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Asterisk media architecture conversion - no more format bitfields

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This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal.  For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal

The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs.  Functionally
no change in behavior should be present in this patch.  Thanks to twilson
and russell for all the time they spent reviewing these changes.

Review: https://reviewboard.asterisk.org/r/1083/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
David Vossel
2011-02-03 16:22:10 +00:00
parent 652fb64a01
commit c26c190711
168 changed files with 8120 additions and 3764 deletions
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37
channels/h323/ast_h323.cxx
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@@ -1770,20 +1770,21 @@ PBoolean MyH323Connection::OnReceivedCapabilitySet(const H323Capabilities & remo
if ((subType == codecs[x].h245_cap) && (!codecs[x].formatName || (!strcmp(codecs[x].formatName, (const char *)remoteCapabilities[i].GetFormatName())))) {
int ast_codec = codecs[x].asterisk_codec;
int ms = 0;
if (!(peer_capabilities & ast_codec)) {
struct ast_format tmpfmt;
if (!(peer_capabilities & ast_format_id_to_old_bitfield((enum ast_format_id) ast_codec))) {
struct ast_format_list format;
ast_codec_pref_append(&prefs, ast_codec);
format = ast_codec_pref_getsize(&prefs, ast_codec);
ast_codec_pref_append(&prefs, ast_format_set(&tmpfmt, (enum ast_format_id) ast_codec, 0));
format = ast_codec_pref_getsize(&prefs, &tmpfmt);
if ((ast_codec == AST_FORMAT_ALAW) || (ast_codec == AST_FORMAT_ULAW)) {
ms = remoteCapabilities[i].GetTxFramesInPacket();
} else
ms = remoteCapabilities[i].GetTxFramesInPacket() * format.inc_ms;
ast_codec_pref_setsize(&prefs, ast_codec, ms);
ast_codec_pref_setsize(&prefs, &tmpfmt, ms);
}
if (h323debug) {
cout << "Found peer capability " << remoteCapabilities[i] << ", Asterisk code is " << ast_codec << ", frame size (in ms) is " << ms << endl;
}
peer_capabilities |= ast_codec;
peer_capabilities |= ast_format_id_to_old_bitfield((enum ast_format_id) ast_codec);
}
}
break;
@@ -1846,12 +1847,7 @@ PBoolean MyH323Connection::OnReceivedCapabilitySet(const H323Capabilities & remo
break;
}
}
if (h323debug) {
char caps_str[1024], caps2_str[1024];
ast_codec_pref_string(&prefs, caps2_str, sizeof(caps2_str));
cout << "Peer capabilities = " << ast_getformatname_multiple(caps_str, sizeof(caps_str), peer_capabilities)
<< ", ordered list is " << caps2_str << endl;
}
#if 0
redir_capabilities &= peer_capabilities;
#endif
@@ -1897,38 +1893,37 @@ void MyH323Connection::SetCapabilities(int caps, int dtmf_mode, void *_prefs, in
int alreadysent = 0;
int codec;
int x, y;
char caps_str[1024];
struct ast_codec_pref *prefs = (struct ast_codec_pref *)_prefs;
struct ast_format_list format;
int frames_per_packet;
struct ast_format tmpfmt;
H323Capability *cap;
localCapabilities.RemoveAll();
if (h323debug) {
cout << "Setting capabilities to " << ast_getformatname_multiple(caps_str, sizeof(caps_str), caps) << endl;
ast_codec_pref_string(prefs, caps_str, sizeof(caps_str));
cout << "Capabilities in preference order is " << caps_str << endl;
}
/* Add audio codecs in preference order first, then
audio codecs without preference as allowed by mask */
for (y = 0, x = -1; x < 32 + 32; ++x) {
ast_format_clear(&tmpfmt);
if (x < 0)
codec = pref_codec;
else if (y || (!(codec = ast_codec_pref_index(prefs, x)))) {
else if (y || (!(ast_codec_pref_index(prefs, x, &tmpfmt)))) {
if (!y)
y = 1;
else
y <<= 1;
codec = y;
}
if (!(caps & codec) || (alreadysent & codec) || !(codec & AST_FORMAT_AUDIO_MASK))
if (tmpfmt.id) {
codec = ast_format_to_old_bitfield(&tmpfmt);
}
if (!(caps & codec) || (alreadysent & codec) || (AST_FORMAT_GET_TYPE(ast_format_id_from_old_bitfield(codec)) != AST_FORMAT_TYPE_AUDIO))
continue;
alreadysent |= codec;
/* format.cur_ms will be set to default if packetization is not explicitly set */
format = ast_codec_pref_getsize(prefs, codec);
format = ast_codec_pref_getsize(prefs, ast_format_from_old_bitfield(&tmpfmt, codec));
frames_per_packet = (format.inc_ms ? format.cur_ms / format.inc_ms : format.cur_ms);
switch(codec) {
switch(ast_format_id_from_old_bitfield(codec)) {
#if 0
case AST_FORMAT_SPEEX:
/* Not real sure if Asterisk acutally supports all