WEBRTC_TRANSCEIVER_DIRECTION_SENDRECV, /**< Send and receive */
} webrtc_transceiver_direction_e;
+/**
+ * @brief Enumeration for WebRTC transceiver codec.
+ * @since_tizen 7.0
+ */
+typedef enum {
+ WEBRTC_TRANSCEIVER_CODEC_PCMU = 0x00000100 | 0x01, /**< PCMU audio codec */
+ WEBRTC_TRANSCEIVER_CODEC_PCMA = 0x00000100 | 0x02, /**< PCMA audio codec */
+ WEBRTC_TRANSCEIVER_CODEC_OPUS = 0x00000100 | 0x03, /**< OPUS audio codec */
+ WEBRTC_TRANSCEIVER_CODEC_VP8 = 0x00000200 | 0x01, /**< VP8 video codec */
+ WEBRTC_TRANSCEIVER_CODEC_VP9 = 0x00000200 | 0x02, /**< VP9 video codec */
+ WEBRTC_TRANSCEIVER_CODEC_H264 = 0x00000200 | 0x03, /**< H264 video codec */
+} webrtc_transceiver_codec_e;
+
/**
* @}
*/
*/
typedef void (*webrtc_media_packet_source_buffer_state_changed_cb)(unsigned int source_id, webrtc_media_packet_source_buffer_state_e state, void *user_data);
+
+/**
+ * @brief Called iteratively to report all the supported transceiver codecs.
+ * @since_tizen 7.0
+ * @param[in] codec The supported codec
+ * @param[in] user_data The user data passed from the callback registration function
+ * @return @c true to continue with the next iteration of the loop,
+ * otherwise @c false to break out of the loop
+ * @see webrtc_media_source_foreach_supported_transceiver_codec()
+ */
+typedef bool (*webrtc_media_source_supported_transceiver_codec_cb)(webrtc_transceiver_codec_e codec, void *user_data);
+
/**
* @}
*/
*/
int webrtc_media_source_get_transceiver_direction(webrtc_h webrtc, unsigned int source_id, webrtc_media_type_e media_type, webrtc_transceiver_direction_e *direction);
+/**
+ * @brief Retrieves all the supported transceiver codecs.
+ * @since_tizen 7.0
+ * @remarks If @a source_id is a media source of #WEBRTC_MEDIA_SOURCE_TYPE_FILE or #WEBRTC_MEDIA_SOURCE_TYPE_MEDIA_PACKET,
+ * this function will return #WEBRTC_ERROR_INVALID_PARAMETER.
+ * @param[in] webrtc WebRTC handle
+ * @param[in] source_id The media source id
+ * @param[in] media_type The media type
+ * @param[in] callback Callback function pointer
+ * @param[in] user_data The user data to be passed to the callback function
+ * @return @c 0 on success,
+ * otherwise a negative error value
+ * @retval #WEBRTC_ERROR_NONE Successful
+ * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
+ * @retval #WEBRTC_ERROR_INVALID_OPERATION Invalid operation
+ * @see webrtc_media_source_supported_transceiver_codec_cb()
+ */
+int webrtc_media_source_foreach_supported_transceiver_codec(webrtc_h webrtc, unsigned int source_id, webrtc_media_type_e media_type, webrtc_media_source_supported_transceiver_codec_cb callback, void *user_data);
+
/**
* @brief Sets pause to the media source.
* @since_tizen 6.5
return _get_transceiver_direction(webrtc, source_id, media_type, direction);
}
+int webrtc_media_source_foreach_supported_transceiver_codec(webrtc_h webrtc, unsigned int source_id, webrtc_media_type_e media_type, webrtc_media_source_supported_transceiver_codec_cb callback, void *user_data)
+{
+ g_autoptr(GMutexLocker) locker = NULL;
+ webrtc_s *_webrtc = (webrtc_s *)webrtc;
+
+ RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+ RET_VAL_IF(callback == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "callback is NULL");
+
+ locker = g_mutex_locker_new(&_webrtc->mutex);
+
+ return _foreach_supported_transceiver_codec(_webrtc, source_id, media_type, callback, user_data);
+}
+
int webrtc_media_source_set_pause(webrtc_h webrtc, unsigned int source_id, webrtc_media_type_e media_type, bool pause)
{
g_autoptr(GMutexLocker) locker = NULL;
return WEBRTC_ERROR_NONE;
}
+static int __convert_audio_codec(const char *codec_name, webrtc_transceiver_codec_e *codec)
+{
+ RET_VAL_IF(codec_name == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "codec_name is NULL");
+ RET_VAL_IF(codec == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "codec is NULL");
+
+ if (!strcmp(codec_name, "pcmu") || !strcmp(codec_name, "PCMU")) {
+ *codec = WEBRTC_TRANSCEIVER_CODEC_PCMU;
+ return WEBRTC_ERROR_NONE;
+ }
+ if (!strcmp(codec_name, "pcma") || !strcmp(codec_name, "PCMA")) {
+ *codec = WEBRTC_TRANSCEIVER_CODEC_PCMA;
+ return WEBRTC_ERROR_NONE;
+ }
+ if (!strcmp(codec_name, "opus") || !strcmp(codec_name, "OPUS")) {
+ *codec = WEBRTC_TRANSCEIVER_CODEC_OPUS;
+ return WEBRTC_ERROR_NONE;
+ }
+
+ LOG_ERROR("not supported audio codec_name[%s]", codec_name);
+
+ return WEBRTC_ERROR_INVALID_PARAMETER;
+}
+
+static int __convert_video_codec(const char *codec_name, webrtc_transceiver_codec_e *codec)
+{
+ RET_VAL_IF(codec_name == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "codec_name is NULL");
+ RET_VAL_IF(codec == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "codec is NULL");
+
+ if (!strcmp(codec_name, "vp8") || !strcmp(codec_name, "VP8")) {
+ *codec = WEBRTC_TRANSCEIVER_CODEC_VP8;
+ return WEBRTC_ERROR_NONE;
+ }
+ if (!strcmp(codec_name, "vp9") || !strcmp(codec_name, "VP9")) {
+ *codec = WEBRTC_TRANSCEIVER_CODEC_VP9;
+ return WEBRTC_ERROR_NONE;
+ }
+ if (!strcmp(codec_name, "h264") || !strcmp(codec_name, "H264")) {
+ *codec = WEBRTC_TRANSCEIVER_CODEC_H264;
+ return WEBRTC_ERROR_NONE;
+ }
+
+ LOG_ERROR("not supported video codec_name[%s]", codec_name);
+
+ return WEBRTC_ERROR_INVALID_PARAMETER;
+}
+
+typedef int (*convert_codec_func)(const char *codec_name, webrtc_transceiver_codec_e *codec);
+
+static convert_codec_func convert_codec_funcs[] = {
+ [WEBRTC_MEDIA_TYPE_AUDIO] = __convert_audio_codec,
+ [WEBRTC_MEDIA_TYPE_VIDEO] = __convert_video_codec,
+};
+
+int _foreach_supported_transceiver_codec(webrtc_s *webrtc, unsigned int source_id, webrtc_media_type_e media_type, webrtc_media_source_supported_transceiver_codec_cb callback, void *user_data)
+{
+ int ret;
+ webrtc_gst_slot_s *source;
+ const ini_item_media_source_s *ini_source;
+ webrtc_transceiver_codec_e codec;
+ GStrv codecs;
+ guint i;
+
+ RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+ RET_VAL_IF(callback == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "callback is NULL");
+ RET_VAL_IF(webrtc->gst.webrtcbin == NULL, WEBRTC_ERROR_INVALID_OPERATION, "webrtcbin is NULL");
+ RET_VAL_IF(webrtc->gst.source_slots == NULL, WEBRTC_ERROR_INVALID_OPERATION, "source_slots is NULL");
+ RET_VAL_IF((source = _get_slot_by_id(webrtc->gst.source_slots, source_id)) == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
+ RET_VAL_IF((source->type == WEBRTC_MEDIA_SOURCE_TYPE_FILE), WEBRTC_ERROR_INVALID_PARAMETER, "this API does not support the file source");
+ RET_VAL_IF((source->type == WEBRTC_MEDIA_SOURCE_TYPE_MEDIA_PACKET), WEBRTC_ERROR_INVALID_PARAMETER, "this API does not support the media packet source");
+
+ ini_source = _ini_get_source_by_type(&webrtc->ini, source->type);
+ RET_VAL_IF(ini_source == NULL, WEBRTC_ERROR_INVALID_OPERATION, "ini_source is NULL");
+
+ if (media_type == WEBRTC_MEDIA_TYPE_AUDIO && source->media_types & MEDIA_TYPE_AUDIO) {
+ codecs = ini_source->a_codecs;
+
+ } else if (media_type == WEBRTC_MEDIA_TYPE_VIDEO && source->media_types & MEDIA_TYPE_VIDEO) {
+ codecs = ini_source->v_codecs;
+
+ } else {
+ LOG_ERROR("invalid media_type[%d] for source[media_types:0x%x, id:%u]", media_type, source->media_types, source_id);
+ return WEBRTC_ERROR_INVALID_PARAMETER;
+ }
+
+ LOG_INFO("webrtc[%p] source_id[%u] media_type[%d] callback[%p] user_data[%p]", webrtc, source_id, media_type, callback, user_data);
+
+ for (i = 0; i < g_strv_length(codecs); i++) {
+ if ((ret = convert_codec_funcs[media_type](codecs[i], &codec)) != WEBRTC_ERROR_NONE)
+ return ret;
+ LOG_INFO(" - supported %s codec: %s", media_type == WEBRTC_MEDIA_TYPE_AUDIO ? "audio" : "video", codecs[i]);
+ if (!callback(codec, user_data)) {
+ LOG_DEBUG("stop foreach callback");
+ break;
+ }
+ }
+
+ return WEBRTC_ERROR_NONE;
+}
+
int _set_pause(webrtc_s *webrtc, unsigned int source_id, webrtc_media_type_e media_type, bool pause)
{
webrtc_gst_slot_s *source;
projects / platform / core / api / webrtc.git / commitdiff