#include "webrtc.h"
#include "webrtc_private.h"
#include <tbm_surface_internal.h>
+#include <media_packet_internal.h>
#define GST_KLASS_NAME_ENCODER_AUDIO "Codec/Encoder/Audio"
#define GST_KLASS_NAME_ENCODER_VIDEO "Codec/Encoder/Video"
return WEBRTC_ERROR_NONE;
}
+static bool __is_valid_encoded_format(media_type_e type, media_format_h format)
+{
+ int ret;
+ media_format_mimetype_e mime_type;
+
+ RET_VAL_IF(format == NULL, false, "format is NULL");
+
+ if (type == MEDIA_TYPE_AUDIO) {
+ ret = media_format_get_audio_info(format, &mime_type, NULL, NULL, NULL, NULL);
+ RET_VAL_IF(ret != MEDIA_PACKET_ERROR_NONE, false, "failed to media_format_get_audio_info()");
+
+ } else if (type == MEDIA_TYPE_VIDEO) {
+ ret = media_format_get_video_info(format, &mime_type, NULL, NULL, NULL, NULL);
+ RET_VAL_IF(ret != MEDIA_PACKET_ERROR_NONE, false, "failed to media_format_get_video_info()");
+
+ } else {
+ LOG_ERROR_IF_REACHED("type(%u)", type);
+ return false;
+ }
+
+ return (mime_type & MEDIA_FORMAT_ENCODED);
+}
+
int _push_media_packet(webrtc_s *webrtc, unsigned int source_id, media_packet_h packet)
{
int ret = WEBRTC_ERROR_NONE;
webrtc_gst_slot_s *source;
- guchar *data_ptr;
GstElement *appsrc;
+ GstBuffer *buffer = NULL;
GstBuffer *new_buffer;
guint64 pts = 0;
guint64 dts = 0;
RET_VAL_IF(source->media_format == NULL, WEBRTC_ERROR_INVALID_OPERATION, "media_format is NULL");
appsrc = __find_element_in_bin(source->bin, "appsrc");
- if (appsrc == NULL) {
- LOG_ERROR("source_id[%u] is not for media packet source", source_id);
- return WEBRTC_ERROR_INVALID_PARAMETER;
- }
+ RET_VAL_IF(appsrc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source_id[%u] is not for media packet source", source_id);
if (packet == NULL) {
LOG_INFO("packet is NULL, emit EOS signal");
return WEBRTC_ERROR_NONE;
}
- media_packet_get_buffer_data_ptr(packet, (void **)&data_ptr);
- if (data_ptr == NULL) {
- LOG_ERROR("invalid packet, data_ptr is NULL");
- return WEBRTC_ERROR_INVALID_OPERATION;
+ /* FIXME: we assume that extra field has gstreamer buffer pointer */
+ ret = media_packet_get_extra(packet, (void **)&buffer);
+ RET_VAL_IF(ret != MEDIA_PACKET_ERROR_NONE, WEBRTC_ERROR_INVALID_OPERATION, "failed to media_packet_get_extra()");
+ if (buffer) {
+ media_format_h format;
+ /* allow only when both media packet and appsrc are for encoded data */
+ if (!__is_valid_encoded_format(source->media_types, source->media_format)) {
+ LOG_ERROR("media packet source is not for encoded format");
+ return WEBRTC_ERROR_INVALID_OPERATION;
+ }
+ ret = media_packet_get_format(packet, &format);
+ RET_VAL_IF(ret != MEDIA_PACKET_ERROR_NONE, WEBRTC_ERROR_INVALID_OPERATION, "failed to media_packet_get_format()");
+ media_format_unref(format);
+ if (!__is_valid_encoded_format(source->media_types, format)) {
+ LOG_ERROR("invalid format of packet");
+ return WEBRTC_ERROR_INVALID_OPERATION;
+ }
+
+ /* FIXME: need more API to check if it is valid gstreamer buffer. */
+ LOG_DEBUG("external gst buffer[%p]", buffer);
+ g_signal_emit_by_name(G_OBJECT(appsrc), "push-buffer", buffer, &gst_ret, NULL);
+ if (gst_ret != GST_FLOW_OK) {
+ LOG_ERROR("failed to 'push-buffer', gst_ret[0x%x]", gst_ret);
+ return WEBRTC_ERROR_INVALID_OPERATION;
+ }
+ return WEBRTC_ERROR_NONE;
}
- media_packet_get_buffer_size(packet, &size);
+ /* FIXME: make subfunction for codes below */
+ ret = media_packet_get_buffer_size(packet, &size);
+ RET_VAL_IF(ret != MEDIA_PACKET_ERROR_NONE, WEBRTC_ERROR_INVALID_OPERATION, "failed to media_packet_get_buffer_size()");
new_buffer = gst_buffer_new_and_alloc(size);
- if (!new_buffer) {
- LOG_ERROR("failed to gst_buffer_new_and_alloc()");
- return WEBRTC_ERROR_INVALID_OPERATION;
- }
+ RET_VAL_IF(new_buffer == NULL, WEBRTC_ERROR_INVALID_OPERATION, "failed to gst_buffer_new_and_alloc()");
ret = __fill_gst_buffer_mapped_data_from_packet(new_buffer, packet);
if (ret != WEBRTC_ERROR_NONE) {
LOG_ERROR("failed to __fill_gst_buffer_mapped_data_from_packet()");
- return ret;
+ goto exit;
}
- media_packet_get_pts(packet, &pts);
+ ret = media_packet_get_pts(packet, &pts);
+ if (ret != MEDIA_PACKET_ERROR_NONE) {
+ LOG_ERROR("failed to media_packet_get_pts()");
+ ret = WEBRTC_ERROR_INVALID_OPERATION;
+ goto exit;
+ }
GST_BUFFER_PTS(new_buffer) = pts;
- media_packet_get_dts(packet, &dts);
+
+ ret = media_packet_get_dts(packet, &dts);
+ if (ret != MEDIA_PACKET_ERROR_NONE) {
+ LOG_ERROR("failed to media_packet_get_dts()");
+ ret = WEBRTC_ERROR_INVALID_OPERATION;
+ goto exit;
+ }
GST_BUFFER_DTS(new_buffer) = dts;
- media_packet_get_duration(packet, &duration);
+ ret = media_packet_get_duration(packet, &duration);
+ if (ret != MEDIA_PACKET_ERROR_NONE) {
+ LOG_ERROR("failed to media_packet_get_duration()");
+ ret = WEBRTC_ERROR_INVALID_OPERATION;
+ goto exit;
+ }
GST_BUFFER_DURATION(new_buffer) = duration;
LOG_DEBUG("new gst buffer[%p, pts:%llu, dts:%llu, duration:%llu]", new_buffer, pts, dts, duration);
g_signal_emit_by_name(G_OBJECT(appsrc), "push-buffer", new_buffer, &gst_ret, NULL);
- gst_buffer_unref(new_buffer);
-
if (gst_ret != GST_FLOW_OK) {
LOG_ERROR("failed to 'push-buffer', gst_ret[0x%x]", gst_ret);
- return WEBRTC_ERROR_INVALID_OPERATION;
+ ret = WEBRTC_ERROR_INVALID_OPERATION;
}
- return WEBRTC_ERROR_NONE;
+exit:
+ gst_buffer_unref(new_buffer);
+
+ return ret;
}
#include <webrtc.h>
#include <webrtc_internal.h>
#include <media_format.h>
+#include <media_packet_internal.h>
#include <appcore-efl.h>
#include <Elementary.h>
#include <tbm_surface_internal.h>
#define MAX_CONNECTION_LEN 4
#define MAX_MEDIA_PACKET_SOURCE_LEN 4
+#define USE_GSTBUFFER_WITHOUT_COPY true
+
enum {
CURRENT_STATUS_MAINMENU,
CURRENT_STATUS_ADD_MEDIA_SOURCE,
static int __media_packet_finalize_cb(media_packet_h packet, int error_code, void *user_data)
{
+ GstBuffer *buffer = NULL;
+
g_print("__media_packet_finalize_cb(), packet[%p], error_code[0x%x], user_data[%p]\n", packet, error_code, user_data);
+ media_packet_get_extra(packet, (void **)&buffer);
+ if (buffer) {
+ g_print("unref gstbuffer[%p]\n", buffer);
+ gst_buffer_unref(buffer);
+ }
+
return MEDIA_PACKET_FINALIZE;
}
-static media_packet_h __make_media_packet(media_packet_source_s *packet_source, GstBuffer *buffer)
+static media_packet_h __make_media_packet(media_packet_source_s *packet_source, GstBuffer *buffer, bool set_buffer_as_external_memory)
{
int ret;
media_packet_h packet = NULL;
return NULL;
}
+ if (set_buffer_as_external_memory) {
+ gst_buffer_ref(buffer);
+
+ if (!gst_buffer_map(buffer, &buff_info, GST_MAP_READ)) {
+ g_printerr("failed to gst_buffer_map()\n");
+ gst_buffer_unref(buffer);
+ return NULL;
+ }
+
+ ret = media_packet_create_from_external_memory(packet_source->format, buff_info.data, gst_buffer_get_size(buffer), __media_packet_finalize_cb, packet_source, &packet);
+ if (ret != MEDIA_PACKET_ERROR_NONE) {
+ g_printerr("failed to media_packet_create_from_external_memory()\n");
+ gst_buffer_unmap(buffer, &buff_info);
+ gst_buffer_unref(buffer);
+ return NULL;
+ }
+ gst_buffer_unmap(buffer, &buff_info);
+
+ ret |= media_packet_set_pts(packet, GST_BUFFER_PTS(buffer));
+ ret |= media_packet_set_dts(packet, GST_BUFFER_DTS(buffer));
+ ret |= media_packet_set_duration(packet, GST_BUFFER_DURATION(buffer));
+
+ /* FIXME: We put the gstbuffer to extra field of media packet. It does not guarantee the validity of the gstbuffer
+ * after destroying this media packet. Currently, the gstbuffer must be used before destroying the packet. */
+ ret |= media_packet_set_extra(packet, (void *)buffer);
+ if (ret != MEDIA_PACKET_ERROR_NONE) {
+ g_printerr("failed to media_packet_set_*()\n");
+ gst_buffer_unref(buffer);
+ return NULL;
+ }
+
+ return packet;
+ }
+
ret = media_packet_create_alloc(packet_source->format, __media_packet_finalize_cb, packet_source, &packet);
if (ret != MEDIA_PACKET_ERROR_NONE) {
g_printerr("failed to media_packet_create_alloc()\n");
goto error;
}
+ gst_buffer_unref(buffer);
+
return packet;
error:
guint64 dts = 0;
guint64 duration = 0;
guint64 size = 0;
+ GstBuffer *buffer = NULL;
+
+ media_packet_get_extra(packet, (void **)&buffer);
+ if (buffer) {
+ g_print("external gstbuffer[%p]\n", buffer);
+ return buffer;
+ }
media_packet_get_buffer_data_ptr(packet, (void **)&data_ptr);
if (data_ptr == NULL) {
g_free(caps_str);
gst_caps_unref(caps);
- packet = __make_media_packet(packet_source, buffer);
+ packet = __make_media_packet(packet_source, buffer, USE_GSTBUFFER_WITHOUT_COPY);
if (!packet) {
g_printerr("failed to __make_media_packet()\n");
return;
g_signal_emit_by_name(G_OBJECT(packet_source->appsrc), "push-buffer", buffer_from_packet, &gst_ret, NULL);
if (gst_ret != GST_FLOW_OK)
g_printerr("failed to 'push-buffer', gst_ret[0x%x]\n", gst_ret);
-
- gst_buffer_unref(buffer_from_packet);
-
#else
if (webrtc_media_packet_source_push_packet(packet_source->webrtc, packet_source->source_id, packet) != WEBRTC_ERROR_NONE)
g_printerr("failed to webrtc_media_packet_source_push_packet()\n");
media_type = gst_structure_get_name(gst_caps_get_structure(gst_pad_get_current_caps(pad), 0));
if (g_strrstr(media_type, "audio")) {
- g_print("skip pad for audio");
+ g_print("skip pad for audio\n");
return;
}
projects / platform / core / api / webrtc.git / commitdiff