codecs: Add support for G.722 and G.726

Integrate the Spandsp library for G.722 and G.726 support. Adds support
for G.722 and all eight variants of G.726.

Note: this also fixes a crash in Qt (buffer overrun, reading too much
data) caused by confusion of the larger output buffer (resample_buff)
with the smaller input buffer (decode_buff). It was not triggered before
because the sample rate was always 8k, but with the addition of the new
codecs, a different sample rate became possible (16k).

Fix also a crash which occurs when the RTP_STREAM_DEBUG macro is enabled
and the VOIP Calls dialog is opened (the begin frame, start_fd, is not
yet known and therfore a NULL dereference could occur).

Passes testing (plays normally without bad RTP timing errors) with
SampleCaptures files: sip-rtp-g722.pcap and sip-rtp-g726.pcap. Tested
with cmake (Qt), autotools (Qt and GTK+) with ASAN enabled.

Bug: 5619
Change-Id: I5661908d193927bba50901079119eeff0c04991f
Reviewed-on: https://code.wireshark.org/review/18939
Petri-Dish: Peter Wu <peter@lekensteyn.nl>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Alexis La Goutte <alexis.lagoutte@gmail.com>
Reviewed-by: Peter Wu <peter@lekensteyn.nl>

2 files changed, 9 insertions, 3 deletions

diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp
index 1d8971c1d1..798d73fcbe 100644
--- a/ui/qt/rtp_audio_stream.cpp
+++ b/ui/qt/rtp_audio_stream.cpp
@@ -252,7 +252,13 @@ void RtpAudioStream::decode()
         }
         last_sequence = rtp_packet->info->info_seq_num;
 
-        double rtp_time = (double)(rtp_packet->info->info_timestamp-start_timestamp)/sample_rate - start_rtp_time;
+        unsigned rtp_clock_rate = sample_rate;
+        if (rtp_packet->info->info_payload_type == PT_G722) {
+            // G.722 sample rate is 16kHz, but RTP clock rate is 8kHz for historic reasons.
+            rtp_clock_rate = 8000;
+        }
+
+        double rtp_time = (double)(rtp_packet->info->info_timestamp-start_timestamp)/rtp_clock_rate - start_rtp_time;
         double arrive_time;
         if (timing_mode_ == RtpTimestamp) {
             arrive_time = rtp_time;
@@ -351,7 +357,7 @@ void RtpAudioStream::decode()
             }
 
             speex_resampler_process_int(audio_resampler_, 0, decode_buff, &in_len, resample_buff, &out_len);
-            write_buff = (char *) decode_buff;
+            write_buff = (char *) resample_buff;
             write_bytes = out_len * sample_bytes_;
         }
 
diff --git a/ui/qt/rtp_player_dialog.cpp b/ui/qt/rtp_player_dialog.cpp
index 100daa1adb..743685c307 100644
--- a/ui/qt/rtp_player_dialog.cpp
+++ b/ui/qt/rtp_player_dialog.cpp
@@ -392,7 +392,7 @@ void RtpPlayerDialog::addRtpStream(struct _rtp_stream_info *rtp_stream)
                      ui->streamTreeWidget->topLevelItemCount(),
                      rtp_stream->packet_count,
                      g_list_length(rtp_stream->rtp_packet_list),
-                     rtp_stream->start_fd->num);
+                     rtp_stream->start_fd ? rtp_stream->start_fd->num : 0);
 }
 
 void RtpPlayerDialog::showEvent(QShowEvent *)