*/
int webrtc_media_source_get_transceiver_codec(webrtc_h webrtc, unsigned int source_id, webrtc_media_type_e media_type, webrtc_transceiver_codec_e *codec);
+/**
+ * @brief Sets the RTP payload type to the media source.
+ * @since_tizen 9.0
+ * @remarks This function would be used to set a specific payload type of a codec complying with a remote peer's offer description.
+ * @param[in] webrtc WebRTC handle
+ * @param[in] source_id The media source id
+ * @param[in] codec The transceiver codec
+ * @param[in] pt The RTP payload type
+ * @return @c 0 on success,
+ * otherwise a negative error value
+ * @retval #WEBRTC_ERROR_NONE Successful
+ * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
+ * @pre Add media source to @a webrtc to get @a source_id by calling webrtc_add_media_source().
+ * @see webrtc_media_source_get_payload_type()
+ * @see webrtc_media_source_get_transceiver_codec()
+ * @see webrtc_media_source_set_transceiver_codec()
+ * @see webrtc_media_source_foreach_supported_transceiver_codec()
+ */
+int webrtc_media_source_set_payload_type(webrtc_h webrtc, unsigned int source_id, webrtc_transceiver_codec_e codec, unsigned int pt);
+
+/**
+ * @brief Gets the RTP payload type of the media source.
+ * @since_tizen 9.0
+ * @param[in] webrtc WebRTC handle
+ * @param[in] source_id The media source id
+ * @param[in] codec The transceiver codec
+ * @param[out] pt The RTP payload type
+ * @return @c 0 on success,
+ * otherwise a negative error value
+ * @retval #WEBRTC_ERROR_NONE Successful
+ * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
+ * @pre Add media source to @a webrtc to get @a source_id by calling webrtc_add_media_source().
+ * @see webrtc_media_source_set_payload_type()
+ * @see webrtc_media_source_get_transceiver_codec()
+ * @see webrtc_media_source_set_transceiver_codec()
+ * @see webrtc_media_source_foreach_supported_transceiver_codec()
+ */
+int webrtc_media_source_get_payload_type(webrtc_h webrtc, unsigned int source_id, webrtc_transceiver_codec_e codec, unsigned int *pt);
+
/**
* @brief Sets pause to the media source.
* @since_tizen 6.5
*/
int webrtc_add_media_source_internal(webrtc_h webrtc, webrtc_media_source_type_internal_e type, unsigned int *source_id);
-/**
- * @internal
- * @brief Sets the RTP payload type to the media source.
- * @since_tizen 7.0
- * @remarks This function would be useful when @a webrtc is operated as an answerer and a remote peer is using another implementation, not this API.\n
- * #WEBRTC_ERROR_INVALID_STATE will no longer occur. (Since 9.0)
- * @param[in] webrtc WebRTC handle
- * @param[in] source_id The media source id
- * @param[in] codec The transceiver codec
- * @param[in] pt The RTP payload type
- * @return @c 0 on success,
- * otherwise a negative error value
- * @retval #WEBRTC_ERROR_NONE Successful
- * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
- * @pre Add media source to @a webrtc to get @a source_id by calling webrtc_add_media_source().
- * @see webrtc_media_source_get_payload_type()
- * @see webrtc_media_source_get_transceiver_codec()
- * @see webrtc_media_source_set_transceiver_codec()
- * @see webrtc_media_source_foreach_supported_transceiver_codec()
- */
-int webrtc_media_source_set_payload_type(webrtc_h webrtc, unsigned int source_id, webrtc_transceiver_codec_e codec, unsigned int pt);
-
-/**
- * @internal
- * @brief Gets the RTP payload type of the media source.
- * @since_tizen 7.0
- * @param[in] webrtc WebRTC handle
- * @param[in] source_id The media source id
- * @param[in] codec The transceiver codec
- * @param[out] pt The RTP payload type
- * @return @c 0 on success,
- * otherwise a negative error value
- * @retval #WEBRTC_ERROR_NONE Successful
- * @retval #WEBRTC_ERROR_INVALID_PARAMETER Invalid parameter
- * @pre Add media source to @a webrtc to get @a source_id by calling webrtc_add_media_source().
- * @see webrtc_media_source_set_payload_type()
- * @see webrtc_media_source_get_transceiver_codec()
- * @see webrtc_media_source_set_transceiver_codec()
- * @see webrtc_media_source_foreach_supported_transceiver_codec()
- */
-int webrtc_media_source_get_payload_type(webrtc_h webrtc, unsigned int source_id, webrtc_transceiver_codec_e codec, unsigned int *pt);
-
/**
* @internal
* @brief Adds the transceiver encoding option to the media source.
void _source_slot_destroy_cb(gpointer data);
int _add_media_source(webrtc_s *webrtc, int type, unsigned int *source_id);
int _add_media_source_internal(webrtc_s *webrtc, int type, unsigned int *source_id);
+int _set_manual_payload_type(webrtc_s *webrtc, unsigned int source_id, webrtc_transceiver_codec_e codec, unsigned int pt);
+int _get_manual_payload_type(webrtc_s *webrtc, unsigned int source_id, webrtc_transceiver_codec_e codec, unsigned int *pt);
int _start_media_source(webrtc_s *webrtc, unsigned int source_id);
int _remove_media_source(webrtc_s *webrtc, unsigned int source_id);
bool _check_if_codec_is_set_to_null_sources(webrtc_s *webrtc);
Name: capi-media-webrtc
Summary: A WebRTC library in Tizen Native API
-Version: 1.1.23
+Version: 1.1.24
Release: 0
Group: Multimedia/API
License: Apache-2.0
return _get_transceiver_codec(webrtc, source_id, media_type, codec);
}
+int webrtc_media_source_set_payload_type(webrtc_h webrtc, unsigned int source_id, webrtc_transceiver_codec_e codec, unsigned int pt)
+{
+ webrtc_s *_webrtc = (webrtc_s *)webrtc;
+ g_autoptr(GMutexLocker) locker = NULL;
+
+ RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+ RET_VAL_IF(source_id == 0, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is 0");
+
+ locker = g_mutex_locker_new(&_webrtc->mutex);
+
+ return _set_manual_payload_type(_webrtc, source_id, codec, pt);
+}
+
+int webrtc_media_source_get_payload_type(webrtc_h webrtc, unsigned int source_id, webrtc_transceiver_codec_e codec, unsigned int *pt)
+{
+ webrtc_s *_webrtc = (webrtc_s *)webrtc;
+ g_autoptr(GMutexLocker) locker = NULL;
+
+ RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+ RET_VAL_IF(source_id == 0, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is 0");
+ RET_VAL_IF(pt == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "pt is NULL");
+
+ locker = g_mutex_locker_new(&_webrtc->mutex);
+
+ return _get_manual_payload_type(_webrtc, source_id, codec, pt);
+}
+
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 _add_media_source_internal(webrtc, type, source_id);
}
-static bool __match_codec_with_media_type(webrtc_transceiver_codec_e codec, const char *media_type)
-{
- RET_VAL_IF(media_type == NULL, false, "media_type is NULL");
-
- switch (codec) {
- case WEBRTC_TRANSCEIVER_CODEC_PCMU:
- if (g_strrstr(media_type, "mulaw") || g_strrstr(media_type, "MULAW"))
- return true;
- break;
- case WEBRTC_TRANSCEIVER_CODEC_PCMA:
- if (g_strrstr(media_type, "alaw") || g_strrstr(media_type, "ALAW"))
- return true;
- break;
- case WEBRTC_TRANSCEIVER_CODEC_OPUS:
- if (g_strrstr(media_type, "opus") || g_strrstr(media_type, "OPUS"))
- return true;
- break;
- case WEBRTC_TRANSCEIVER_CODEC_VP8:
- if (g_strrstr(media_type, "vp8") || g_strrstr(media_type, "VP8"))
- return true;
- break;
- case WEBRTC_TRANSCEIVER_CODEC_VP9:
- if (g_strrstr(media_type, "vp9") || g_strrstr(media_type, "VP9"))
- return true;
- break;
- case WEBRTC_TRANSCEIVER_CODEC_H264:
- if (g_strrstr(media_type, "h264") || g_strrstr(media_type, "H264"))
- return true;
- break;
- default:
- LOG_ERROR("not supported codec[0x%x]", codec);
- }
-
- return false;
-}
-
-int webrtc_media_source_set_payload_type(webrtc_h webrtc, unsigned int source_id, webrtc_transceiver_codec_e codec, unsigned int pt)
-{
- webrtc_s *_webrtc = (webrtc_s *)webrtc;
- webrtc_gst_slot_s *source;
- g_autoptr(GMutexLocker) locker = NULL;
- const char *_media_type;
- int _pt;
- int av_idx = GET_AV_IDX(codec & CODEC_TYPE_AUDIO);
- int mc_idx = -1;
- int i;
-
- RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
- RET_VAL_IF(source_id == 0, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is 0");
-
- locker = g_mutex_locker_new(&_webrtc->mutex);
-
- RET_VAL_IF((source = _get_slot_by_id(_webrtc->gst.source_slots, source_id)) == NULL,
- WEBRTC_ERROR_INVALID_PARAMETER, "could not find source");
-
- if (source->av[av_idx].num_of_codecs > 0) {
- for (i = 0; i < source->av[av_idx].num_of_codecs; i++) {
- const char *codec_str = source->av[av_idx].multiple_codecs[i].codec;
- _media_type = (codec & CODEC_TYPE_AUDIO) ?
- _get_audio_media_type(codec_str) : _get_video_media_type(codec_str);
- if (__match_codec_with_media_type(codec, _media_type)) {
- mc_idx = i;
- break;
- }
- }
- RET_VAL_IF(mc_idx == -1, WEBRTC_ERROR_INVALID_PARAMETER, "matched codec is not found");
-
- } else {
- _media_type = (codec & CODEC_TYPE_AUDIO) ?
- _get_audio_media_type(source->av[av_idx].codec) : _get_video_media_type(source->av[av_idx].codec);
- RET_VAL_IF(_media_type == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "_media_type is NULL");
- RET_VAL_IF(!__match_codec_with_media_type(codec, _media_type), WEBRTC_ERROR_INVALID_PARAMETER, "codec does not match with media_type");
- }
-
- _pt = _get_fixed_payload_type(_media_type);
- if (_pt != -1) {
- LOG_ERROR("this media_type[%s] only allows fixed payload type[%d]", _media_type, _pt);
- return WEBRTC_ERROR_INVALID_PARAMETER;
- }
-
- if (mc_idx != -1 && pt == source->av[av_idx].multiple_codecs[mc_idx].pt) {
- if (!source->av[av_idx].multiple_codecs[mc_idx].pt_set_by_api)
- _return_payload_type(_webrtc, source->av[av_idx].multiple_codecs[mc_idx].pt);
- source->av[av_idx].multiple_codecs[mc_idx].pt_set_by_api = true;
- LOG_INFO("already payload type[%u] was set", source->av[av_idx].multiple_codecs[mc_idx].pt);
- return WEBRTC_ERROR_NONE;
-
- } else if (pt == source->av[av_idx].pt) {
- if (!source->av[av_idx].pt_set_by_api)
- _return_payload_type(_webrtc, source->av[av_idx].pt);
- source->av[av_idx].pt_set_by_api = true;
- LOG_INFO("already payload type[%u] was set", source->av[av_idx].pt);
- return WEBRTC_ERROR_NONE;
-
- } else if (pt < 96 || pt > 127) {
- LOG_ERROR("invalid value[%u] for dynamic payload type (96 ~ 127)", pt);
- return WEBRTC_ERROR_INVALID_PARAMETER;
- }
-
- /* release the previous pt */
- if (mc_idx != -1) {
- if (!source->av[av_idx].multiple_codecs[mc_idx].pt_set_by_api &&
- source->av[av_idx].multiple_codecs[mc_idx].pt > 0)
- _return_payload_type(_webrtc, source->av[av_idx].multiple_codecs[mc_idx].pt);
-
- source->av[av_idx].multiple_codecs[mc_idx].pt = pt;
- source->av[av_idx].multiple_codecs[mc_idx].pt_set_by_api = true;
- } else {
- if (!source->av[av_idx].pt_set_by_api && source->av[av_idx].pt > 0)
- _return_payload_type(_webrtc, source->av[av_idx].pt);
-
- source->av[av_idx].pt = pt;
- source->av[av_idx].pt_set_by_api = true;
- }
-
- LOG_INFO("webrtc[%p] source_id[%u] codec[0x%x] payload type[%u]", _webrtc, source_id, codec, pt);
-
- return _update_pt_if_media_packet_source(webrtc, source);
-}
-
-int webrtc_media_source_get_payload_type(webrtc_h webrtc, unsigned int source_id, webrtc_transceiver_codec_e codec, unsigned int *pt)
-{
- webrtc_s *_webrtc = (webrtc_s *)webrtc;
- webrtc_gst_slot_s *source;
- g_autoptr(GMutexLocker) locker = NULL;
- const char *_media_type;
- int av_idx = GET_AV_IDX(codec & CODEC_TYPE_AUDIO);
- int i;
-
- RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
- RET_VAL_IF(source_id == 0, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is 0");
- RET_VAL_IF(pt == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "pt is NULL");
-
- locker = g_mutex_locker_new(&_webrtc->mutex);
-
- RET_VAL_IF((source = _get_slot_by_id(_webrtc->gst.source_slots, source_id)) == NULL,
- WEBRTC_ERROR_INVALID_PARAMETER, "could not find source");
-
- if (source->av[av_idx].num_of_codecs > 0) {
- for (i = 0; i < source->av[av_idx].num_of_codecs; i++) {
- const char *codec_str = source->av[av_idx].multiple_codecs[i].codec;
- _media_type = (codec & CODEC_TYPE_AUDIO) ?
- _get_audio_media_type(codec_str) : _get_video_media_type(codec_str);
- if (__match_codec_with_media_type(codec, _media_type)) {
- *pt = source->av[av_idx].multiple_codecs[i].pt;
-
- LOG_INFO("webrtc[%p] source_id[%u] codec[0x%x] payload type[%u]", _webrtc, source_id, codec, *pt);
- return WEBRTC_ERROR_NONE;
- }
- }
- LOG_ERROR("could not find matched codec[0x%x]", codec);
- return WEBRTC_ERROR_INVALID_PARAMETER;
- }
-
- _media_type = (codec & CODEC_TYPE_AUDIO) ?
- _get_audio_media_type(source->av[av_idx].codec) : _get_video_media_type(source->av[av_idx].codec);
- RET_VAL_IF(!__match_codec_with_media_type(codec, _media_type), WEBRTC_ERROR_INVALID_PARAMETER, "codec does not match with media_type");
-
- *pt = source->av[GET_AV_IDX(codec & CODEC_TYPE_AUDIO)].pt;
-
- LOG_INFO("webrtc[%p] source_id[%u] codec[0x%x] payload type[%u]", _webrtc, source_id, codec, *pt);
-
- return WEBRTC_ERROR_NONE;
-}
-
int webrtc_media_source_add_transceiver_encoding(webrtc_h webrtc, unsigned int source_id, webrtc_media_type_e media_type, const char *rid, int target_bitrate, int width, int height, int *ssrc)
{
g_autoptr(GMutexLocker) locker = NULL;
return ret;
}
+static bool __match_codec_with_media_type(webrtc_transceiver_codec_e codec, const char *media_type)
+{
+ RET_VAL_IF(media_type == NULL, false, "media_type is NULL");
+
+ switch (codec) {
+ case WEBRTC_TRANSCEIVER_CODEC_PCMU:
+ if (g_strrstr(media_type, "mulaw") || g_strrstr(media_type, "MULAW"))
+ return true;
+ break;
+ case WEBRTC_TRANSCEIVER_CODEC_PCMA:
+ if (g_strrstr(media_type, "alaw") || g_strrstr(media_type, "ALAW"))
+ return true;
+ break;
+ case WEBRTC_TRANSCEIVER_CODEC_OPUS:
+ if (g_strrstr(media_type, "opus") || g_strrstr(media_type, "OPUS"))
+ return true;
+ break;
+ case WEBRTC_TRANSCEIVER_CODEC_VP8:
+ if (g_strrstr(media_type, "vp8") || g_strrstr(media_type, "VP8"))
+ return true;
+ break;
+ case WEBRTC_TRANSCEIVER_CODEC_VP9:
+ if (g_strrstr(media_type, "vp9") || g_strrstr(media_type, "VP9"))
+ return true;
+ break;
+ case WEBRTC_TRANSCEIVER_CODEC_H264:
+ if (g_strrstr(media_type, "h264") || g_strrstr(media_type, "H264"))
+ return true;
+ break;
+ default:
+ LOG_ERROR("not supported codec[0x%x]", codec);
+ }
+
+ return false;
+}
+
+int _set_manual_payload_type(webrtc_s *webrtc, unsigned int source_id, webrtc_transceiver_codec_e codec, unsigned int pt)
+{
+ webrtc_gst_slot_s *source;
+ g_autoptr(GMutexLocker) locker = NULL;
+ const char *_media_type;
+ int _pt;
+ int av_idx = GET_AV_IDX(codec & CODEC_TYPE_AUDIO);
+ int mc_idx = -1;
+ int i;
+
+ ASSERT(webrtc);
+ ASSERT(source_id != 0);
+
+ RET_VAL_IF((source = _get_slot_by_id(webrtc->gst.source_slots, source_id)) == NULL,
+ WEBRTC_ERROR_INVALID_PARAMETER, "could not find source");
+
+ if (source->av[av_idx].num_of_codecs > 0) {
+ for (i = 0; i < source->av[av_idx].num_of_codecs; i++) {
+ const char *codec_str = source->av[av_idx].multiple_codecs[i].codec;
+ _media_type = (codec & CODEC_TYPE_AUDIO) ?
+ _get_audio_media_type(codec_str) : _get_video_media_type(codec_str);
+ if (__match_codec_with_media_type(codec, _media_type)) {
+ mc_idx = i;
+ break;
+ }
+ }
+ RET_VAL_IF(mc_idx == -1, WEBRTC_ERROR_INVALID_PARAMETER, "matched codec is not found");
+
+ } else {
+ _media_type = (codec & CODEC_TYPE_AUDIO) ?
+ _get_audio_media_type(source->av[av_idx].codec) : _get_video_media_type(source->av[av_idx].codec);
+ RET_VAL_IF(_media_type == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "_media_type is NULL");
+ RET_VAL_IF(!__match_codec_with_media_type(codec, _media_type), WEBRTC_ERROR_INVALID_PARAMETER, "codec does not match with media_type");
+ }
+
+ _pt = _get_fixed_payload_type(_media_type);
+ if (_pt != -1) {
+ LOG_ERROR("this media_type[%s] only allows fixed payload type[%d]", _media_type, _pt);
+ return WEBRTC_ERROR_INVALID_PARAMETER;
+ }
+
+ if (mc_idx != -1 && pt == source->av[av_idx].multiple_codecs[mc_idx].pt) {
+ if (!source->av[av_idx].multiple_codecs[mc_idx].pt_set_by_api)
+ _return_payload_type(webrtc, source->av[av_idx].multiple_codecs[mc_idx].pt);
+ source->av[av_idx].multiple_codecs[mc_idx].pt_set_by_api = true;
+ LOG_INFO("already payload type[%u] was set", source->av[av_idx].multiple_codecs[mc_idx].pt);
+ return WEBRTC_ERROR_NONE;
+
+ } else if (pt == source->av[av_idx].pt) {
+ if (!source->av[av_idx].pt_set_by_api)
+ _return_payload_type(webrtc, source->av[av_idx].pt);
+ source->av[av_idx].pt_set_by_api = true;
+ LOG_INFO("already payload type[%u] was set", source->av[av_idx].pt);
+ return WEBRTC_ERROR_NONE;
+
+ } else if (pt < 96 || pt > 127) {
+ LOG_ERROR("invalid value[%u] for dynamic payload type (96 ~ 127)", pt);
+ return WEBRTC_ERROR_INVALID_PARAMETER;
+ }
+
+ /* release the previous pt */
+ if (mc_idx != -1) {
+ if (!source->av[av_idx].multiple_codecs[mc_idx].pt_set_by_api &&
+ source->av[av_idx].multiple_codecs[mc_idx].pt > 0)
+ _return_payload_type(webrtc, source->av[av_idx].multiple_codecs[mc_idx].pt);
+
+ source->av[av_idx].multiple_codecs[mc_idx].pt = pt;
+ source->av[av_idx].multiple_codecs[mc_idx].pt_set_by_api = true;
+ } else {
+ if (!source->av[av_idx].pt_set_by_api && source->av[av_idx].pt > 0)
+ _return_payload_type(webrtc, source->av[av_idx].pt);
+
+ source->av[av_idx].pt = pt;
+ source->av[av_idx].pt_set_by_api = true;
+ }
+
+ LOG_INFO("webrtc[%p] source_id[%u] codec[0x%x] payload type[%u]", webrtc, source_id, codec, pt);
+
+ return _update_pt_if_media_packet_source(webrtc, source);
+}
+
+int _get_manual_payload_type(webrtc_s *webrtc, unsigned int source_id, webrtc_transceiver_codec_e codec, unsigned int *pt)
+{
+ webrtc_gst_slot_s *source;
+ g_autoptr(GMutexLocker) locker = NULL;
+ const char *_media_type;
+ int av_idx = GET_AV_IDX(codec & CODEC_TYPE_AUDIO);
+ int i;
+
+ ASSERT(webrtc);
+ ASSERT(source_id != 0);
+ ASSERT(pt);
+
+ RET_VAL_IF((source = _get_slot_by_id(webrtc->gst.source_slots, source_id)) == NULL,
+ WEBRTC_ERROR_INVALID_PARAMETER, "could not find source");
+
+ if (source->av[av_idx].num_of_codecs > 0) {
+ for (i = 0; i < source->av[av_idx].num_of_codecs; i++) {
+ const char *codec_str = source->av[av_idx].multiple_codecs[i].codec;
+ _media_type = (codec & CODEC_TYPE_AUDIO) ?
+ _get_audio_media_type(codec_str) : _get_video_media_type(codec_str);
+ if (__match_codec_with_media_type(codec, _media_type)) {
+ *pt = source->av[av_idx].multiple_codecs[i].pt;
+
+ LOG_INFO("webrtc[%p] source_id[%u] codec[0x%x] payload type[%u]", webrtc, source_id, codec, *pt);
+ return WEBRTC_ERROR_NONE;
+ }
+ }
+ LOG_ERROR("could not find matched codec[0x%x]", codec);
+ return WEBRTC_ERROR_INVALID_PARAMETER;
+ }
+
+ _media_type = (codec & CODEC_TYPE_AUDIO) ?
+ _get_audio_media_type(source->av[av_idx].codec) : _get_video_media_type(source->av[av_idx].codec);
+ RET_VAL_IF(!__match_codec_with_media_type(codec, _media_type), WEBRTC_ERROR_INVALID_PARAMETER, "codec does not match with media_type");
+
+ *pt = source->av[GET_AV_IDX(codec & CODEC_TYPE_AUDIO)].pt;
+
+ LOG_INFO("webrtc[%p] source_id[%u] codec[0x%x] payload type[%u]", webrtc, source_id, codec, *pt);
+
+ 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;
g_print("fc. Get supported transceiver codecs\t");
g_print("tc. Set transceiver codec\t");
g_print("gc. Get transceiver codec\n");
- g_print("spt. *Set payload type\t");
- g_print("gpt. *Get payload type\n");
+ g_print("spt. Set payload type\t");
+ g_print("gpt. Get payload type\n");
g_print("nsm. *Set media type to null source\t");
g_print("pa. Set media path to file source\n");
g_print("sfl. Set file source looping\t");
projects / platform / core / api / webrtc.git / commitdiff