mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-01-12 02:15:31 +00:00
g726pay: add RFC compliant packetizing
Shuffle the input bits according to RFC 3551 for G726 payloads. Add option to force the previous behaviour. Fixes #567140
This commit is contained in:
Wim Taymans
parent
e213debe8e
commit
84c3d1a2c8
2 changed files with 269 additions and 8 deletions
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@ -30,6 +30,18 @@
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#include "gstrtpg726pay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpg726pay_debug);
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#define GST_CAT_DEFAULT (rtpg726pay_debug)
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#define DEFAULT_FORCE_AAL2 TRUE
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enum
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{
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PROP_0,
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PROP_FORCE_AAL2,
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PROP_LAST
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};
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static const GstElementDetails gst_rtp_g726_pay_details =
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GST_ELEMENT_DETAILS ("RTP G.726 payloader",
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"Codec/Payloader/Network",
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@ -55,11 +67,19 @@ GST_STATIC_PAD_TEMPLATE ("src",
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 8000, "
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"encoding-name = (string) { \"G726-16\", \"G726-24\", \"G726-32\", \"G726-40\" } ")
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"encoding-name = (string) { \"G726-16\", \"G726-24\", \"G726-32\", \"G726-40\", "
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" \"AAL2-G726-16\", \"AAL2-G726-24\", \"AAL2-G726-32\", \"AAL2-G726-40\" } ")
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);
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static void gst_rtp_g726_pay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_rtp_g726_pay_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static gboolean gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_g726_pay_handle_buffer (GstBaseRTPPayload *
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payload, GstBuffer * buffer);
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GST_BOILERPLATE (GstRtpG726Pay, gst_rtp_g726_pay, GstBaseRTPAudioPayload,
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
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@ -83,9 +103,19 @@ gst_rtp_g726_pay_class_init (GstRtpG726PayClass * klass)
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->set_property = gst_rtp_g726_pay_set_property;
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gobject_class->get_property = gst_rtp_g726_pay_get_property;
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_FORCE_AAL2,
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g_param_spec_boolean ("force-aal2", "Force AAL2",
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"Force AAL2 encoding for compatibility with bad depayloaders",
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DEFAULT_FORCE_AAL2, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gstbasertppayload_class->set_caps = gst_rtp_g726_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_g726_pay_handle_buffer;
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GST_DEBUG_CATEGORY_INIT (rtpg726pay_debug, "rtpg726pay", 0,
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"G.726 RTP Payloader");
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}
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static void
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@ -97,6 +127,8 @@ gst_rtp_g726_pay_init (GstRtpG726Pay * rtpg726pay, GstRtpG726PayClass * klass)
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GST_BASE_RTP_PAYLOAD (rtpg726pay)->clock_rate = 8000;
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rtpg726pay->force_aal2 = DEFAULT_FORCE_AAL2;
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/* sample based codec */
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gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
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}
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@ -105,16 +137,25 @@ static gboolean
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gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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{
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gchar *encoding_name;
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GstStructure *structure = gst_caps_get_structure (caps, 0);
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GstStructure *structure;
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GstBaseRTPAudioPayload *basertpaudiopayload;
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gint bitrate;
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GstRtpG726Pay *pay;
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GstCaps *peercaps;
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (payload);
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pay = GST_RTP_G726_PAY (payload);
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if (!gst_structure_get_int (structure, "bitrate", &bitrate))
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bitrate = 32000;
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structure = gst_caps_get_structure (caps, 0);
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switch (bitrate) {
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if (!gst_structure_get_int (structure, "bitrate", &pay->bitrate))
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pay->bitrate = 32000;
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GST_DEBUG_OBJECT (payload, "using bitrate %d", pay->bitrate);
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pay->aal2 = FALSE;
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/* first see what we can do with the bitrate */
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switch (pay->bitrate) {
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case 16000:
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encoding_name = g_strdup ("G726-16");
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gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
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@ -139,6 +180,65 @@ gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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goto invalid_bitrate;
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}
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GST_DEBUG_OBJECT (payload, "selected base encoding %s", encoding_name);
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/* now see if we need to produce AAL2 or not */
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peercaps = gst_pad_peer_get_caps (payload->srcpad);
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if (peercaps) {
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GstCaps *filter, *intersect;
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gchar *capsstr;
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GST_DEBUG_OBJECT (payload, "have peercaps %" GST_PTR_FORMAT, peercaps);
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capsstr = g_strdup_printf ("application/x-rtp, "
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"media = (string) \"audio\", "
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"clock-rate = (int) 8000, "
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"encoding-name = (string) %s; "
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"clock-rate = (int) 8000, "
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"encoding-name = (string) AAL2-%s", encoding_name, encoding_name);
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filter = gst_caps_from_string (capsstr);
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g_free (capsstr);
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/* intersect to filter */
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intersect = gst_caps_intersect (peercaps, filter);
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gst_caps_unref (peercaps);
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GST_DEBUG_OBJECT (payload, "intersected to %" GST_PTR_FORMAT, intersect);
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if (!intersect)
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goto no_format;
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if (gst_caps_is_empty (intersect)) {
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gst_caps_unref (intersect);
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goto no_format;
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}
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structure = gst_caps_get_structure (intersect, 0);
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/* now see what encoding name we settled on, we need to dup because the
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* string goes away when we unref the intersection below. */
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encoding_name =
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g_strdup (gst_structure_get_string (structure, "encoding-name"));
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/* if we managed to negotiate to AAL2, we definatly are going to do AAL2
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* encoding. Else we only encode AAL2 when explicitly set by the
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* property. */
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if (g_str_has_prefix (encoding_name, "AAL2-"))
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pay->aal2 = TRUE;
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else
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pay->aal2 = pay->force_aal2;
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GST_DEBUG_OBJECT (payload, "final encoding %s, AAL2 %d", encoding_name,
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pay->aal2);
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gst_caps_unref (intersect);
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} else {
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/* downstream can do anything but we prefer the better supported non-AAL2 */
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pay->aal2 = pay->force_aal2;
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GST_DEBUG_OBJECT (payload, "no peer caps, AAL2 %d", pay->aal2);
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}
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gst_basertppayload_set_options (payload, "audio", TRUE, encoding_name, 8000);
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gst_basertppayload_set_outcaps (payload, NULL);
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@ -149,9 +249,166 @@ gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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/* ERRORS */
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invalid_bitrate:
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{
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GST_ERROR_OBJECT (payload, "invalid bitrate %d specified", bitrate);
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GST_ERROR_OBJECT (payload, "invalid bitrate %d specified", pay->bitrate);
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return FALSE;
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}
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no_format:
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{
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GST_ERROR_OBJECT (payload, "could not negotiate format");
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return FALSE;
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}
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}
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static GstFlowReturn
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gst_rtp_g726_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer)
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{
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GstFlowReturn res;
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GstRtpG726Pay *pay;
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pay = GST_RTP_G726_PAY (payload);
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if (!pay->aal2) {
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guint8 *data, tmp;
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guint len;
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/* for non AAL2, we need to reshuffle the bytes, we can do this in-place
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* when the buffer is writable. */
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buffer = gst_buffer_make_writable (buffer);
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data = GST_BUFFER_DATA (buffer);
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len = GST_BUFFER_SIZE (buffer);
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GST_LOG_OBJECT (pay, "packing %u bytes of data", len);
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/* we need to reshuffle the bytes, output is of the form:
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* A B C D .. with the number of bits depending on the bitrate. */
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switch (pay->bitrate) {
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case 16000:
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{
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/* 0
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* 0 1 2 3 4 5 6 7
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* +-+-+-+-+-+-+-+-+-
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* |D D|C C|B B|A A| ...
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* |0 1|0 1|0 1|0 1|
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* +-+-+-+-+-+-+-+-+-
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*/
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while (len > 0) {
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tmp = *data;
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*data++ = ((tmp & 0xc0) >> 6) |
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((tmp & 0x30) >> 2) | ((tmp & 0x0c) << 2) | ((tmp & 0x03) << 6);
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len--;
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}
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break;
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}
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case 24000:
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{
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/* 0 1 2
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* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
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* |C C|B B B|A A A|F|E E E|D D D|C|H H H|G G G|F F| ...
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* |1 2|0 1 2|0 1 2|2|0 1 2|0 1 2|0|0 1 2|0 1 2|0 1|
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
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*/
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while (len > 2) {
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tmp = *data;
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*data++ = ((tmp & 0xc0) >> 6) |
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((tmp & 0x38) >> 1) | ((tmp & 0x07) << 5);
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tmp = *data;
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*data++ = ((tmp & 0x80) >> 7) |
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((tmp & 0x70) >> 3) | ((tmp & 0x0e) << 4) | ((tmp & 0x01) << 7);
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tmp = *data;
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*data++ = ((tmp & 0xe0) >> 5) |
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((tmp & 0x1c) >> 2) | ((tmp & 0x03) << 6);
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len -= 3;
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}
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break;
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}
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case 32000:
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{
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/* 0 1
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* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
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* |B B B B|A A A A|D D D D|C C C C| ...
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* |0 1 2 3|0 1 2 3|0 1 2 3|0 1 2 3|
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
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*/
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while (len > 0) {
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tmp = *data;
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*data++ = ((tmp & 0xf0) >> 4) | ((tmp & 0x0f) << 4);
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len--;
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}
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break;
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}
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case 40000:
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{
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/* 0 1 2 3 4
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* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
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* |B B B|A A A A A|D|C C C C C|B B|E E E E|D D D D|G G|F F F F F|E|H H H H H|G G G|
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* |2 3 4|0 1 2 3 4|4|0 1 2 3 4|0 1|1 2 3 4|0 1 2 3|3 4|0 1 2 3 4|0|0 1 2 3 4|0 1 2|
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
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*/
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while (len > 4) {
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tmp = *data;
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*data++ = ((tmp & 0xe0) >> 5) | ((tmp & 0x1f) << 3);
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tmp = *data;
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*data++ = ((tmp & 0x80) >> 7) |
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((tmp & 0x7c) >> 2) | ((tmp & 0x03) << 6);
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tmp = *data;
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*data++ = ((tmp & 0xf0) >> 4) | ((tmp & 0x0f) << 4);
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tmp = *data;
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*data++ = ((tmp & 0xc0) >> 6) |
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((tmp & 0x3e) << 2) | ((tmp & 0x01) << 7);
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tmp = *data;
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*data++ = ((tmp & 0xf8) >> 3) | ((tmp & 0x07) << 5);
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len -= 5;
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}
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break;
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}
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}
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}
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res =
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GST_BASE_RTP_PAYLOAD_CLASS (parent_class)->handle_buffer (payload,
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buffer);
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return res;
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}
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static void
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gst_rtp_g726_pay_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstRtpG726Pay *rtpg726pay;
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rtpg726pay = GST_RTP_G726_PAY (object);
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switch (prop_id) {
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case PROP_FORCE_AAL2:
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rtpg726pay->force_aal2 = g_value_get_boolean (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtp_g726_pay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstRtpG726Pay *rtpg726pay;
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rtpg726pay = GST_RTP_G726_PAY (object);
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switch (prop_id) {
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case PROP_FORCE_AAL2:
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g_value_set_boolean (value, rtpg726pay->force_aal2);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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gboolean
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@ -36,6 +36,10 @@ typedef struct _GstRtpG726PayClass GstRtpG726PayClass;
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struct _GstRtpG726Pay
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{
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GstBaseRTPAudioPayload audiopayload;
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gboolean aal2;
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gboolean force_aal2;
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gint bitrate;
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};
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struct _GstRtpG726PayClass
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