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Added Gst-Python-sample applications
Signed-off-by: s.sushma <s.sushma@globaledgesoft.com>
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‎gst-python-samples/Gstreamer-Applications/gst_audio/README.md‎

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# PulseAudio Record and Playback
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This application demonstrates the audio record and playback with pulse audio.
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## Audio Record
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Audio_record application records audio from a pulse audio source. Two formats are supported: WAV and MP3.
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**Note:**
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- On board, DIP_SW_0 SW1 must be switched to ON.
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- If there is Alsa service enabled, please remove Alsa service to make PulseAudio source work
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### Start Recording:
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Steps for running the audio recording python application in wav and mp3 format.
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- Go to the downloaded directory on RB5 /gst-python-samples/Gstreamer-Applications/gst_audio
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```bash
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$ cd sample-apps-for-Qualcomm-Robotics-RB5-platform/gst-python-samples/Gstreamer-Applications/gst_audio
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```
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- Recording audio in wav and mp3 format. Provide the filename along with the format as an argument for the command.
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```bash
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$ python3 audio_record.py file.wav
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$ python3 audio_record.py file.mp3
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```
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### Stop Automatically:
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+ The recording stops automatically in 10 seconds.
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## Audio Playback
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This application plays audio from a file to pulseaudio sink. Three formats are supported: WAV, AAC and MP3.
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**Note:** Speaker must be connected to the board
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### Start Playback:
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Steps for running the audio playback python application in wav, mp3 and aac format
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- Go to the downloaded directory on RB5
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```bash
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$ cd sample-apps-for-Qualcomm-Robotics-RB5-platform/gst-python-samples/Gstreamer-Applications/gst_audio
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```
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- Download the audio files of the wav, mp3 and aac formats in the same folder.
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- Audio playback supports three formats wav, mp3 and aac.
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Provide the filename along with the format as an argument for the command.
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```bash
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$ python3 audio_playback.py file.wav
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$ python3 audio_playback.py file.mp3
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$ python3 audio_playback.py file.aac
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```
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‎gst-python-samples/Gstreamer-Applications/gst_audio/audio_playback.py‎

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import sys
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import gi
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import logging
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import os
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gi.require_version("GLib", "2.0")
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gi.require_version("GObject", "2.0")
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gi.require_version("Gst", "1.0")
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from gi.repository import Gst, GLib
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logging.basicConfig(level=logging.DEBUG,
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format="[%(name)s] [%(levelname)8s] - %(message)s")
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logger = logging.getLogger(__name__)
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if __name__ == "__main__":
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# Initialize GStreamer passing command line argument
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Gst.init(sys.argv)
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loop = GLib.MainLoop()
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if len(sys.argv) < 1:
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print("Missing <output-location> parameter")
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sys.exit(1)
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file_format = os.path.splitext(sys.argv[1])[1]
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print(file_format)
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# Create the empty pipeline
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pipeline = Gst.Pipeline.new("audio playback")
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filesrc = Gst.ElementFactory.make("filesrc", "file_src")
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pulsesink = Gst.ElementFactory.make("pulsesink", "pulseaudio_sink")
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if not pipeline or not filesrc or not pulsesink:
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print("Create element failed.")
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sys.exit(1)
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filesrc.set_property('location', sys.argv[1])
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if file_format == ".wav":
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wavparse = Gst.ElementFactory.make("wavparse", "wav_parse")
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if not wavparse:
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print("Create element failed.\n")
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sys.exit(1)
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pipeline.add(filesrc)
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pipeline.add(wavparse)
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pipeline.add(pulsesink)
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if not filesrc.link(wavparse):
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print("ERROR: Could not link filesrc to wavparse")
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sys.exit(1)
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if not wavparse.link(pulsesink):
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print("ERROR: Could not link decoder to video_convert")
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sys.exit(1)
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elif file_format == ".aac":
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aacparse = Gst.ElementFactory.make("aacparse", "aac_parse")
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avdec_aac = Gst.ElementFactory.make("avdec_aac", "avdec_aac")
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if not aacparse or not avdec_aac:
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print("Create element failed.\n")
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sys.exit(1)
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pipeline.add(filesrc)
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pipeline.add(aacparse)
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pipeline.add(avdec_aac)
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pipeline.add(pulsesink)
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if not filesrc.link(aacparse):
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print("ERROR: Could not link filesrc to aacparse")
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sys.exit(1)
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if not aacparse.link(avdec_aac):
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print("ERROR: Could not link aacparse to avdec_aac")
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sys.exit(1)
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if not avdec_aac.link(pulsesink):
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print("ERROR: Could not link avdec_aac to pulsesink")
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sys.exit(1)
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elif file_format == ".mp3":
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mpegaudioparse = Gst.ElementFactory.make(
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"mpegaudioparse", "mpegaudio_parse")
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avdec_mp3 = Gst.ElementFactory.make("avdec_mp3", "avdec_mp3")
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if not mpegaudioparse or not avdec_mp3:
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print("Create element failed.\n")
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sys.exit(1)
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pipeline.add(filesrc)
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pipeline.add(mpegaudioparse)
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pipeline.add(avdec_mp3)
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pipeline.add(pulsesink)
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if not filesrc.link(mpegaudioparse):
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print("ERROR: Could not link filesrc to mpegaudioparse")
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sys.exit(1)
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if not mpegaudioparse.link(avdec_mp3):
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print("ERROR: Could not link mpegaudioparse to avdec_mp3")
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sys.exit(1)
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if not avdec_mp3.link(pulsesink):
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print("ERROR: Could not link avdec_mp3 to pulsesink")
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sys.exit(1)
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else:
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print("Format not supported\n")
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sys.exit(1)
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pipeline.set_state(Gst.State.PLAYING)
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print("audio playback started")
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bus = pipeline.get_bus()
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bus_id = bus.add_signal_watch()
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msg = bus.timed_pop_filtered(
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Gst.CLOCK_TIME_NONE,
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Gst.MessageType.ERROR | Gst.MessageType.EOS)
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if msg:
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t = msg.type
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if t == Gst.MessageType.ERROR:
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err, dbg = msg.parse_error()
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print("ERROR:", msg.src.get_name(), ":", err.message)
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if dbg:
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print("debugging info:", dbg)
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elif t == Gst.MessageType.EOS:
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print("End-Of-Stream reached stoped")
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else:
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print("ERROR: Unexpected message received.")
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pipeline.set_state(Gst.State.NULL)
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Gst.Object.unref(bus)
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Gst.deinit()

‎gst-python-samples/Gstreamer-Applications/gst_audio/audio_record.py‎

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import sys
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import gi
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import logging
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import os
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gi.require_version("GLib", "2.0")
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gi.require_version("GObject", "2.0")
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gi.require_version("Gst", "1.0")
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from gi.repository import Gst, GLib, GObject
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logging.basicConfig(level=logging.DEBUG,
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format="[%(name)s] [%(levelname)8s] - %(message)s")
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logger = logging.getLogger(__name__)
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if __name__ == "__main__":
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# Initialize GStreamer passing command line argument
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Gst.init(sys.argv)
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loop = GLib.MainLoop()
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if len(sys.argv) < 1:
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print("Missing <output-location> parameter")
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sys.exit(1)
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file_format = os.path.splitext(sys.argv[1])[1]
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# Create the empty pipeline
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pipeline = Gst.Pipeline.new("audio record")
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pulsesrc = Gst.ElementFactory.make("pulsesrc", "pulseaudio_src");
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audioconvert = Gst.ElementFactory.make("audioconvert", "audio_convert");
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capsfilter = Gst.ElementFactory.make("capsfilter", "caps_filter");
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queue = Gst.ElementFactory.make("queue", "queue");
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filesink = Gst.ElementFactory.make("filesink", "file_sink");
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if not pipeline or not pulsesrc or not audioconvert or not capsfilter or not queue or not filesink:
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print("Create element failed")
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sys.exit(1)
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pulsesrc.set_property('num-buffers', 1000)
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pulsesrc.set_property('buffer-time', 30000)
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caps=Gst.Caps.new_empty_simple("audio/x-raw")
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capsfilter.set_property('caps', caps)
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filesink.set_property('location', sys.argv[1])
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if file_format == ".wav":
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wavenc=Gst.ElementFactory.make("wavenc", "wav_enc");
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if not wavenc:
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print("Create element failed.\n");
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sys.exit(1)
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pipeline.add(pulsesrc)
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pipeline.add(capsfilter)
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pipeline.add(audioconvert)
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pipeline.add(wavenc)
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pipeline.add(queue)
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pipeline.add(filesink)
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if not pulsesrc.link(capsfilter):
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print("ERROR: Could not link pulsesrc to capsfilter")
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sys.exit(1)
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if not capsfilter.link(audioconvert):
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print("ERROR: Could not link capsfilter to audioconvert")
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sys.exit(1)
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if not audioconvert.link(wavenc):
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print("ERROR: Could not link audioconvert to wavenc")
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sys.exit(1)
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if not wavenc.link(queue):
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print("ERROR: Could not link wavenc to queue")
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sys.exit(1)
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if not queue.link(filesink):
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print("ERROR: Could not link queue to filesink")
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sys.exit(1)
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elif file_format == ".aac":
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avenc_aac=Gst.ElementFactory.make("avenc_aac", "aac_enc");
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aacparse=Gst.ElementFactory.make("aacparse", "aac_parse");
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aac_caps=Gst.ElementFactory.make("capsfilter", "aac_caps_filter");
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if not avenc_aac or not aacparse or not aac_caps:
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print("Create element failed.\n");
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sys.exit(1)
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caps=Gst.Caps.new_empty_simple('audio/mpeg')
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aac_caps.set_property('caps', caps)
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pipeline.add(pulsesrc)
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pipeline.add(capsfilter)
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pipeline.add(audioconvert)
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pipeline.add(avenc_aac)
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pipeline.add(aacparse)
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pipeline.add(aac_caps)
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pipeline.add(queue)
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pipeline.add(filesink)
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if not pulsesrc.link(capsfilter):
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print("ERROR: Could not link pulsesrc to capsfilter")
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sys.exit(1)
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if not capsfilter.link(audioconvert):
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print("ERROR: Could not link capsfilter to audioconvert")
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sys.exit(1)
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if not audioconvert.link(avenc_aac):
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print("ERROR: Could not link audioconvert to avenc_aac")
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sys.exit(1)
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if not avenc_aac.link(aacparse):
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print("ERROR: Could not link avenc_aac to aacparse")
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sys.exit(1)
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if not aacparse.link(aac_caps):
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print("ERROR: Could not link aacparse to aac_caps")
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sys.exit(1)
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if not aac_caps.link(queue):
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print("ERROR: Could not link aac_caps to queue")
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sys.exit(1)
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if not queue.link(filesink):
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print("ERROR: Could not link queue to filesink")
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sys.exit(1)
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elif file_format == ".mp3":
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lamemp3enc=Gst.ElementFactory.make("lamemp3enc", "mp3_enc");
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if not lamemp3enc:
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print("Create element failed.\n");
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sys.exit(1)
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pipeline.add(pulsesrc)
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pipeline.add(capsfilter)
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pipeline.add(audioconvert)
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pipeline.add(lamemp3enc)
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pipeline.add(queue)
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pipeline.add(filesink)
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if not pulsesrc.link(capsfilter):
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print("ERROR: Could not link pulsesrc to capsfilter")
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sys.exit(1)
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if not capsfilter.link(audioconvert):
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print("ERROR: Could not link capsfilter to audioconvert")
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sys.exit(1)
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if not audioconvert.link(lamemp3enc):
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print("ERROR: Could not link audioconvert to lamemp3enc")
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sys.exit(1)
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if not lamemp3enc.link(queue):
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print("ERROR: Could not link lamemp3enc to queue")
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sys.exit(1)
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if not queue.link(filesink):
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print("ERROR: Could not link queue to filesink")
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sys.exit(1)
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else:
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print("Format %s not supported\n", format);
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sys.exit(1)
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pipeline.set_state(Gst.State.PLAYING)
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print("Audio record started");
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bus = pipeline.get_bus()
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bus_id = bus.add_signal_watch()
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msg = bus.timed_pop_filtered(Gst.CLOCK_TIME_NONE,Gst.MessageType.ERROR | Gst.MessageType.EOS)
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if msg:
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t = msg.type
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if t == Gst.MessageType.ERROR:
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err, dbg = msg.parse_error()
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print("ERROR:", msg.src.get_name(), ":", err.message)
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if dbg:
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print("debugging info:", dbg)
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elif t == Gst.MessageType.EOS:
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print("stopped recording")
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else:
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print("ERROR: Unexpected message received.")
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pipeline.set_state(Gst.State.NULL)
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Gst.Object.unref(bus)
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Gst.deinit()

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