videolan
/
ffmpeg
mirror of https://git.videolan.org/git/ffmpeg.git
1
Fork 0
Code Releases Activity

rtp: Rename a static variable to normal naming conventions 

Signed-off-by: Martin Storsjö <martin@martin.st>
This commit is contained in:
Martin Storsjö 2013-01-11 14:46:54 +02:00
parent 58b5971881
commit c44784c9bb
1 changed files with 24 additions and 24 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

48
libavformat/rtp.c
View File

@ -37,7 +37,7 @@ static const struct {
enum AVCodecID codec_id;
int clock_rate;
int audio_channels;
} AVRtpPayloadTypes[] = {
} rtp_payload_types[] = {
{0, "PCMU", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_PCM_MULAW, 8000, 1},
{3, "GSM", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 8000, 1},
{4, "G723", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_G723_1, 8000, 1},
@ -71,15 +71,15 @@ int ff_rtp_get_codec_info(AVCodecContext *codec, int payload_type)
{
int i = 0;
for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
if (AVRtpPayloadTypes[i].pt == payload_type) {
if (AVRtpPayloadTypes[i].codec_id != AV_CODEC_ID_NONE) {
codec->codec_type = AVRtpPayloadTypes[i].codec_type;
codec->codec_id = AVRtpPayloadTypes[i].codec_id;
if (AVRtpPayloadTypes[i].audio_channels > 0)
codec->channels = AVRtpPayloadTypes[i].audio_channels;
if (AVRtpPayloadTypes[i].clock_rate > 0)
codec->sample_rate = AVRtpPayloadTypes[i].clock_rate;
for (i = 0; rtp_payload_types[i].pt >= 0; i++)
if (rtp_payload_types[i].pt == payload_type) {
if (rtp_payload_types[i].codec_id != AV_CODEC_ID_NONE) {
codec->codec_type = rtp_payload_types[i].codec_type;
codec->codec_id = rtp_payload_types[i].codec_id;
if (rtp_payload_types[i].audio_channels > 0)
codec->channels = rtp_payload_types[i].audio_channels;
if (rtp_payload_types[i].clock_rate > 0)
codec->sample_rate = rtp_payload_types[i].clock_rate;
return 0;
}
}
@ -101,8 +101,8 @@ int ff_rtp_get_payload_type(AVFormatContext *fmt,
}
/* static payload type */
for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) {
for (i = 0; rtp_payload_types[i].pt >= 0; ++i)
if (rtp_payload_types[i].codec_id == codec->codec_id) {
if (codec->codec_id == AV_CODEC_ID_H263 && (!fmt ||
!fmt->oformat->priv_class ||
!av_opt_flag_is_set(fmt->priv_data, "rtpflags", "rfc2190")))
@ -111,14 +111,14 @@ int ff_rtp_get_payload_type(AVFormatContext *fmt,
* see section 4.5.2 in RFC 3551. */
if (codec->codec_id == AV_CODEC_ID_ADPCM_G722 &&
codec->sample_rate == 16000 && codec->channels == 1)
return AVRtpPayloadTypes[i].pt;
return rtp_payload_types[i].pt;
if (codec->codec_type == AVMEDIA_TYPE_AUDIO &&
((AVRtpPayloadTypes[i].clock_rate > 0 &&
codec->sample_rate != AVRtpPayloadTypes[i].clock_rate) ||
(AVRtpPayloadTypes[i].audio_channels > 0 &&
codec->channels != AVRtpPayloadTypes[i].audio_channels)))
((rtp_payload_types[i].clock_rate > 0 &&
codec->sample_rate != rtp_payload_types[i].clock_rate) ||
(rtp_payload_types[i].audio_channels > 0 &&
codec->channels != rtp_payload_types[i].audio_channels)))
continue;
return AVRtpPayloadTypes[i].pt;
return rtp_payload_types[i].pt;
}
if (idx < 0)
@ -132,9 +132,9 @@ const char *ff_rtp_enc_name(int payload_type)
{
int i;
for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
if (AVRtpPayloadTypes[i].pt == payload_type)
return AVRtpPayloadTypes[i].enc_name;
for (i = 0; rtp_payload_types[i].pt >= 0; i++)
if (rtp_payload_types[i].pt == payload_type)
return rtp_payload_types[i].enc_name;
return "";
}
@ -143,9 +143,9 @@ enum AVCodecID ff_rtp_codec_id(const char *buf, enum AVMediaType codec_type)
{
int i;
for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++)
if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec_type == AVRtpPayloadTypes[i].codec_type))
return AVRtpPayloadTypes[i].codec_id;
for (i = 0; rtp_payload_types[i].pt >= 0; i++)
if (!strcmp(buf, rtp_payload_types[i].enc_name) && (codec_type == rtp_payload_types[i].codec_type))
return rtp_payload_types[i].codec_id;
return AV_CODEC_ID_NONE;
}