root;
xmlNodePtr nodeiter;
gchar *live_str;
GstMapInfo mapinfo;
if (!gst_buffer_map (data, &mapinfo, GST_MAP_READ)) {
return NULL;
}
manifest = g_malloc0 (sizeof (GstMssManifest));
manifest->xml = xmlReadMemory ((const gchar *) mapinfo.data,
mapinfo.size, "manifest", NULL, 0);
root = manifest->xmlrootnode = xmlDocGetRootElement (manifest->xml);
live_str = (gchar *) xmlGetProp (root, (xmlChar *) "IsLive");
if (live_str) {
manifest->is_live = g_ascii_strcasecmp (live_str, "true") == 0;
xmlFree (live_str);
}
for (nodeiter = root->children; nodeiter; nodeiter = nodeiter->next) {
if (nodeiter->type == XML_ELEMENT_NODE
&& (strcmp ((const char *) nodeiter->name, "StreamIndex") == 0)) {
GstMssStream *stream = g_new0 (GstMssStream, 1);
manifest->streams = g_slist_append (manifest->streams, stream);
_gst_mss_stream_init (stream, nodeiter);
}
}
gst_buffer_unmap (data, &mapinfo);
return manifest;
}
static void
gst_mss_stream_free (GstMssStream * stream)
{
g_list_free_full (stream->fragments, g_free);
g_list_free_full (stream->qualities,
(GDestroyNotify) gst_mss_stream_quality_free);
xmlFree (stream->url);
xmlFree (stream->lang);
g_regex_unref (stream->regex_position);
g_regex_unref (stream->regex_bitrate);
g_free (stream);
}
void
gst_mss_manifest_free (GstMssManifest * manifest)
{
g_return_if_fail (manifest != NULL);
g_slist_free_full (manifest->streams, (GDestroyNotify) gst_mss_stream_free);
xmlFreeDoc (manifest->xml);
g_free (manifest);
}
GSList *
gst_mss_manifest_get_streams (GstMssManifest * manifest)
{
return manifest->streams;
}
GstMssStreamType
gst_mss_stream_get_type (GstMssStream * stream)
{
gchar *prop = (gchar *) xmlGetProp (stream->xmlnode, (xmlChar *) "Type");
GstMssStreamType ret = MSS_STREAM_TYPE_UNKNOWN;
if (prop == NULL)
return MSS_STREAM_TYPE_UNKNOWN;
if (strcmp (prop, "video") == 0) {
ret = MSS_STREAM_TYPE_VIDEO;
} else if (strcmp (prop, "audio") == 0) {
ret = MSS_STREAM_TYPE_AUDIO;
}
xmlFree (prop);
return ret;
}
static GstCaps *
_gst_mss_stream_video_caps_from_fourcc (gchar * fourcc)
{
if (!fourcc)
return NULL;
if (strcmp (fourcc, "H264") == 0 || strcmp (fourcc, "AVC1") == 0) {
return gst_caps_new_simple ("video/x-h264", "stream-format", G_TYPE_STRING,
"avc", NULL);
} else if (strcmp (fourcc, "WVC1") == 0) {
return gst_caps_new_simple ("video/x-wmv", "wmvversion", G_TYPE_INT, 3,
"format", G_TYPE_STRING, "WVC1", NULL);
}
return NULL;
}
static GstCaps *
_gst_mss_stream_audio_caps_from_fourcc (gchar * fourcc)
{
if (!fourcc)
return NULL;
if (strcmp (fourcc, "AACL") == 0) {
return gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 4,
NULL);
} else if (strcmp (fourcc, "WmaPro") == 0 || strcmp (fourcc, "WMAP") == 0) {
return gst_caps_new_simple ("audio/x-wma", "wmaversion", G_TYPE_INT, 3,
NULL);
}
return NULL;
}
static GstCaps *
_gst_mss_stream_audio_caps_from_audio_tag (gint audiotag)
{
switch (audiotag) {
case 83: /* MP3 */
return gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 1,
"layer", G_TYPE_INT, 3, NULL);
case 255: /* AAC */
return gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 4,
NULL);
default:
break;
}
return NULL;
}
/* copied and adapted from h264parse */
static GstBuffer *
_make_h264_codec_data (GstBuffer * sps, GstBuffer * pps)
{
GstBuffer *buf;
gint sps_size = 0, pps_size = 0, num_sps = 0, num_pps = 0;
guint8 profile_idc = 0, profile_comp = 0, level_idc = 0;
guint8 *data;
gint nl;
GstMapInfo spsinfo, ppsinfo, codecdatainfo;
if (gst_buffer_get_size (sps) < 4)
return NULL;
gst_buffer_map (sps, &spsinfo, GST_MAP_READ);
gst_buffer_map (pps, &ppsinfo, GST_MAP_READ);
sps_size += spsinfo.size + 2;
profile_idc = spsinfo.data[1];
profile_comp = spsinfo.data[2];
level_idc = spsinfo.data[3];
num_sps = 1;
pps_size += ppsinfo.size + 2;
num_pps = 1;
buf = gst_buffer_new_and_alloc (5 + 1 + sps_size + 1 + pps_size);
gst_buffer_map (buf, &codecdatainfo, GST_MAP_WRITE);
data = codecdatainfo.data;
nl = 4;
data[0] = 1; /* AVC Decoder Configuration Record ver. 1 */
data[1] = profile_idc; /* profile_idc */
data[2] = profile_comp; /* profile_compability */
data[3] = level_idc; /* level_idc */
data[4] = 0xfc | (nl - 1); /* nal_length_size_minus1 */
data[5] = 0xe0 | num_sps; /* number of SPSs */
data += 6;
GST_WRITE_UINT16_BE (data, spsinfo.size);
memcpy (data + 2, spsinfo.data, spsinfo.size);
data += 2 + spsinfo.size;
data[0] = num_pps;
data++;
GST_WRITE_UINT16_BE (data, ppsinfo.size);
memcpy (data + 2, ppsinfo.data, ppsinfo.size);
data += 2 + ppsinfo.size;
gst_buffer_unmap (sps, &spsinfo);
gst_buffer_unmap (pps, &ppsinfo);
gst_buffer_unmap (buf, &codecdatainfo);
return buf;
}
static void
_gst_mss_stream_add_h264_codec_data (GstCaps * caps, const gchar * codecdatastr)
{
GstBuffer *sps;
GstBuffer *pps;
GstBuffer *buffer;
gchar *sps_str;
gchar *pps_str;
GstH264NalUnit nalu;
GstH264SPS sps_struct;
GstH264ParserResult parseres;
GstMapInfo spsinfo;
/* search for the sps start */
if (g_str_has_prefix (codecdatastr, "00000001")) {
sps_str = (gchar *) codecdatastr + 8;
} else {
return; /* invalid mss codec data */
}
/* search for the pps start */
pps_str = g_strstr_len (sps_str, -1, "00000001");
if (!pps_str) {
return; /* invalid mss codec data */
}
pps_str[0] = '\0';
sps = gst_buffer_from_hex_string (sps_str);
pps_str[0] = '0';
pps_str = pps_str + 8;
pps = gst_buffer_from_hex_string (pps_str);
gst_buffer_map (sps, &spsinfo, GST_MAP_READ);
nalu.ref_idc = (spsinfo.data[0] & 0x60) >> 5;
nalu.type = GST_H264_NAL_SPS;
nalu.size = spsinfo.size;
nalu.data = spsinfo.data;
nalu.offset = 0;
nalu.sc_offset = 0;
nalu.valid = TRUE;
parseres = gst_h264_parse_sps (&nalu, &sps_struct, TRUE);
if (parseres == GST_H264_PARSER_OK) {
gst_caps_set_simple (caps, "framerate", GST_TYPE_FRACTION,
sps_struct.fps_num, sps_struct.fps_den, NULL);
}
buffer = _make_h264_codec_data (sps, pps);
gst_buffer_unmap (sps, &spsinfo);
gst_buffer_unref (sps);
gst_buffer_unref (pps);
if (buffer != NULL) {
gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, buffer, NULL);
gst_buffer_unref (buffer);
}
}
static GstCaps *
_gst_mss_stream_video_caps_from_qualitylevel_xml (GstMssStreamQuality * q)
{
xmlNodePtr node = q->xmlnode;
GstCaps *caps;
GstStructure *structure;
gchar *fourcc = (gchar *) xmlGetProp (node, (xmlChar *) "FourCC");
gchar *max_width = (gchar *) xmlGetProp (node, (xmlChar *) "MaxWidth");
gchar *max_height = (gchar *) xmlGetProp (node, (xmlChar *) "MaxHeight");
gchar *codec_data =
(gchar *) xmlGetProp (node, (xmlChar *) "CodecPrivateData");
if (!max_width)
max_width = (gchar *) xmlGetProp (node, (xmlChar *) "Width");
if (!max_height)
max_height = (gchar *) xmlGetProp (node, (xmlChar *) "Height");
caps = _gst_mss_stream_video_caps_from_fourcc (fourcc);
if (!caps)
goto end;
structure = gst_caps_get_structure (caps, 0);
if (max_width) {
gst_structure_set (structure, "width", G_TYPE_INT,
(int) g_ascii_strtoull (max_width, NULL, 10), NULL);
}
if (max_height) {
gst_structure_set (structure, "height", G_TYPE_INT,
(int) g_ascii_strtoull (max_height, NULL, 10), NULL);
}
if (codec_data && strlen (codec_data)) {
if (strcmp (fourcc, "H264") == 0 || strcmp (fourcc, "AVC1") == 0) {
_gst_mss_stream_add_h264_codec_data (caps, codec_data);
} else {
GstBuffer *buffer = gst_buffer_from_hex_string ((gchar *) codec_data);
gst_structure_set (structure, "codec_data", GST_TYPE_BUFFER, buffer,
NULL);
gst_buffer_unref (buffer);
}
}
end:
xmlFree (fourcc);
xmlFree (max_width);
xmlFree (max_height);
xmlFree (codec_data);
return caps;
}
static guint8
_frequency_index_from_sampling_rate (guint sampling_rate)
{
static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000, 44100,
32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350
};
guint8 i;
for (i = 0; i < G_N_ELEMENTS (aac_sample_rates); i++) {
if (aac_sample_rates[i] == sampling_rate)
return i;
}
return 15;
}
static GstBuffer *
_make_aacl_codec_data (guint64 sampling_rate, guint64 channels)
{
GstBuffer *buf;
guint8 *data;
guint8 frequency_index;
guint8 buf_size;
GstMapInfo info;
buf_size = 2;
frequency_index = _frequency_index_from_sampling_rate (sampling_rate);
if (frequency_index == 15)
buf_size += 3;
buf = gst_buffer_new_and_alloc (buf_size);
gst_buffer_map (buf, &info, GST_MAP_WRITE);
data = info.data;
data[0] = 2 << 3; /* AAC-LC object type is 2 */
data[0] += frequency_index >> 1;
data[1] = (frequency_index & 0x01) << 7;
/* Sampling rate is not in frequencies table, write manually */
if (frequency_index == 15) {
data[1] += sampling_rate >> 17;
data[2] = (sampling_rate >> 9) & 0xFF;
data[3] = (sampling_rate >> 1) & 0xFF;
data[4] = sampling_rate & 0x01;
data += 3;
}
data[1] += (channels & 0x0F) << 3;
gst_buffer_unmap (buf, &info);
return buf;
}
static GstCaps *
_gst_mss_stream_audio_caps_from_qualitylevel_xml (GstMssStreamQuality * q)
{
xmlNodePtr node = q->xmlnode;
GstCaps *caps = NULL;
GstStructure *structure;
gchar *fourcc = (gchar *) xmlGetProp (node, (xmlChar *) "FourCC");
gchar *audiotag = (gchar *) xmlGetProp (node, (xmlChar *) "AudioTag");
gchar *channels_str = (gchar *) xmlGetProp (node, (xmlChar *) "Channels");
gchar *rate_str = (gchar *) xmlGetProp (node, (xmlChar *) "SamplingRate");
gchar *block_align_str =
(gchar *) xmlGetProp (node, (xmlChar *) "PacketSize");
gchar *codec_data_str =
(gchar *) xmlGetProp (node, (xmlChar *) "CodecPrivateData");
GstBuffer *codec_data = NULL;
gint block_align = 0;
gint rate = 0;
gint channels = 0;
gint atag = 0;
if (!fourcc) /* sometimes the fourcc is omitted, we fallback to the Subtype in the StreamIndex node */
fourcc = (gchar *) xmlGetProp (node->parent, (xmlChar *) "Subtype");
if (fourcc) {
caps = _gst_mss_stream_audio_caps_from_fourcc (fourcc);
} else if (audiotag) {
atag = g_ascii_strtoull (audiotag, NULL, 10);
caps = _gst_mss_stream_audio_caps_from_audio_tag (atag);
}
if (!caps)
goto end;
structure = gst_caps_get_structure (caps, 0);
if (codec_data_str && strlen (codec_data_str)) {
codec_data = gst_buffer_from_hex_string ((gchar *) codec_data_str);
}
if (rate_str)
rate = (gint) g_ascii_strtoull (rate_str, NULL, 10);
if (channels_str)
channels = (int) g_ascii_strtoull (channels_str, NULL, 10);
if (block_align_str)
block_align = (int) g_ascii_strtoull (block_align_str, NULL, 10);
if (!codec_data) {
gint codec_data_len;
codec_data_str = (gchar *) xmlGetProp (node, (xmlChar *) "WaveFormatEx");
if (codec_data_str != NULL) {
codec_data_len = strlen (codec_data_str) / 2;
/* a WAVEFORMATEX structure is 18 bytes */
if (codec_data_str && codec_data_len >= 18) {
GstMapInfo mapinfo;
codec_data = gst_buffer_from_hex_string ((gchar *) codec_data_str);
/* since this is a WAVEFORMATEX, try to get the block_align and rate */
gst_buffer_map (codec_data, &mapinfo, GST_MAP_READ);
if (!channels_str) {
channels = GST_READ_UINT16_LE (mapinfo.data + 2);
}
if (!rate_str) {
rate = GST_READ_UINT32_LE (mapinfo.data + 4);
}
if (!block_align) {
block_align = GST_READ_UINT16_LE (mapinfo.data + 12);
}
gst_buffer_unmap (codec_data, &mapinfo);
/* Consume all the WAVEFORMATEX structure, and pass only the rest of
* the data as the codec private data */
gst_buffer_resize (codec_data, 18, -1);
} else {
GST_WARNING ("Dropping WaveFormatEx: data is %d bytes, "
"but at least 18 bytes are expected", codec_data_len);
}
}
}
if (!codec_data && ((fourcc && strcmp (fourcc, "AACL") == 0) || atag == 255)
&& rate && channels) {
codec_data = _make_aacl_codec_data (rate, channels);
}
if (block_align)
gst_structure_set (structure, "block_align", G_TYPE_INT, block_align, NULL);
if (channels)
gst_structure_set (structure, "channels", G_TYPE_INT, channels, NULL);
if (rate)
gst_structure_set (structure, "rate", G_TYPE_INT, rate, NULL);
if (q->bitrate)
gst_structure_set (structure, "bitrate", G_TYPE_INT, q->bitrate, NULL);
if (codec_data)
gst_structure_set (structure, "codec_data", GST_TYPE_BUFFER, codec_data,
NULL);
end:
if (codec_data)
gst_buffer_unref (codec_data);
xmlFree (fourcc);
xmlFree (audiotag);
xmlFree (channels_str);
xmlFree (rate_str);
xmlFree (block_align_str);
xmlFree (codec_data_str);
return caps;
}
void
gst_mss_stream_set_active (GstMssStream * stream, gboolean active)
{
stream->active = active;
}
guint64
gst_mss_stream_get_timescale (GstMssStream * stream)
{
gchar *timescale;
guint64 ts = DEFAULT_TIMESCALE;
timescale =
(gchar *) xmlGetProp (stream->xmlnode, (xmlChar *) MSS_PROP_TIMESCALE);
if (!timescale) {
timescale =
(gchar *) xmlGetProp (stream->xmlnode->parent,
(xmlChar *) MSS_PROP_TIMESCALE);
}
if (timescale) {
ts = g_ascii_strtoull (timescale, NULL, 10);
xmlFree (timescale);
}
return ts;
}
guint64
gst_mss_manifest_get_timescale (GstMssManifest * manifest)
{
gchar *timescale;
guint64 ts = DEFAULT_TIMESCALE;
timescale =
(gchar *) xmlGetProp (manifest->xmlrootnode,
(xmlChar *) MSS_PROP_TIMESCALE);
if (timescale) {
ts = g_ascii_strtoull (timescale, NULL, 10);
xmlFree (timescale);
}
return ts;
}
guint64
gst_mss_manifest_get_duration (GstMssManifest * manifest)
{
gchar *duration;
guint64 dur = -1;
duration =
(gchar *) xmlGetProp (manifest->xmlrootnode,
(xmlChar *) MSS_PROP_STREAM_DURATION);
if (duration) {
dur = g_ascii_strtoull (duration, NULL, 10);
xmlFree (duration);
}
return dur;
}
/**
* Gets the duration in nanoseconds
*/
GstClockTime
gst_mss_manifest_get_gst_duration (GstMssManifest * manifest)
{
guint64 duration = -1;
guint64 timescale;
GstClockTime gstdur = GST_CLOCK_TIME_NONE;
duration = gst_mss_manifest_get_duration (manifest);
timescale = gst_mss_manifest_get_timescale (manifest);
if (duration != -1 && timescale != -1)
gstdur =
(GstClockTime) gst_util_uint64_scale_round (duration, GST_SECOND,
timescale);
return gstdur;
}
GstCaps *
gst_mss_stream_get_caps (GstMssStream * stream)
{
GstMssStreamType streamtype = gst_mss_stream_get_type (stream);
GstMssStreamQuality *qualitylevel = stream->current_quality->data;
GstCaps *caps = NULL;
if (streamtype == MSS_STREAM_TYPE_VIDEO)
caps = _gst_mss_stream_video_caps_from_qualitylevel_xml (qualitylevel);
else if (streamtype == MSS_STREAM_TYPE_AUDIO)
caps = _gst_mss_stream_audio_caps_from_qualitylevel_xml (qualitylevel);
return caps;
}
GstFlowReturn
gst_mss_stream_get_fragment_url (GstMssStream * stream, gchar ** url)
{
gchar *tmp;
gchar *start_time_str;
GstMssStreamFragment *fragment;
GstMssStreamQuality *quality = stream->current_quality->data;
g_return_val_if_fail (stream->active, GST_FLOW_ERROR);
if (stream->current_fragment == NULL) /* stream is over */
return GST_FLOW_EOS;
fragment = stream->current_fragment->data;
start_time_str = g_strdup_printf ("%" G_GUINT64_FORMAT, fragment->time);
tmp = g_regex_replace_literal (stream->regex_bitrate, stream->url,
strlen (stream->url), 0, quality->bitrate_str, 0, NULL);
*url = g_regex_replace_literal (stream->regex_position, tmp,
strlen (tmp), 0, start_time_str, 0, NULL);
g_free (tmp);
g_free (start_time_str);
if (*url == NULL)
return GST_FLOW_ERROR;
return GST_FLOW_OK;
}
GstClockTime
gst_mss_stream_get_fragment_gst_timestamp (GstMssStream * stream)
{
guint64 time;
guint64 timescale;
GstMssStreamFragment *fragment;
g_return_val_if_fail (stream->active, GST_FLOW_ERROR);
if (!stream->current_fragment)
return GST_CLOCK_TIME_NONE;
fragment = stream->current_fragment->data;
time = fragment->time;
timescale = gst_mss_stream_get_timescale (stream);
return (GstClockTime) gst_util_uint64_scale_round (time, GST_SECOND,
timescale);
}
GstClockTime
gst_mss_stream_get_fragment_gst_duration (GstMssStream * stream)
{
guint64 dur;
guint64 timescale;
GstMssStreamFragment *fragment;
g_return_val_if_fail (stream->active, GST_FLOW_ERROR);
if (!stream->current_fragment)
return GST_CLOCK_TIME_NONE;
fragment = stream->current_fragment->data;
dur = fragment->duration;
timescale = gst_mss_stream_get_timescale (stream);
return (GstClockTime) gst_util_uint64_scale_round (dur, GST_SECOND,
timescale);
}
GstFlowReturn
gst_mss_stream_advance_fragment (GstMssStream * stream)
{
g_return_val_if_fail (stream->active, GST_FLOW_ERROR);
if (stream->current_fragment == NULL)
return GST_FLOW_EOS;
stream->current_fragment = g_list_next (stream->current_fragment);
if (stream->current_fragment == NULL)
return GST_FLOW_EOS;
return GST_FLOW_OK;
}
const gchar *
gst_mss_stream_type_name (GstMssStreamType streamtype)
{
switch (streamtype) {
case MSS_STREAM_TYPE_VIDEO:
return "video";
case MSS_STREAM_TYPE_AUDIO:
return "audio";
case MSS_STREAM_TYPE_UNKNOWN:
default:
return "unknown";
}
}
/**
* Seeks all streams to the fragment that contains the set time
*
* @time: time in nanoseconds
*/
gboolean
gst_mss_manifest_seek (GstMssManifest * manifest, guint64 time)
{
gboolean ret = TRUE;
GSList *iter;
for (iter = manifest->streams; iter; iter = g_slist_next (iter)) {
ret = gst_mss_stream_seek (iter->data, time) & ret;
}
return ret;
}
/**
* Seeks this stream to the fragment that contains the sample at time
*
* @time: time in nanoseconds
*/
gboolean
gst_mss_stream_seek (GstMssStream * stream, guint64 time)
{
GList *iter;
guint64 timescale;
timescale = gst_mss_stream_get_timescale (stream);
time = gst_util_uint64_scale_round (time, timescale, GST_SECOND);
for (iter = stream->fragments; iter; iter = g_list_next (iter)) {
GList *next = g_list_next (iter);
if (next) {
GstMssStreamFragment *fragment = next->data;
if (fragment->time > time) {
stream->current_fragment = iter;
break;
}
} else {
GstMssStreamFragment *fragment = iter->data;
if (fragment->time + fragment->duration > time) {
stream->current_fragment = iter;
} else {
stream->current_fragment = NULL; /* EOS */
}
break;
}
}
return TRUE;
}
guint64
gst_mss_manifest_get_current_bitrate (GstMssManifest * manifest)
{
guint64 bitrate = 0;
GSList *iter;
for (iter = gst_mss_manifest_get_streams (manifest); iter;
iter = g_slist_next (iter)) {
GstMssStream *stream = iter->data;
if (stream->active && stream->current_quality) {
GstMssStreamQuality *q = stream->current_quality->data;
bitrate += q->bitrate;
}
}
return bitrate;
}
gboolean
gst_mss_manifest_is_live (GstMssManifest * manifest)
{
return manifest->is_live;
}
static void
gst_mss_stream_reload_fragments (GstMssStream * stream, xmlNodePtr streamIndex)
{
xmlNodePtr iter;
GList *new_fragments = NULL;
GstMssStreamFragment *previous_fragment = NULL;
GstMssStreamFragment *current_fragment =
stream->current_fragment ? stream->current_fragment->data : NULL;
guint64 current_time = gst_mss_stream_get_fragment_gst_timestamp (stream);
guint fragment_number = 0;
guint64 fragment_time_accum = 0;
if (!current_fragment && stream->fragments) {
current_fragment = g_list_last (stream->fragments)->data;
} else if (g_list_previous (stream->current_fragment)) {
/* rewind one as this is the next to be pushed */
current_fragment = g_list_previous (stream->current_fragment)->data;
} else {
current_fragment = NULL;
}
if (current_fragment) {
current_time = current_fragment->time;
fragment_number = current_fragment->number;
fragment_time_accum = current_fragment->time;
}
for (iter = streamIndex->children; iter; iter = iter->next) {
if (node_has_type (iter, MSS_NODE_STREAM_FRAGMENT)) {
gchar *duration_str;
gchar *time_str;
gchar *seqnum_str;
GstMssStreamFragment *fragment = g_new (GstMssStreamFragment, 1);
duration_str = (gchar *) xmlGetProp (iter, (xmlChar *) MSS_PROP_DURATION);
time_str = (gchar *) xmlGetProp (iter, (xmlChar *) MSS_PROP_TIME);
seqnum_str = (gchar *) xmlGetProp (iter, (xmlChar *) MSS_PROP_NUMBER);
/* use the node's seq number or use the previous + 1 */
if (seqnum_str) {
fragment->number = g_ascii_strtoull (seqnum_str, NULL, 10);
xmlFree (seqnum_str);
} else {
fragment->number = fragment_number;
}
fragment_number = fragment->number + 1;
if (time_str) {
fragment->time = g_ascii_strtoull (time_str, NULL, 10);
xmlFree (time_str);
fragment_time_accum = fragment->time;
} else {
fragment->time = fragment_time_accum;
}
/* if we have a previous fragment, means we need to set its duration */
if (previous_fragment)
previous_fragment->duration = fragment->time - previous_fragment->time;
if (duration_str) {
fragment->duration = g_ascii_strtoull (duration_str, NULL, 10);
previous_fragment = NULL;
fragment_time_accum += fragment->duration;
xmlFree (duration_str);
} else {
/* store to set the duration at the next iteration */
previous_fragment = fragment;
}
if (fragment->time > current_time) {
new_fragments = g_list_append (new_fragments, fragment);
} else {
previous_fragment = NULL;
g_free (fragment);
}
} else {
/* TODO gst log this */
}
}
/* store the new fragments list */
if (new_fragments) {
g_list_free_full (stream->fragments, g_free);
stream->fragments = new_fragments;
stream->current_fragment = new_fragments;
}
}
static void
gst_mss_manifest_reload_fragments_from_xml (GstMssManifest * manifest,
xmlNodePtr root)
{
xmlNodePtr nodeiter;
GSList *streams = manifest->streams;
/* we assume the server is providing the streams in the same order in
* every manifest */
for (nodeiter = root->children; nodeiter && streams;
nodeiter = nodeiter->next) {
if (nodeiter->type == XML_ELEMENT_NODE
&& (strcmp ((const char *) nodeiter->name, "StreamIndex") == 0)) {
gst_mss_stream_reload_fragments (streams->data, nodeiter);
streams = g_slist_next (streams);
}
}
}
void
gst_mss_manifest_reload_fragments (GstMssManifest * manifest, GstBuffer * data)
{
xmlDocPtr xml;
xmlNodePtr root;
GstMapInfo info;
g_return_if_fail (manifest->is_live);
gst_buffer_map (data, &info, GST_MAP_READ);
xml = xmlReadMemory ((const gchar *) info.data,
info.size, "manifest", NULL, 0);
root = xmlDocGetRootElement (xml);
gst_mss_manifest_reload_fragments_from_xml (manifest, root);
xmlFreeDoc (xml);
gst_buffer_unmap (data, &info);
}
gboolean
gst_mss_stream_select_bitrate (GstMssStream * stream, guint64 bitrate)
{
GList *iter = stream->current_quality;
GList *next;
GstMssStreamQuality *q = iter->data;
while (q->bitrate > bitrate) {
next = g_list_previous (iter);
if (next) {
iter = next;
q = iter->data;
} else {
break;
}
}
while (q->bitrate < bitrate) {
GstMssStreamQuality *next_q;
next = g_list_next (iter);
if (next) {
next_q = next->data;
if (next_q->bitrate < bitrate) {
iter = next;
q = iter->data;
} else {
break;
}
} else {
break;
}
}
if (iter == stream->current_quality)
return FALSE;
stream->current_quality = iter;
return TRUE;
}
guint64
gst_mss_stream_get_current_bitrate (GstMssStream * stream)
{
GstMssStreamQuality *q;
if (stream->current_quality == NULL)
return 0;
q = stream->current_quality->data;
return q->bitrate;
}
/**
* gst_mss_manifest_change_bitrate:
* @manifest: the manifest
* @bitrate: the maximum bitrate to use (bps)
*
* Iterates over the active streams and changes their bitrates to the maximum
* value so that the bitrates of all streams are not larger than
* @bitrate.
*
* Return: %TRUE if any stream changed its bitrate
*/
gboolean
gst_mss_manifest_change_bitrate (GstMssManifest * manifest, guint64 bitrate)
{
gboolean ret = FALSE;
GSList *iter;
/* TODO This algorithm currently sets the same bitrate for all streams,
* it should actually use the sum of all streams bitrates to compare to
* the target value */
if (bitrate == 0) {
/* use maximum */
bitrate = G_MAXUINT64;
}
for (iter = gst_mss_manifest_get_streams (manifest); iter;
iter = g_slist_next (iter)) {
GstMssStream *stream = iter->data;
if (stream->active) {
ret = ret | gst_mss_stream_select_bitrate (stream, bitrate);
}
}
return ret;
}
static GstBuffer *
gst_buffer_from_hex_string (const gchar * s)
{
GstBuffer *buffer = NULL;
gint len;
gchar ts[3];
guint8 *data;
gint i;
GstMapInfo info;
len = strlen (s);
if (len & 1)
return NULL;
buffer = gst_buffer_new_and_alloc (len / 2);
gst_buffer_map (buffer, &info, GST_MAP_WRITE);
data = info.data;
for (i = 0; i < len / 2; i++) {
if (!isxdigit ((int) s[i * 2]) || !isxdigit ((int) s[i * 2 + 1])) {
gst_buffer_unref (buffer);
return NULL;
}
ts[0] = s[i * 2 + 0];
ts[1] = s[i * 2 + 1];
ts[2] = 0;
data[i] = (guint8) strtoul (ts, NULL, 16);
}
gst_buffer_unmap (buffer, &info);
return buffer;
}
const gchar *
gst_mss_stream_get_lang (GstMssStream * stream)
{
return stream->lang;
}