From: Sangchul Lee <sc11.lee@samsung.com>
Date: Mon, 4 Jan 2021 10:31:35 +0000 (+0900)
Subject: webrtc_source: Use gst_audio_info_* functions of gst-plugins-base
X-Git-Tag: submit/tizen/20210729.023123~163
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=df3a44b1fb91e8212556c01f868edc0c57ffd77a;p=platform%2Fcore%2Fapi%2Fwebrtc.git

webrtc_source: Use gst_audio_info_* functions of gst-plugins-base

gst_audio_info_set_format() and gst_audio_info_to_caps() are
used to make caps for raw media format.

[Version] 0.1.78
[Issue Type] Improvement

Change-Id: I1dd2aa407a2d5f6627c1856d83a992154a746954
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/CMakeLists.txt b/CMakeLists.txt
index e6a5639b..70fe5504 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -11,7 +11,8 @@ SET(LIB_INSTALL_DIR "${CMAKE_INSTALL_PREFIX}/lib")
 SET(INC_DIR include)
 INCLUDE_DIRECTORIES(${INC_DIR})
 
-SET(dependents "dlog glib-2.0 gstreamer-1.0 gstreamer-webrtc-1.0 gstreamer-video-1.0 json-glib-1.0 iniparser mm-common mm-display-interface capi-media-tool libtbm")
+SET(dependents "dlog glib-2.0 gstreamer-1.0 gstreamer-webrtc-1.0 gstreamer-video-1.0 gstreamer-audio-1.0 \
+                json-glib-1.0 iniparser mm-common mm-display-interface capi-media-tool libtbm")
 SET(pc_dependents "capi-base-common" )
 
 INCLUDE(FindPkgConfig)
diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index f02d8718..1bdc65d7 100644
--- a/packaging/capi-media-webrtc.spec
+++ b/packaging/capi-media-webrtc.spec
@@ -1,6 +1,6 @@
 Name:       capi-media-webrtc
 Summary:    A WebRTC library in Tizen Native API
-Version:    0.1.77
+Version:    0.1.78
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
@@ -15,6 +15,7 @@ BuildRequires:  pkgconfig(capi-media-tool)
 BuildRequires:  pkgconfig(gstreamer-1.0)
 BuildRequires:  pkgconfig(gstreamer-webrtc-1.0)
 BuildRequires:  pkgconfig(gstreamer-video-1.0)
+BuildRequires:  pkgconfig(gstreamer-audio-1.0)
 BuildRequires:  pkgconfig(appcore-efl)
 BuildRequires:  pkgconfig(elementary)
 BuildRequires:  pkgconfig(json-glib-1.0)
diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index 51ceeed1..8a3b17ad 100644
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -14,6 +14,7 @@
  * limitations under the License.
  */
 
+#include <gst/audio/audio.h>
 #include "webrtc.h"
 #include "webrtc_private.h"
 #include <tbm_surface_internal.h>
@@ -169,6 +170,24 @@ static const char *__get_format_name(media_format_mimetype_e mime_type)
 	}
 }
 
+static GstAudioFormat __get_gst_audio_format(media_format_mimetype_e mime_type)
+{
+	switch (mime_type) {
+	/* RAW formats */
+	case MEDIA_FORMAT_PCM_S16LE:
+		return GST_AUDIO_FORMAT_S16LE;
+	case MEDIA_FORMAT_PCM_S24LE:
+		return GST_AUDIO_FORMAT_S24_32LE;
+	/* ENCODED formats */
+	case MEDIA_FORMAT_OPUS:
+	case MEDIA_FORMAT_VORBIS:
+		return GST_AUDIO_FORMAT_ENCODED;
+	default:
+		LOG_ERROR("not supported mime_type(0x%x)", mime_type);
+		return GST_AUDIO_FORMAT_UNKNOWN;
+	}
+}
+
 static bool __is_supported_mime_type(media_format_mimetype_e mime_type)
 {
 	switch (mime_type) {
@@ -285,14 +304,17 @@ static GstCaps *__make_raw_caps_from_media_format(webrtc_gst_slot_s *source)
 	if (source->media_types == MEDIA_TYPE_AUDIO) {
 		int channels;
 		int samplerate;
+		GstAudioInfo info;
+		GstAudioFormat format;
 
 		RET_VAL_IF(media_format_get_audio_info(source->media_format, &mime_type, &channels, &samplerate, NULL, NULL) != MEDIA_FORMAT_ERROR_NONE,
 			NULL, "failed to media_format_get_audio_info()");
-		caps = gst_caps_new_simple(MEDIA_TYPE_AUDIO_RAW,
-						"format", G_TYPE_STRING, __get_format_name(mime_type),
-						"channels", G_TYPE_INT, channels,
-						"rate", G_TYPE_INT, samplerate,
-						NULL);
+
+		format = __get_gst_audio_format(mime_type);
+		RET_VAL_IF(format == GST_AUDIO_FORMAT_ENCODED || format == GST_AUDIO_FORMAT_UNKNOWN, NULL, "could not get valid GstAudioFormat for PCM");
+
+		gst_audio_info_set_format(&info, format, samplerate, channels, NULL);
+		caps = gst_audio_info_to_caps(&info);
 
 	} else if (source->media_types == MEDIA_TYPE_VIDEO) {
 		int width;
@@ -856,7 +878,7 @@ static int __build_mediapacketsrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 		return WEBRTC_ERROR_INVALID_OPERATION;
 	}
 
-	g_object_set(G_OBJECT(appsrc), "emit-signals", TRUE, "is-live", TRUE, "format", 3, NULL); /* format 3 for time format of the segment events and seek */
+	g_object_set(G_OBJECT(appsrc), "emit-signals", TRUE, "is-live", TRUE, "format", GST_FORMAT_TIME, NULL);
 
 	_connect_and_append_signal(&source->signals, G_OBJECT(appsrc), "need-data", G_CALLBACK(_appsrc_need_data_cb), source);
 	_connect_and_append_signal(&source->signals, G_OBJECT(appsrc), "enough-data", G_CALLBACK(_appsrc_enough_data_cb), source);