Add --gst-pipeline option for custom nput pipelines

4 files changed, 76 insertions, 26 deletions

diff --git a/README.md b/README.md
index 0449898..d103b41 100644
--- a/README.md
+++ b/README.md
@@ -200,6 +200,15 @@ libVLC.
 If the webstream bitrate is slightly wrong (bad clock at the source), you can
 enable drift compensation with `-D`.
 
+## Scenario *Custom GStreamer pipeline*
+
+The `--gst-pipeline` option lets you run custom pipelines, using the same
+syntax as `gst-launch`, which can be necessary for sources that you cannot specify through a URI through the `-G` option.
+For example, you may use udpsrc to receive an RTP stream:
+
+    odr-audioenc --gst-pipeline 'udpsrc port=5004 caps=application/x-rtp,media=(string)audio,payload=(int)10,clock-rate=44100 ! rtpL16depay ! audioconvert ! audioresample' \
+    -e $DST -l -b $BITRATE
+
 ## Scenario *JACK input*
 JACK input: Instead of `-i (file input)` or `-d (ALSA input)`, use `-j *name*`, where *name* specifies the JACK
 name for the encoder:
diff --git a/src/GSTInput.cpp b/src/GSTInput.cpp
index 5c63147..6c437d7 100644
--- a/src/GSTInput.cpp
+++ b/src/GSTInput.cpp
@@ -41,10 +41,12 @@ GSTData::GSTData(SampleQueue<uint8_t>& samplequeue) :
 { }
 
 GSTInput::GSTInput(const std::string& uri,
+        const std::string& pipeline,
         int rate,
         unsigned channels,
         SampleQueue<uint8_t>& queue) :
     m_uri(uri),
+    m_pipeline(pipeline),
     m_channels(channels),
     m_rate(rate),
     m_gst_data(queue)
@@ -110,25 +112,34 @@ void GSTInput::prepare()
 {
     gst_init(nullptr, nullptr);
 
-    m_gst_data.uridecodebin = gst_element_factory_make("uridecodebin", "uridecodebin");
-    assert(m_gst_data.uridecodebin != nullptr);
-    g_object_set(m_gst_data.uridecodebin, "uri", m_uri.c_str(), nullptr);
-    g_signal_connect(m_gst_data.uridecodebin, "pad-added", G_CALLBACK(cb_newpad), &m_gst_data);
+    if (not m_uri.empty()) {
+        m_gst_data.uridecodebin = gst_element_factory_make("uridecodebin", "uridecodebin");
+        assert(m_gst_data.uridecodebin != nullptr);
+        g_object_set(m_gst_data.uridecodebin, "uri", m_uri.c_str(), nullptr);
+        g_signal_connect(m_gst_data.uridecodebin, "pad-added", G_CALLBACK(cb_newpad), &m_gst_data);
 
-    m_gst_data.audio_convert = gst_element_factory_make("audioconvert", "audio_convert");
-    assert(m_gst_data.audio_convert != nullptr);
+        m_gst_data.audio_convert = gst_element_factory_make("audioconvert", "audio_convert");
+        assert(m_gst_data.audio_convert != nullptr);
 
-    m_gst_data.audio_resample = gst_element_factory_make("audioresample", "audio_resample");
-    assert(m_gst_data.audio_resample != nullptr);
-    g_object_set(m_gst_data.audio_resample,
+        m_gst_data.audio_resample = gst_element_factory_make("audioresample", "audio_resample");
+        assert(m_gst_data.audio_resample != nullptr);
+        g_object_set(m_gst_data.audio_resample,
 #if (GST_VERSION_MAJOR == 1 && GST_VERSION_MINOR >= 10) || GST_VERSION_MAJOR > 1
-            "sinc-filter-mode", GST_AUDIO_RESAMPLER_FILTER_MODE_FULL,
+                "sinc-filter-mode", GST_AUDIO_RESAMPLER_FILTER_MODE_FULL,
 #else
 #warning "GStreamer version is too old to set GST_AUDIO_RESAMPLER_FILTER_MODE_FULL" GST_VERSION_MAJOR
 #endif
-            "quality", 6, // between 0 and 10, 10 being best
-            /* default audio-resampler-method: GST_AUDIO_RESAMPLER_METHOD_KAISER */
-            NULL);
+                "quality", 6, // between 0 and 10, 10 being best
+                /* default audio-resampler-method: GST_AUDIO_RESAMPLER_METHOD_KAISER */
+                NULL);
+
+    }
+    else if (not m_pipeline.empty()) {
+        m_gst_data.custom_bin = gst_parse_bin_from_description(m_pipeline.c_str(), true, nullptr);
+        if (m_gst_data.custom_bin == nullptr) {
+            throw runtime_error("Could not instantiate pipeline");
+        }
+    }
 
     m_gst_data.caps_filter = gst_element_factory_make("capsfilter", "caps_filter");
     assert(m_gst_data.caps_filter != nullptr);
@@ -149,19 +160,34 @@ void GSTInput::prepare()
     g_signal_connect(m_gst_data.app_sink, "new-sample", G_CALLBACK(new_sample), &m_gst_data);
     gst_caps_unref(audio_caps);
 
-    gst_bin_add_many(GST_BIN(m_gst_data.pipeline),
-            m_gst_data.uridecodebin,
-            m_gst_data.audio_convert,
-            m_gst_data.audio_resample,
-            m_gst_data.caps_filter,
-            m_gst_data.app_sink, NULL);
-
-    if (gst_element_link_many(
+    if (not m_uri.empty()) {
+        gst_bin_add_many(GST_BIN(m_gst_data.pipeline),
+                m_gst_data.uridecodebin,
                 m_gst_data.audio_convert,
                 m_gst_data.audio_resample,
                 m_gst_data.caps_filter,
-                m_gst_data.app_sink, NULL) != true) {
-        throw runtime_error("Could not link GST elements");
+                m_gst_data.app_sink, NULL);
+
+        if (gst_element_link_many(
+                    m_gst_data.audio_convert,
+                    m_gst_data.audio_resample,
+                    m_gst_data.caps_filter,
+                    m_gst_data.app_sink, NULL) != true) {
+            throw runtime_error("Could not link GST elements");
+        }
+    }
+    else if (not m_pipeline.empty()) {
+        gst_bin_add_many(GST_BIN(m_gst_data.pipeline),
+                m_gst_data.custom_bin,
+                m_gst_data.caps_filter,
+                m_gst_data.app_sink, NULL);
+
+        if (gst_element_link_many(
+                    m_gst_data.custom_bin,
+                    m_gst_data.caps_filter,
+                    m_gst_data.app_sink, NULL) != true) {
+            throw runtime_error("Could not link GST elements");
+        }
     }
 
     m_gst_data.bus = gst_element_get_bus(m_gst_data.pipeline);
diff --git a/src/GSTInput.h b/src/GSTInput.h
index 4bfae34..3c2971d 100644
--- a/src/GSTInput.h
+++ b/src/GSTInput.h
@@ -43,6 +43,7 @@
 struct GSTData {
     GSTData(SampleQueue<uint8_t>& samplequeue);
 
+    // When using URL and uridecodebin
     GstElement *pipeline = nullptr;
     GstElement *uridecodebin = nullptr;
     GstElement *audio_convert = nullptr;
@@ -50,6 +51,9 @@ struct GSTData {
     GstElement *caps_filter = nullptr;
     GstElement *app_sink = nullptr;
 
+    // When using pipeline
+    GstElement *custom_bin = nullptr;
+
     GstBus *bus = nullptr;
 
     SampleQueue<uint8_t>& samplequeue;
@@ -59,6 +63,7 @@ class GSTInput : public InputInterface
 {
     public:
         GSTInput(const std::string& uri,
+                 const std::string& pipeline,
                  int rate,
                  unsigned channels,
                  SampleQueue<uint8_t>& queue);
@@ -78,6 +83,7 @@ class GSTInput : public InputInterface
         virtual bool fault_detected(void) const override { return m_fault; };
     private:
         std::string m_uri;
+        std::string m_pipeline;
         unsigned m_channels;
         int m_rate;
 
diff --git a/src/odr-audioenc.cpp b/src/odr-audioenc.cpp
index e3708c1..21abb91 100644
--- a/src/odr-audioenc.cpp
+++ b/src/odr-audioenc.cpp
@@ -162,6 +162,9 @@ static void usage(const char* name)
     "   For the GStreamer input:\n"
 #if HAVE_GST
     "     -G, --gst-uri=uri                    Enable GStreamer input and use the URI given as source\n"
+    "         --gst-pipeline=pipeline          Specify a GStreamer pipeline that receives your source.\n"
+    "                                          The last pipeline element is connected to a caps filter that specifies\n"
+    "                                          the audio format and sample rate.\n"
 #else
     "     The GStreamer input was disabled at compile-time\n"
 #endif
@@ -427,6 +430,7 @@ public:
 
     // For the GST input
     string gst_uri;
+    string gst_pipeline;
 
     string jack_name;
 
@@ -517,6 +521,7 @@ int AudioEnc::run()
 #endif
 #if HAVE_GST
     if (not gst_uri.empty()) num_inputs++;
+    if (not gst_pipeline.empty()) num_inputs++;
 #endif
 
     if (num_inputs == 0) {
@@ -987,7 +992,7 @@ int AudioEnc::run()
             }
 #endif
 #if HAVE_GST
-            else if (not gst_uri.empty()) {
+            else if ((not gst_uri.empty()) or (not gst_pipeline.empty())) {
                 GSTInput *gst_input = (GSTInput*)(input.get());
                 text = gst_input->get_icy_text();
             }
@@ -1342,8 +1347,8 @@ shared_ptr<InputInterface> AudioEnc::initialise_input()
     }
 #endif
 #if HAVE_GST
-    else if (not gst_uri.empty()) {
-        input = make_shared<GSTInput>(gst_uri, sample_rate, channels, queue);
+    else if ((not gst_uri.empty()) or (not gst_pipeline.empty())) {
+        input = make_shared<GSTInput>(gst_uri, gst_pipeline, sample_rate, channels, queue);
     }
 #endif
 #if HAVE_ALSA
@@ -1381,6 +1386,7 @@ int main(int argc, char *argv[])
         {"decode",                 required_argument,  0,  6 },
         {"format",                 required_argument,  0, 'f'},
         {"gst-uri",                required_argument,  0, 'G'},
+        {"gst-pipeline",           required_argument,  0, 11 },
         {"identifier",             required_argument,  0,  7 },
         {"input",                  required_argument,  0, 'i'},
         {"jack",                   required_argument,  0, 'j'},
@@ -1543,6 +1549,9 @@ int main(int argc, char *argv[])
         case 'G':
             audio_enc.gst_uri = optarg;
             break;
+        case 11: // --gst-pipeline
+            audio_enc.gst_pipeline = optarg;
+            break;
 #endif
         case 'i':
             audio_enc.infile = optarg;