gst/rtp/gstrtpg729pay.*: Replace G729 payloader with an improved version. Fixes #532409.

Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtp/gstrtpg729pay.c: (gst_rtp_g729_pay_base_init),
(gst_rtp_g729_pay_class_init), (gst_rtp_g729_pay_init),
(gst_rtp_g729_pay_set_caps), (gst_rtp_g729_pay_handle_buffer):
* gst/rtp/gstrtpg729pay.h:
Replace G729 payloader with an improved version. Fixes #532409.

2 files changed, 232 insertions, 74 deletions

diff --git a/gst/rtp/gstrtpg729pay.c b/gst/rtp/gstrtpg729pay.c
index b2f0dd90..ca8674b2 100644
--- a/gst/rtp/gstrtpg729pay.c
+++ b/gst/rtp/gstrtpg729pay.c
@@ -1,4 +1,7 @@
 /* GStreamer
+ * Copyright (C) <2007> Nokia Corporation
+ * Copyright (C) <2007> Collabora Ltd
+ *  @author: Olivier Crete <olivier.crete@collabora.co.uk>
  *
  * This library is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Library General Public
@@ -16,129 +19,282 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/*
+ * This payloader assumes that the data will ALWAYS come as zero or more
+ * 10 bytes frame of audio followed by 0 or 1 2 byte frame of silence.
+ * Any other buffer format won't work
+ */
+
 #ifdef HAVE_CONFIG_H
-#include "config.h"
+#include <config.h>
 #endif
 
-#include "gstrtpg729pay.h"
+#include <string.h>
 #include <gst/rtp/gstrtpbuffer.h>
+#include <gst/base/gstadapter.h>
+
+#include "gstrtpg729pay.h"
+
+/* TODO: fix gstrtpbuffer.h */
+#undef GST_RTP_PAYLOAD_G729
+#define GST_RTP_PAYLOAD_G729 18
+#undef GST_RTP_PAYLOAD_G729_STRING
+#define GST_RTP_PAYLOAD_G729_STRING "18"
+
+#define G729_FRAME_SIZE 10
+#define G729B_CN_FRAME_SIZE 2
+#define G729_FRAME_DURATION (10 * GST_MSECOND)
+#define G729_FRAME_DURATION_MS (10)
+
+static gboolean
+gst_rtp_g729_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps);
+static GstFlowReturn
+gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf);
 
-/* elementfactory information */
-static GstElementDetails gst_rtpg729pay_details = {
-  "RTP Payloader for G729 Audio",
-  "Codec/Payloader/Network",
-  "Packetize G729 audio streams into RTP packets",
-  "Laurent Glayal <spglegle@yahoo.fr>"
-};
 
-GST_DEBUG_CATEGORY_STATIC (rtpg729pay_debug);
-#define GST_CAT_DEFAULT (rtpg729pay_debug)
+static const GstElementDetails gst_rtp_g729_pay_details =
+GST_ELEMENT_DETAILS ("G729 RTP packet payloader",
+    "Codec/Payloader/Network",
+    "Packetize G729 audio into RTP packets",
+    "Olivier Crete <olivier.crete@collabora.co.uk>");
 
-static GstStaticPadTemplate gst_rtpg729pay_sink_template =
+static GstStaticPadTemplate gst_rtp_g729_pay_sink_template =
 GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/G729, channels=(int)1, rate=(int)8000")
+    GST_STATIC_CAPS ("audio/G729, "     /* according to RFC 3555 */
+        "channels = (int) 1, " "rate = (int) 8000")
     );
 
-static GstStaticPadTemplate gst_rtpg729pay_src_template =
+static GstStaticPadTemplate gst_rtp_g729_pay_src_template =
     GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/x-rtp, "
         "media = (string) \"audio\", "
-        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
-        "clock-rate = (int) 8000, " "encoding-name = (string) \"G729\";"
+        "payload = (int) " GST_RTP_PAYLOAD_G729_STRING ", "
+        "clock-rate = (int) 8000, "
+        "encoding-name = (string) \"G729\"; "
         "application/x-rtp, "
         "media = (string) \"audio\", "
-        "payload = (int) " GST_RTP_PAYLOAD_G729_STRING ", "
-        "clock-rate = (int) 8000")
+        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+        "clock-rate = (int) 8000, " "encoding-name = (string) \"G729\"")
     );
 
-static gboolean gst_rtpg729pay_setcaps (GstBaseRTPPayload * payload,
-    GstCaps * caps);
+static void
+gst_rtp_g729_pay_init (GstRTPG729Pay * pay, GstRTPG729PayClass * klass);
 
-GST_BOILERPLATE (GstRtpG729Pay, gst_rtpg729pay, GstBaseRTPAudioPayload,
+GST_BOILERPLATE (GstRTPG729Pay, gst_rtp_g729_pay, GstBaseRTPAudioPayload,
     GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
 
 static void
-gst_rtpg729pay_base_init (gpointer klass)
+gst_rtp_g729_pay_base_init (gpointer klass)
 {
   GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
 
   gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtpg729pay_sink_template));
+      gst_static_pad_template_get (&gst_rtp_g729_pay_sink_template));
   gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtpg729pay_src_template));
-  gst_element_class_set_details (element_class, &gst_rtpg729pay_details);
+      gst_static_pad_template_get (&gst_rtp_g729_pay_src_template));
+  gst_element_class_set_details (element_class, &gst_rtp_g729_pay_details);
 }
 
 static void
-gst_rtpg729pay_class_init (GstRtpG729PayClass * klass)
+gst_rtp_g729_pay_class_init (GstRTPG729PayClass * klass)
 {
-  GObjectClass *gobject_class;
-  GstElementClass *gstelement_class;
-  GstBaseRTPPayloadClass *gstbasertppayload_class;
+  GstBaseRTPPayloadClass *payload_class = GST_BASE_RTP_PAYLOAD_CLASS (klass);
 
-  gobject_class = (GObjectClass *) klass;
-  gstelement_class = (GstElementClass *) klass;
-  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
-
-  parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
-
-  gstbasertppayload_class->set_caps = gst_rtpg729pay_setcaps;
-
-  GST_DEBUG_CATEGORY_INIT (rtpg729pay_debug, "rtpg729pay", 0,
-      "G729 audio RTP payloader");
+  payload_class->set_caps = gst_rtp_g729_pay_set_caps;
+  payload_class->handle_buffer = gst_rtp_g729_pay_handle_buffer;
 }
 
 static void
-gst_rtpg729pay_init (GstRtpG729Pay * rtpg729pay, GstRtpG729PayClass * klass)
+gst_rtp_g729_pay_init (GstRTPG729Pay * pay, GstRTPG729PayClass * klass)
 {
-  GstBaseRTPPayload *basertppayload;
-  GstBaseRTPAudioPayload *basertpaudiopayload;
+  GstBaseRTPPayload *payload = GST_BASE_RTP_PAYLOAD (pay);
+  GstBaseRTPAudioPayload *audiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (pay);
 
-  basertppayload = GST_BASE_RTP_PAYLOAD (rtpg729pay);
-  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg729pay);
+  payload->pt = GST_RTP_PAYLOAD_G729;
+  gst_basertppayload_set_options (payload, "audio", FALSE, "G729", 8000);
 
-  /* we don't set the payload type, it should be set by the application using
-   * the pt property or the default 96 will be used */
-  basertppayload->clock_rate = 8000;
+  gst_base_rtp_audio_payload_set_frame_based (audiopayload);
+  gst_base_rtp_audio_payload_set_frame_options (audiopayload,
+      G729_FRAME_DURATION_MS, G729_FRAME_SIZE);
 
-  /* tell basertpaudiopayload that this is a frame based codec */
-  gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
-  gst_basertppayload_set_options (basertppayload, "audio", FALSE, "G729", 8000);
-  gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 10, 10);
 }
 
 static gboolean
-gst_rtpg729pay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
+gst_rtp_g729_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps)
 {
-  GstRtpG729Pay *rtpg729pay;
-  GstBaseRTPAudioPayload *basertpaudiopayload;
-  gboolean ret;
   GstStructure *structure;
-  const char *payload_name;
-
-  rtpg729pay = GST_RTP_G729_PAY (basertppayload);
-  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
+  gint pt;
 
   structure = gst_caps_get_structure (caps, 0);
+  if (!gst_structure_get_int (structure, "payload", &pt))
+    pt = GST_RTP_PAYLOAD_G729;
+
+  payload->pt = pt;
+  payload->dynamic = pt != GST_RTP_PAYLOAD_G729;
+
+  gst_basertppayload_set_outcaps (payload, NULL);
+
+  return TRUE;
+}
+
+static GstFlowReturn
+gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
+{
+  GstFlowReturn ret = GST_FLOW_OK;
+  GstBaseRTPAudioPayload *basertpaudiopayload =
+      GST_BASE_RTP_AUDIO_PAYLOAD (payload);
+  GstAdapter *adapter = NULL;
+  guint payload_len;
+  const guint8 *data = NULL;
+  guint available;
+  guint maxptime_octets = G_MAXUINT;
+  guint minptime_octets = 0;
+  guint min_payload_len;
+  guint max_payload_len;
+  gboolean use_adapter = FALSE;
+
+  available = GST_BUFFER_SIZE (buf);
 
-  payload_name = gst_structure_get_name (structure);
-  if (g_strcasecmp ("audio/G729", payload_name) != 0)
-    goto wrong_name;
+  if (available % G729_FRAME_SIZE != 0 &&
+      available % G729_FRAME_SIZE != G729B_CN_FRAME_SIZE)
+    goto invalid_size;
 
-  ret = gst_basertppayload_set_outcaps (basertppayload, NULL);
+  /* max number of bytes based on given ptime, has to be multiple of
+   * frame_duration */
+  if (payload->max_ptime != -1) {
+    guint ptime_ms = payload->max_ptime / 1000000;
+
+    maxptime_octets = G729_FRAME_SIZE *
+        (int) (ptime_ms / G729_FRAME_DURATION_MS);
+
+    if (maxptime_octets < G729_FRAME_SIZE) {
+      GST_WARNING_OBJECT (basertpaudiopayload, "Given ptime %d is smaller than"
+          " minimum %d ns, overwriting to minimum",
+          payload->max_ptime, G729_FRAME_DURATION_MS);
+      maxptime_octets = G729_FRAME_SIZE;
+    }
+  }
+
+  max_payload_len = MIN (
+      /* MTU max */
+      (int) (gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
+              (basertpaudiopayload), 0, 0) / G729_FRAME_SIZE) * G729_FRAME_SIZE,
+      /* ptime max */
+      maxptime_octets);
+
+  /* min number of bytes based on a given ptime, has to be a multiple
+     of frame duration */
+  {
+    guint64 min_ptime;
+
+    g_object_get (G_OBJECT (payload), "min-ptime", &min_ptime, NULL);
+
+    min_ptime = min_ptime / 1000000;
+    minptime_octets = G729_FRAME_SIZE *
+        (int) (min_ptime / G729_FRAME_DURATION_MS);
+  }
+
+  min_payload_len = MAX (minptime_octets, G729_FRAME_SIZE);
+
+  if (min_payload_len > max_payload_len) {
+    min_payload_len = max_payload_len;
+  }
+
+  GST_DEBUG_OBJECT (basertpaudiopayload,
+      "Calculated min_payload_len %u and max_payload_len %u",
+      min_payload_len, max_payload_len);
+
+  adapter = gst_base_rtp_audio_payload_get_adapter (basertpaudiopayload);
+
+  if (adapter && gst_adapter_available (adapter)) {
+    /* If there is always data in the adapter, we have to use it */
+    gst_adapter_push (adapter, buf);
+    available = gst_adapter_available (adapter);
+    use_adapter = TRUE;
+  } else {
+    /* let's set the base timestamp */
+    basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buf);
+
+    /* If buffer fits on an RTP packet, let's just push it through */
+    /* this will check against max_ptime and max_mtu */
+    if (GST_BUFFER_SIZE (buf) >= min_payload_len &&
+        GST_BUFFER_SIZE (buf) <= max_payload_len) {
+      ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
+          GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
+          GST_BUFFER_TIMESTAMP (buf));
+      gst_buffer_unref (buf);
+
+      return ret;
+    }
+
+    available = GST_BUFFER_SIZE (buf);
+    data = (guint8 *) GST_BUFFER_DATA (buf);
+  }
+
+  /* as long as we have full frames */
+  /* this loop will push all available buffers till the last frame */
+  while (available >= min_payload_len ||
+      available % G729_FRAME_SIZE == G729B_CN_FRAME_SIZE) {
+    guint num;
+
+    /* We send as much as we can */
+    if (available <= max_payload_len) {
+      payload_len = available;
+    } else {
+      payload_len = MIN (max_payload_len,
+          (available / G729_FRAME_SIZE) * G729_FRAME_SIZE);
+    }
+
+    if (use_adapter) {
+      data = gst_adapter_peek (adapter, payload_len);
+    }
+
+    ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, data,
+        payload_len, basertpaudiopayload->base_ts);
+
+    num = payload_len / G729_FRAME_SIZE;
+    basertpaudiopayload->base_ts += G729_FRAME_DURATION * num;
+
+    if (use_adapter) {
+      gst_adapter_flush (adapter, payload_len);
+      available = gst_adapter_available (adapter);
+    } else {
+      available -= payload_len;
+      data += payload_len;
+    }
+  }
+
+  if (!use_adapter) {
+    if (available != 0 && adapter) {
+      GstBuffer *buf2;
+      buf2 = gst_buffer_create_sub (buf,
+          GST_BUFFER_SIZE (buf) - available, available);
+      gst_adapter_push (adapter, buf2);
+    } else {
+      gst_buffer_unref (buf);
+    }
+  }
+
+  if (adapter) {
+    g_object_unref (adapter);
+  }
 
   return ret;
 
   /* ERRORS */
-wrong_name:
+invalid_size:
   {
-    GST_ERROR_OBJECT (rtpg729pay, "wrong name, expected 'audio/G729', got '%s'",
-        payload_name);
-    return FALSE;
+    GST_ELEMENT_ERROR (payload, STREAM, WRONG_TYPE,
+        ("Invalid input buffer size"),
+        ("Invalid buffer size, should be a multiple of"
+            " G729_FRAME_SIZE(10) with an optional G729B_CN_FRAME_SIZE(2)"
+            " added to it, but it is %u", available));
+    gst_buffer_unref (buf);
+    return GST_FLOW_ERROR;
   }
 }
 
diff --git a/gst/rtp/gstrtpg729pay.h b/gst/rtp/gstrtpg729pay.h
index e6a7da75..ce0e1d6b 100644
--- a/gst/rtp/gstrtpg729pay.h
+++ b/gst/rtp/gstrtpg729pay.h
@@ -1,4 +1,6 @@
 /* GStreamer
+ * Copyright (C) <2007> Nokia Corporation
+ * Copyright (C) <2007> Collabora Ltd
  *
  * This library is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Library General Public
@@ -25,25 +27,25 @@
 G_BEGIN_DECLS
 
 #define GST_TYPE_RTP_G729_PAY \
-  (gst_rtpg729pay_get_type())
+  (gst_rtp_g729_pay_get_type())
 #define GST_RTP_G729_PAY(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_G729_PAY,GstRtpG729Pay))
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_G729_PAY,GstRTPG729Pay))
 #define GST_RTP_G729_PAY_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_G729_PAY,GstRtpG729PayClass))
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_G729_PAY,GstRTPG729PayClass))
 #define GST_IS_RTP_G729_PAY(obj) \
   (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_G729_PAY))
 #define GST_IS_RTP_G729_PAY_CLASS(klass) \
   (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_G729_PAY))
 
-typedef struct _GstRtpG729Pay GstRtpG729Pay;
-typedef struct _GstRtpG729PayClass GstRtpG729PayClass;
+typedef struct _GstRTPG729Pay GstRTPG729Pay;
+typedef struct _GstRTPG729PayClass GstRTPG729PayClass;
 
-struct _GstRtpG729Pay
+struct _GstRTPG729Pay
 {
   GstBaseRTPAudioPayload audiopayload;
 };
 
-struct _GstRtpG729PayClass
+struct _GstRTPG729PayClass
 {
   GstBaseRTPAudioPayloadClass parent_class;
 };