From ad0e4442e98f2309ff1bdef2f12ca4bcbb463d06 Mon Sep 17 00:00:00 2001 From: =?utf8?q?Tim-Philipp=20M=C3=BCller?= Date: Fri, 8 Apr 2011 15:59:58 +0100 Subject: [PATCH] audioparsers: update for set_frame_props -> set_frame_rate API change --- gst/audioparsers/gstaacparse.c | 5 ++--- gst/audioparsers/gstac3parse.c | 2 +- gst/audioparsers/gstamrparse.c | 4 ++-- gst/audioparsers/gstdcaparse.c | 2 +- gst/audioparsers/gstmpegaudioparse.c | 2 +- 5 files changed, 7 insertions(+), 8 deletions(-) diff --git a/gst/audioparsers/gstaacparse.c b/gst/audioparsers/gstaacparse.c index e7354ce..df7c401 100644 --- a/gst/audioparsers/gstaacparse.c +++ b/gst/audioparsers/gstaacparse.c @@ -456,8 +456,7 @@ gst_aac_parse_detect_stream (GstAacParse * aacparse, gst_aac_parse_parse_adts_header (aacparse, data, &rate, &channels, &aacparse->object_type, &aacparse->mpegversion); - gst_base_parse_set_frame_props (GST_BASE_PARSE (aacparse), - rate, 1024, 2, 2); + gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate, 1024, 2, 2); GST_DEBUG ("ADTS: samplerate %d, channels %d, objtype %d, version %d", rate, channels, aacparse->object_type, aacparse->mpegversion); @@ -674,7 +673,7 @@ gst_aac_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame) ret = GST_FLOW_NOT_LINKED; } - gst_base_parse_set_frame_props (GST_BASE_PARSE (aacparse), + gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), aacparse->sample_rate, 1024, 2, 2); } diff --git a/gst/audioparsers/gstac3parse.c b/gst/audioparsers/gstac3parse.c index 986b28b..e1654cf 100644 --- a/gst/audioparsers/gstac3parse.c +++ b/gst/audioparsers/gstac3parse.c @@ -492,7 +492,7 @@ gst_ac3_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame) ac3parse->channels = chans; ac3parse->eac = eac; - gst_base_parse_set_frame_props (parse, rate, 256 * blocks, 2, 2); + gst_base_parse_set_frame_rate (parse, rate, 256 * blocks, 2, 2); } return GST_FLOW_OK; diff --git a/gst/audioparsers/gstamrparse.c b/gst/audioparsers/gstamrparse.c index 61fa234..99d31b9 100644 --- a/gst/audioparsers/gstamrparse.c +++ b/gst/audioparsers/gstamrparse.c @@ -214,7 +214,7 @@ gst_amr_parse_sink_setcaps (GstBaseParse * parse, GstCaps * caps) } amrparse->need_header = FALSE; - gst_base_parse_set_frame_props (GST_BASE_PARSE (amrparse), 50, 1, 2, 2); + gst_base_parse_set_frame_rate (GST_BASE_PARSE (amrparse), 50, 1, 2, 2); gst_amr_parse_set_src_caps (amrparse); return TRUE; } @@ -285,7 +285,7 @@ gst_amr_parse_check_valid_frame (GstBaseParse * parse, if (dsize >= AMR_MIME_HEADER_SIZE && gst_amr_parse_parse_header (amrparse, data, skipsize)) { amrparse->need_header = FALSE; - gst_base_parse_set_frame_props (GST_BASE_PARSE (amrparse), 50, 1, 2, 2); + gst_base_parse_set_frame_rate (GST_BASE_PARSE (amrparse), 50, 1, 2, 2); } else { GST_WARNING ("media doesn't look like a AMR format"); } diff --git a/gst/audioparsers/gstdcaparse.c b/gst/audioparsers/gstdcaparse.c index bc52233..2bf0e38 100644 --- a/gst/audioparsers/gstdcaparse.c +++ b/gst/audioparsers/gstdcaparse.c @@ -436,7 +436,7 @@ gst_dca_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame) dcaparse->block_size = block_size; dcaparse->frame_size = size; - gst_base_parse_set_frame_props (parse, rate, block_size, 0, 0); + gst_base_parse_set_frame_rate (parse, rate, block_size, 0, 0); } return GST_FLOW_OK; diff --git a/gst/audioparsers/gstmpegaudioparse.c b/gst/audioparsers/gstmpegaudioparse.c index d3c8d5e..c67f9b8 100644 --- a/gst/audioparsers/gstmpegaudioparse.c +++ b/gst/audioparsers/gstmpegaudioparse.c @@ -1001,7 +1001,7 @@ gst_mpeg_audio_parse_parse_frame (GstBaseParse * parse, * Some mp3 streams have an offset in the timestamps, for which we have to * push the frame *after* the end position in order for the decoder to be * able to decode everything up until the segment.stop position. */ - gst_base_parse_set_frame_props (parse, mp3parse->rate, mp3parse->spf, + gst_base_parse_set_frame_rate (parse, mp3parse->rate, mp3parse->spf, (version == 1) ? 10 : 30, 2); } -- 2.7.4