rtpdec: Add support for G726 audio

This requires using a separate init function, since there
isn't necessarily any fmtp lines for this codec, so
parse_sdp_a_line won't be called. Incorporating it with the
alloc function wouldn't do either, since it is called before
the full rtpmap line is parsed (where the sample rate is
extracted).

Signed-off-by: Martin Storsjö <martin@martin.st>

1 files changed, 102 insertions, 0 deletions

diff --git a/libavformat/rtpdec_g726.c b/libavformat/rtpdec_g726.c
new file mode 100644
index 0000000000..0b7b0f0f94
--- /dev/null
+++ b/libavformat/rtpdec_g726.c
@@ -0,0 +1,102 @@
+/*
+ * Copyright (c) 2011 Miroslav Slugeň <Thunder.m@seznam.cz>
+ *
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * Libav 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "avformat.h"
+#include "rtpdec_formats.h"
+
+static int g726_16_init(AVFormatContext *s, int st_index, PayloadContext *data)
+{
+    AVStream *stream = s->streams[st_index];
+    AVCodecContext *codec = stream->codec;
+
+    codec->bit_rate = 16000;
+    if (codec->sample_rate)
+        codec->bits_per_coded_sample =
+            av_clip((codec->bit_rate + codec->sample_rate/2) / codec->sample_rate, 2, 5);
+
+    return 0;
+}
+
+static int g726_24_init(AVFormatContext *s, int st_index, PayloadContext *data)
+{
+    AVStream *stream = s->streams[st_index];
+    AVCodecContext *codec = stream->codec;
+
+    codec->bit_rate = 24000;
+    if (codec->sample_rate)
+        codec->bits_per_coded_sample =
+            av_clip((codec->bit_rate + codec->sample_rate/2) / codec->sample_rate, 2, 5);
+
+    return 0;
+}
+
+static int g726_32_init(AVFormatContext *s, int st_index, PayloadContext *data)
+{
+    AVStream *stream = s->streams[st_index];
+    AVCodecContext *codec = stream->codec;
+
+    codec->bit_rate = 32000;
+    if (codec->sample_rate)
+        codec->bits_per_coded_sample =
+            av_clip((codec->bit_rate + codec->sample_rate/2) / codec->sample_rate, 2, 5);
+
+    return 0;
+}
+
+static int g726_40_init(AVFormatContext *s, int st_index, PayloadContext *data)
+{
+    AVStream *stream = s->streams[st_index];
+    AVCodecContext *codec = stream->codec;
+
+    codec->bit_rate = 40000;
+    if (codec->sample_rate)
+        codec->bits_per_coded_sample =
+            av_clip((codec->bit_rate + codec->sample_rate/2) / codec->sample_rate, 2, 5);
+
+    return 0;
+}
+
+RTPDynamicProtocolHandler ff_g726_16_dynamic_handler = {
+    .enc_name   = "G726-16",
+    .codec_type = AVMEDIA_TYPE_AUDIO,
+    .codec_id   = CODEC_ID_ADPCM_G726,
+    .init       = g726_16_init,
+};
+
+RTPDynamicProtocolHandler ff_g726_24_dynamic_handler = {
+    .enc_name   = "G726-24",
+    .codec_type = AVMEDIA_TYPE_AUDIO,
+    .codec_id   = CODEC_ID_ADPCM_G726,
+    .init       = g726_24_init,
+};
+
+RTPDynamicProtocolHandler ff_g726_32_dynamic_handler = {
+    .enc_name   = "G726-32",
+    .codec_type = AVMEDIA_TYPE_AUDIO,
+    .codec_id   = CODEC_ID_ADPCM_G726,
+    .init       = g726_32_init,
+};
+
+RTPDynamicProtocolHandler ff_g726_40_dynamic_handler = {
+    .enc_name   = "G726-40",
+    .codec_type = AVMEDIA_TYPE_AUDIO,
+    .codec_id   = CODEC_ID_ADPCM_G726,
+    .init       = g726_40_init,
+};