From 779d38fb2e2d6b52ce6ae43a71396a7f935fd2e8 Mon Sep 17 00:00:00 2001
From: Nitin Nakka <nitinn@qti.qualcomm.com>
Date: Sat, 7 Mar 2026 00:39:22 +0530
Subject: [PATCH 1/7] Adding GStreamer test cases for Audio record and playback
 functionality, supports wav and flac codecs, added support in lib gstreamer
 sh function to handle pipelines

Signed-off-by: Nitin Nakka <nitinn@qti.qualcomm.com>
---
 .../Audio_Record_Playback.yaml                |  26 +
 .../Audio/Audio_Record_Playback/README.md     | 553 +++++++++++++++
 .../Audio/Audio_Record_Playback/run.sh        | 669 ++++++++++++++++++
 Runner/utils/lib_gstreamer.sh                 |  90 ++-
 4 files changed, 1318 insertions(+), 20 deletions(-)
 create mode 100644 Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/Audio_Record_Playback.yaml
 create mode 100644 Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/README.md
 create mode 100644 Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh

diff --git a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/Audio_Record_Playback.yaml b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/Audio_Record_Playback.yaml
new file mode 100644
index 000000000..3bdc919ee
--- /dev/null
+++ b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/Audio_Record_Playback.yaml
@@ -0,0 +1,26 @@
+metadata:
+  name: gstreamer-audio-record-playback
+  format: "Lava-Test Test Definition 1.0"
+  description: >
+    Audio record/playback validation using GStreamer (gst-launch-1.0).
+    Tests 8 scenarios: 4 encode (2 audiotestsrc + 2 pulsesrc) + 4 decode (playback).
+    Supports WAV and FLAC formats with both synthetic audio (audiotestsrc) and hardware capture (pulsesrc).
+    Duration controlled via AUDIO_DURATION (default 10s): audiotestsrc uses num-buffers, pulsesrc uses timeout.
+  os:
+    - linux
+  scope:
+    - functional
+ 
+params:
+  AUDIO_TEST_MODE: "all"
+  AUDIO_FORMATS: "wav,flac"
+  AUDIO_DURATION: "10"
+  AUDIO_GST_DEBUG: "2"
+
+run:
+  steps:
+    - REPO_PATH="$PWD"
+    - cd Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/
+    - export AUDIO_TEST_MODE AUDIO_FORMATS AUDIO_DURATION AUDIO_GST_DEBUG
+    - ./run.sh --mode "${AUDIO_TEST_MODE}" --formats "${AUDIO_FORMATS}" --duration "${AUDIO_DURATION}" --gst-debug "${AUDIO_GST_DEBUG}" || true
+    - $REPO_PATH/Runner/utils/send-to-lava.sh Audio_Record_Playback.res || true
diff --git a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/README.md b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/README.md
new file mode 100644
index 000000000..60ce90f61
--- /dev/null
+++ b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/README.md
@@ -0,0 +1,553 @@
+# Audio_Record_Playback (GStreamer) — Runner Test
+
+This directory contains the **Audio_Record_Playback** validation test for Qualcomm Linux Testkit runners.
+
+It validates audio **recording and playback** using **GStreamer (`gst-launch-1.0`)** with:
+- **audiotestsrc** - Synthetic audio generation (no microphone needed)
+- **pulsesrc** - Hardware audio capture (microphone/line-in)
+- **pulsesink** - PulseAudio playback
+- **wavenc** / **flacenc** - Audio encoding
+- **wavparse** / **flacparse + flacdec** - Audio decoding
+
+The script is designed to be **CI/LAVA-friendly**:
+- Writes **PASS/FAIL/SKIP** into `Audio_Record_Playback.res`
+- Always **exits 0** (even on FAIL/SKIP) to avoid terminating LAVA jobs early
+- Logs the **final `gst-launch-1.0` command** to console and to log files
+- Supports both synthetic audio (audiotestsrc) and hardware capture (pulsesrc)
+- Dynamically calculates buffer count based on duration for audiotestsrc
+- Uses timeout mechanism for pulsesrc duration control
+
+---
+
+## Location in repo
+
+Expected path:
+
+```
+Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh
+```
+
+Required shared utils (sourced from `Runner/utils` via `init_env`):
+- `functestlib.sh`
+- `lib_gstreamer.sh` - **Contains reusable audio pipeline builders** (see Library Functions section below)
+
+---
+
+## What this test does
+
+At a high level, the test:
+
+1. Finds and sources `init_env`
+2. Sources:
+   - `$TOOLS/functestlib.sh`
+   - `$TOOLS/lib_gstreamer.sh`
+3. Checks for required GStreamer elements (audiotestsrc, pulsesrc, pulsesink, wavenc, flacenc, wavparse, flacparse, flacdec)
+4. **Recording phase (ENCODE - 4 tests)**:
+   - **audiotestsrc tests**: Generates sine wave at 440Hz, encodes to WAV/FLAC
+   - **pulsesrc tests**: Captures from hardware audio input, encodes to WAV/FLAC
+   - Saves recorded files to `logs/Audio_Record_Playback/recorded/`
+   - Duration control:
+     - audiotestsrc: Buffer count calculated dynamically: `(44100 * duration) / 1024`
+     - pulsesrc: Uses timeout command to stop recording after specified duration
+5. **Playback phase (DECODE - 4 tests)**:
+   - Reads the previously recorded files (both audiotestsrc and pulsesrc recordings)
+   - Decodes and plays back using pulsesink
+6. Collects test results and emits PASS/FAIL/SKIP
+
+---
+
+## Test Cases
+
+By default, the test runs **8 test cases** with 10 second duration:
+
+### ENCODE PHASE (4 tests)
+
+#### Recording Tests - audiotestsrc (Synthetic Audio)
+1. **record_wav** - Record WAV format using audiotestsrc → audioconvert → wavenc
+2. **record_flac** - Record FLAC format using audiotestsrc → audioconvert → flacenc
+
+#### Recording Tests - pulsesrc (Hardware Audio Capture)
+3. **record_pulsesrc_wav** - Record WAV format using pulsesrc → audioconvert → wavenc
+4. **record_pulsesrc_flac** - Record FLAC format using pulsesrc → audioconvert → flacenc
+
+### DECODE PHASE (4 tests)
+
+#### Playback Tests - audiotestsrc recordings
+5. **playback_wav** - Playback WAV file using filesrc → wavparse → audioconvert → pulsesink
+6. **playback_flac** - Playback FLAC file using filesrc → flacparse → flacdec → pulsesink
+
+#### Playback Tests - pulsesrc recordings
+7. **playback_pulsesrc_wav** - Playback WAV file using filesrc → wavparse → audioconvert → pulsesink
+8. **playback_pulsesrc_flac** - Playback FLAC file using filesrc → flacparse → flacdec → pulsesink
+
+**Total: 8 test cases** (4 encode + 4 decode)
+
+---
+
+## PASS / FAIL / SKIP criteria
+
+### PASS
+- **Recording**: Output file is created and has size > 1000 bytes
+- **Playback**: Pipeline completes successfully (exit code 0, 124, or 143)
+- **Overall**: At least one test passes and no tests fail
+
+### FAIL
+- **Recording**: No output file created or file size too small
+- **Playback**: Pipeline fails or GStreamer errors detected
+- **Overall**: One or more tests fail
+
+### SKIP
+- Missing required tools (`gst-launch-1.0`, `gst-inspect-1.0`)
+- Required GStreamer elements not available (audiotestsrc, pulsesrc, pulsesink, wavenc, flacenc)
+- For playback tests: corresponding recorded file not found (record must run first)
+- For pulsesrc tests: pulsesrc plugin not available
+
+**Note:** The test always exits `0` even for FAIL/SKIP. The `.res` file is the source of truth.
+
+---
+
+## Logs and artifacts
+
+By default, logs are written relative to the script working directory:
+
+```
+./Audio_Record_Playback.res
+./logs/Audio_Record_Playback/
+  gst.log                         # GStreamer debug output
+  record_wav.log                  # Individual test logs
+  record_flac.log
+  record_pulsesrc_wav.log
+  record_pulsesrc_flac.log
+  playback_wav.log
+  playback_flac.log
+  playback_pulsesrc_wav.log
+  playback_pulsesrc_flac.log
+  recorded/                       # Recorded audio files
+    record_wav.wav
+    record_flac.flac
+    record_pulsesrc_wav.wav
+    record_pulsesrc_flac.flac
+  dmesg/                          # dmesg scan outputs (if available)
+    dmesg_errors.log
+```
+
+---
+
+## Dependencies
+
+### Required
+- `gst-launch-1.0`
+- `gst-inspect-1.0`
+- `audiotestsrc` GStreamer plugin
+- `audioconvert` GStreamer plugin
+- `pulsesink` GStreamer plugin
+
+### Optional (for hardware capture tests)
+- `pulsesrc` GStreamer plugin
+- Working audio input device (microphone/line-in)
+
+### Encoder Elements
+- `wavenc` - WAV encoder
+- `flacenc` - FLAC encoder
+
+### Decoder Elements
+- `wavparse` - WAV parser
+- `flacparse` - FLAC parser
+- `flacdec` - FLAC decoder
+
+### Audio Backend
+- PulseAudio server running
+
+---
+
+## Usage
+
+Run:
+
+```bash
+./run.sh [options]
+```
+
+Help:
+
+```bash
+./run.sh --help
+```
+
+### Options
+
+- `--mode <all|record|playback>`
+  - Default: `all` (run both record and playback tests)
+  - `record`: Run only recording tests (both audiotestsrc and pulsesrc)
+  - `playback`: Run only playback tests (requires recorded files from previous record run)
+
+- `--formats <wav,flac>`
+  - Comma-separated list of formats to test
+  - Default: `wav,flac` (both formats)
+  - Examples: `wav`, `flac`, `wav,flac`
+
+- `--duration <seconds>`
+  - Duration for recording (in seconds)
+  - Default: `10`
+  - For audiotestsrc: Determines buffer count: `(44100 * duration) / 1024`
+  - For pulsesrc: Uses timeout command to stop recording after this duration
+  - Example: 10 seconds → 430 buffers (audiotestsrc) or 10 second timeout (pulsesrc)
+
+- `--gst-debug <level>`
+  - Sets `GST_DEBUG=<level>` (1-9)
+  - Values:
+    - `1` ERROR
+    - `2` WARNING (default)
+    - `3` FIXME
+    - `4` INFO
+    - `5` DEBUG
+    - `6` LOG
+    - `7` TRACE
+    - `8` MEMDUMP
+    - `9` MEMDUMP
+  - Default: `2`
+
+---
+
+## Examples
+
+### 1) Run all tests (default - 8 tests: 4 encode + 4 decode for WAV and FLAC with 10 second duration)
+
+```bash
+./run.sh
+```
+
+### 2) Run only recording tests (4 encode tests)
+
+```bash
+./run.sh --mode record
+```
+
+### 3) Run only playback tests (4 decode tests - requires recorded files from previous run)
+
+```bash
+./run.sh --mode playback
+```
+
+### 4) Test only WAV format (4 tests: 2 encode + 2 decode)
+
+```bash
+./run.sh --formats wav
+```
+
+### 5) Test only FLAC format (4 tests: 2 encode + 2 decode)
+
+```bash
+./run.sh --formats flac
+```
+
+### 6) Test with longer duration (30 seconds)
+
+```bash
+./run.sh --duration 30
+```
+
+### 7) Quick test - WAV only with 3 second duration (4 tests)
+
+```bash
+./run.sh --formats wav --duration 3
+```
+
+### 8) Increase GStreamer debug verbosity
+
+```bash
+./run.sh --gst-debug 5
+```
+
+---
+
+## Pipeline Details
+
+### Recording Pipeline - audiotestsrc (WAV)
+
+```
+audiotestsrc wave=sine freq=440 volume=1.0 num-buffers=<N>
+  ! audioconvert
+  ! wavenc
+  ! filesink location=<output_file>
+```
+
+Where:
+- `wave=sine` generates sine wave test tone
+- `freq=440` sets frequency to 440Hz (A4 note)
+- `volume=1.0` sets full volume
+- `num-buffers` = (44100 * duration) / 1024
+  - Example: 10 seconds → (44100 * 10) / 1024 = 430 buffers
+
+### Recording Pipeline - audiotestsrc (FLAC)
+
+```
+audiotestsrc wave=sine freq=440 volume=1.0 num-buffers=<N>
+  ! audioconvert
+  ! flacenc
+  ! filesink location=<output_file>
+```
+
+### Recording Pipeline - pulsesrc (WAV)
+
+```
+pulsesrc volume=10
+  ! audioconvert
+  ! wavenc
+  ! filesink location=<output_file>
+```
+
+**Duration Control**: Uses `timeout` command to stop recording after specified duration (e.g., 10 seconds)
+
+### Recording Pipeline - pulsesrc (FLAC)
+
+```
+pulsesrc volume=10
+  ! audioconvert
+  ! flacenc
+  ! filesink location=<output_file>
+```
+
+**Duration Control**: Uses `timeout` command to stop recording after specified duration (e.g., 10 seconds)
+
+### Playback Pipeline (WAV)
+
+```
+filesrc location=<input_file>
+  ! wavparse
+  ! audioconvert
+  ! pulsesink volume=10
+```
+
+### Playback Pipeline (FLAC)
+
+```
+filesrc location=<input_file>
+  ! flacparse
+  ! flacdec
+  ! pulsesink volume=10
+```
+
+---
+
+## Audio Parameters
+
+The test uses these audio parameters for buffer calculation (audiotestsrc only):
+
+- **Sample Rate**: 44100 Hz (44.1 kHz - CD quality)
+- **Samples per Buffer**: 1024 (standard buffer size)
+- **Buffer Count Formula**: `(sample_rate * duration) / samples_per_buffer`
+
+Example calculations:
+- 3 seconds: (44100 * 3) / 1024 = 129 buffers
+- 10 seconds: (44100 * 10) / 1024 = 430 buffers
+- 30 seconds: (44100 * 30) / 1024 = 1291 buffers
+
+**Note**: pulsesrc tests use timeout mechanism instead of buffer count for duration control.
+
+---
+
+## Troubleshooting
+
+### A) "SKIP: Missing gstreamer runtime"
+- Ensure `gst-launch-1.0` and `gst-inspect-1.0` are installed in the image.
+
+### B) "audiotestsrc element not available"
+- Check if audiotestsrc plugin is available:
+  ```bash
+  gst-inspect-1.0 audiotestsrc
+  ```
+- Install gst-plugins-base if missing
+
+### C) "pulsesrc element not available"
+- Check if pulsesrc plugin is available:
+  ```bash
+  gst-inspect-1.0 pulsesrc
+  ```
+- Install gst-plugins-good if missing
+- Note: pulsesrc tests will be skipped if plugin is not available, but audiotestsrc tests will still run
+
+### D) "pulsesink element not available"
+- Check if pulsesink plugin is available:
+  ```bash
+  gst-inspect-1.0 pulsesink
+  ```
+- Ensure PulseAudio is installed and running:
+  ```bash
+  pulseaudio --check
+  pactl info
+  ```
+
+### E) "wavenc plugin not available" or "flacenc plugin not available"
+- Check if encoder plugins are available:
+  ```bash
+  gst-inspect-1.0 wavenc
+  gst-inspect-1.0 flacenc
+  ```
+- Install gst-plugins-good if missing
+
+### F) Playback tests skip with "recorded file not found"
+- Run record tests first: `./run.sh --mode record`
+- Or run all tests: `./run.sh --mode all`
+
+### G) Recording fails or produces small files
+- Check available disk space
+- Check `logs/Audio_Record_Playback/record_*.log` for errors
+- Try with shorter duration: `./run.sh --duration 3`
+- Increase debug level: `./run.sh --gst-debug 5`
+
+### H) "FAIL: file too small"
+- Recording may have failed silently
+- Check individual test logs in `logs/Audio_Record_Playback/`
+- Verify GStreamer plugins are properly installed
+
+### I) Playback fails with PulseAudio errors
+- Check PulseAudio status:
+  ```bash
+  pulseaudio --check
+  pactl info
+  ```
+- Restart PulseAudio if needed:
+  ```bash
+  pulseaudio --kill
+  pulseaudio --start
+  ```
+- Check for audio sinks:
+  ```bash
+  pactl list sinks short
+  ```
+
+### J) pulsesrc recording fails or produces no audio
+- Check if audio input device is available:
+  ```bash
+  pactl list sources short
+  ```
+- Test audio input manually:
+  ```bash
+  gst-launch-1.0 pulsesrc ! audioconvert ! wavenc ! filesink location=/tmp/test.wav
+  # Let it run for a few seconds, then Ctrl+C
+  aplay /tmp/test.wav
+  ```
+- Verify microphone is not muted:
+  ```bash
+  pactl list sources | grep -A 10 "Name:"
+  ```
+- Check microphone permissions and hardware connection
+
+---
+
+## Library Functions (Runner/utils/lib_gstreamer.sh)
+
+This test uses reusable helper functions from `lib_gstreamer.sh` that other GStreamer tests can leverage:
+
+### Audio Pipeline Builders
+
+**`gstreamer_build_audio_record_pipeline <source_type> <format> <output_file> [num_buffers]`**
+- Builds audio recording pipeline with specified source
+- Parameters:
+  - `source_type`: `audiotestsrc` or `pulsesrc`
+  - `format`: `wav` or `flac`
+  - `output_file`: Output file path
+  - `num_buffers`: (optional) Number of buffers for audiotestsrc (ignored for pulsesrc)
+- Returns: Complete pipeline string (or empty if format/source not supported)
+- Example:
+  ```sh
+  # audiotestsrc recording
+  NUM_BUFFERS=$(( (44100 * 10) / 1024 ))  # 10 seconds
+  pipeline=$(gstreamer_build_audio_record_pipeline "audiotestsrc" "wav" "/tmp/test.wav" "$NUM_BUFFERS")
+  gstreamer_run_gstlaunch_timeout 20 "$pipeline"
+  
+  # pulsesrc recording (uses timeout for duration control)
+  pipeline=$(gstreamer_build_audio_record_pipeline "pulsesrc" "flac" "/tmp/hw_capture.flac")
+  gstreamer_run_gstlaunch_timeout 10 "$pipeline"  # Records for 10 seconds
+  ```
+
+**`gstreamer_build_audio_playback_pipeline <format> <input_file>`**
+- Builds audio playback pipeline using pulsesink
+- Parameters:
+  - `format`: `wav` or `flac`
+  - `input_file`: Input file path
+- Returns: Complete pipeline string (or empty if format not supported)
+- Example:
+  ```sh
+  pipeline=$(gstreamer_build_audio_playback_pipeline "flac" "/tmp/test.flac")
+  gstreamer_run_gstlaunch_timeout 20 "$pipeline"
+  ```
+
+### Usage in Other Tests
+
+To use these functions in your GStreamer audio test:
+
+```sh
+#!/bin/sh
+# Source init_env and lib_gstreamer.sh
+. "$INIT_ENV"
+. "$TOOLS/functestlib.sh"
+. "$TOOLS/lib_gstreamer.sh"
+
+# Calculate buffer count for 5 seconds (audiotestsrc)
+SAMPLE_RATE=44100
+SAMPLES_PER_BUFFER=1024
+duration=5
+NUM_BUFFERS=$(( (SAMPLE_RATE * duration) / SAMPLES_PER_BUFFER ))
+
+# Build and run audiotestsrc record pipeline
+pipeline=$(gstreamer_build_audio_record_pipeline "audiotestsrc" "wav" "/tmp/output.wav" "$NUM_BUFFERS")
+if [ -n "$pipeline" ]; then
+  gstreamer_run_gstlaunch_timeout 15 "$pipeline"
+fi
+
+# Build and run pulsesrc record pipeline (timeout controls duration)
+pipeline=$(gstreamer_build_audio_record_pipeline "pulsesrc" "flac" "/tmp/hw_capture.flac")
+if [ -n "$pipeline" ]; then
+  gstreamer_run_gstlaunch_timeout 10 "$pipeline"  # 10 second recording
+fi
+
+# Build and run playback pipeline
+pipeline=$(gstreamer_build_audio_playback_pipeline "wav" "/tmp/output.wav")
+if [ -n "$pipeline" ]; then
+  gstreamer_run_gstlaunch_timeout 15 "$pipeline"
+fi
+```
+
+---
+
+## Notes for CI / LAVA
+
+- The test always exits `0`.
+- Use the `.res` file for result:
+  - `PASS` - All tests passed
+  - `FAIL` - One or more tests failed
+  - `SKIP` - No tests executed or all skipped
+- Test summary is logged showing pass/fail/skip counts
+- Individual test logs are available in `logs/Audio_Record_Playback/`
+- Recorded files are preserved in `logs/Audio_Record_Playback/recorded/` for debugging
+
+### LAVA Environment Variables
+
+The test supports these environment variables (can be set in LAVA job definition):
+
+- `AUDIO_TEST_MODE` - Test mode (all/record/playback) (default: all)
+- `AUDIO_FORMATS` - Comma-separated format list (default: `wav,flac`)
+- `AUDIO_DURATION` - Recording duration in seconds (default: 10)
+- `RUNTIMESEC` - Alternative to AUDIO_DURATION (for backward compatibility)
+- `AUDIO_GST_DEBUG` - GStreamer debug level (default: 2)
+- `GST_DEBUG_LEVEL` - Alternative to AUDIO_GST_DEBUG
+
+**Priority order for duration**: `AUDIO_DURATION` > `RUNTIMESEC` > default (10)
+
+### Test Counting
+
+- **Total tests**: 8 (when running with default wav,flac formats in all mode)
+  - 4 encode tests (2 audiotestsrc + 2 pulsesrc)
+  - 4 decode tests (2 audiotestsrc playback + 2 pulsesrc playback)
+- **Pass/Fail/Skip are mutually exclusive**: Each test increments exactly one counter
+- **Plugin unavailability**: pulsesrc tests will skip if pulsesrc plugin is not available, but audiotestsrc tests will still run
+
+---
+
+## License
+
+```
+Copyright (c) Qualcomm Technologies, Inc. and/or its subsidiaries.
+SPDX-License-Identifier: BSD-3-Clause-Clear
diff --git a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh
new file mode 100644
index 000000000..b42759979
--- /dev/null
+++ b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh
@@ -0,0 +1,669 @@
+#!/bin/sh
+# Copyright (c) Qualcomm Technologies, Inc. and/or its subsidiaries.
+# SPDX-License-Identifier: BSD-3-Clause-Clear
+#
+# Audio Record/Playback Validation using GStreamer (8 tests total)
+#
+# Test Sequence:
+#   ENCODE PHASE (4 tests):
+#     1. record_wav          - audiotestsrc → wavenc → file
+#     2. record_flac         - audiotestsrc → flacenc → file
+#     3. record_pulsesrc_wav - pulsesrc HW → wavenc → file
+#     4. record_pulsesrc_flac- pulsesrc HW → flacenc → file
+#
+#   DECODE PHASE (4 tests):
+#     5. playback_wav             - file → wavparse → pulsesink
+#     6. playback_flac            - file → flacparse → flacdec → pulsesink
+#     7. playback_pulsesrc_wav    - file → wavparse → pulsesink
+#     8. playback_pulsesrc_flac   - file → flacparse → flacdec → pulsesink
+#
+# Features:
+#   - audiotestsrc: Synthetic audio generation (uses num-buffers for duration control)
+#   - pulsesrc: Hardware audio capture (uses timeout for duration control)
+#   - pulsesink: Audio playback
+#   - Formats: WAV (wavenc/wavparse), FLAC (flacenc/flacparse/flacdec)
+#   - Duration control via AUDIO_DURATION env var (default: 10 seconds)
+#
+# Logs everything to console and local log files.
+# PASS/FAIL/SKIP is emitted to .res. Always exits 0 (LAVA-friendly).
+
+# -------------------- Configuration --------------------
+# Audio parameters for buffer calculation
+SAMPLE_RATE=44100        # 44.1 kHz
+SAMPLES_PER_BUFFER=1024  # Standard buffer size
+
+SCRIPT_DIR="$(
+  cd "$(dirname "$0")" || exit 1
+  pwd
+)"
+
+TESTNAME="Audio_Record_Playback"
+RES_FILE="${SCRIPT_DIR}/${TESTNAME}.res"
+LOG_DIR="${SCRIPT_DIR}/logs"
+OUTDIR="$LOG_DIR/$TESTNAME"
+GST_LOG="$OUTDIR/gst.log"
+DMESG_DIR="$OUTDIR/dmesg"
+RECORDED_DIR="$OUTDIR/recorded"
+
+mkdir -p "$OUTDIR" "$DMESG_DIR" "$RECORDED_DIR" >/dev/null 2>&1 || true
+: >"$RES_FILE"
+: >"$GST_LOG"
+
+SCRIPT_DIR="$(
+  cd "$(dirname "$0")" || exit 1
+  pwd
+)"
+ 
+INIT_ENV=""
+SEARCH="$SCRIPT_DIR"
+while [ "$SEARCH" != "/" ]; do
+  if [ -f "$SEARCH/init_env" ]; then
+    INIT_ENV="$SEARCH/init_env"
+    break
+  fi
+  SEARCH=$(dirname "$SEARCH")
+done
+
+RES_FILE="$SCRIPT_DIR/${TESTNAME}.res"
+
+if [ -z "${INIT_ENV:-}" ]; then
+  echo "[ERROR] Could not find init_env (starting at $SCRIPT_DIR)" >&2
+  echo "$TESTNAME SKIP" >"$RES_FILE" 2>/dev/null || true
+  exit 0
+fi
+
+if [ -z "${__INIT_ENV_LOADED:-}" ]; then
+  # shellcheck disable=SC1090
+  . "$INIT_ENV"
+  __INIT_ENV_LOADED=1
+fi
+
+# shellcheck disable=SC1091
+. "$TOOLS/functestlib.sh"
+
+# shellcheck disable=SC1091
+. "$TOOLS/lib_gstreamer.sh"
+
+result="FAIL"
+reason="unknown"
+pass_count=0
+fail_count=0
+skip_count=0
+total_tests=0
+
+# -------------------- Defaults (LAVA env vars -> defaults; CLI overrides) --------------------
+testMode="${AUDIO_TEST_MODE:-all}"
+formatList="${AUDIO_FORMATS:-wav,flac}"
+duration="${AUDIO_DURATION:-${RUNTIMESEC:-10}}"
+gstDebugLevel="${AUDIO_GST_DEBUG:-${GST_DEBUG_LEVEL:-2}}"
+
+# Calculate num_buffers based on duration
+# Formula: num_buffers = (sample_rate * duration) / samples_per_buffer
+# Example: (44100 * 10) / 1024 = 430 buffers for 10 seconds
+NUM_BUFFERS=$(( (SAMPLE_RATE * duration) / SAMPLES_PER_BUFFER ))
+
+# Validate numeric parameters (only validate if explicitly set)
+for param in AUDIO_DURATION AUDIO_GST_DEBUG GST_DEBUG_LEVEL; do
+  val=""
+  case "$param" in
+    AUDIO_DURATION) val="${AUDIO_DURATION-}" ;;
+    AUDIO_GST_DEBUG) val="${AUDIO_GST_DEBUG-}" ;;
+    GST_DEBUG_LEVEL) val="${GST_DEBUG_LEVEL-}" ;;
+  esac
+
+  if [ -n "$val" ]; then
+    case "$val" in
+      ''|*[!0-9]*)
+        log_warn "$param must be numeric (got '$val')"
+        echo "$TESTNAME SKIP" >"$RES_FILE"
+        exit 0
+        ;;
+      *)
+        if [ "$val" -le 0 ] 2>/dev/null; then
+          log_warn "$param must be positive (got '$val')"
+          echo "$TESTNAME SKIP" >"$RES_FILE"
+          exit 0
+        fi
+        ;;
+    esac
+  fi
+done
+
+cleanup() {
+  pkill -x gst-launch-1.0 >/dev/null 2>&1 || true
+}
+trap cleanup INT TERM EXIT
+
+# -------------------- Arg parse --------------------
+while [ $# -gt 0 ]; do
+  case "$1" in
+    --mode)
+      if [ $# -lt 2 ] || [ "${2#--}" != "$2" ]; then
+        log_warn "Missing/invalid value for --mode"
+        echo "$TESTNAME SKIP" >"$RES_FILE"
+        exit 0
+      fi
+      [ -n "$2" ] && testMode="$2"
+      shift 2
+      ;;
+
+    --formats)
+      if [ $# -lt 2 ] || [ "${2#--}" != "$2" ]; then
+        log_warn "Missing/invalid value for --formats"
+        echo "$TESTNAME SKIP" >"$RES_FILE"
+        exit 0
+      fi
+      [ -n "$2" ] && formatList="$2"
+      shift 2
+      ;;
+
+    --duration)
+      if [ $# -lt 2 ] || [ "${2#--}" != "$2" ]; then
+        log_warn "Missing/invalid value for --duration"
+        echo "$TESTNAME SKIP" >"$RES_FILE"
+        exit 0
+      fi
+      if [ -n "$2" ]; then
+        case "$2" in
+          ''|*[!0-9]*)
+            log_warn "Invalid --duration '$2' (must be numeric)"
+            echo "$TESTNAME SKIP" >"$RES_FILE"
+            exit 0
+            ;;
+          *)
+            duration="$2"
+            ;;
+        esac
+      fi
+      shift 2
+      ;;
+
+    --gst-debug)
+      if [ $# -lt 2 ] || [ "${2#--}" != "$2" ]; then
+        log_warn "Missing/invalid value for --gst-debug"
+        echo "$TESTNAME SKIP" >"$RES_FILE"
+        exit 0
+      fi
+      [ -n "$2" ] && gstDebugLevel="$2"
+      shift 2
+      ;;
+
+    -h|--help)
+      echo "$TESTNAME SKIP" >"$RES_FILE"
+      exit 0
+      ;;
+
+    *)
+      log_warn "Unknown argument: $1"
+      echo "$TESTNAME SKIP" >"$RES_FILE"
+      exit 0
+      ;;
+  esac
+done
+
+# -------------------- Validate parsed values --------------------
+case "$testMode" in all|record|playback) : ;; *)
+  log_warn "Invalid --mode '$testMode'"
+  echo "$TESTNAME SKIP" >"$RES_FILE"
+  exit 0
+  ;;
+esac
+
+case "$gstDebugLevel" in 1|2|3|4|5|6|7|8|9) : ;; *)
+  log_warn "Invalid --gst-debug '$gstDebugLevel' (allowed: 1-9)"
+  echo "$TESTNAME SKIP" >"$RES_FILE"
+  exit 0
+  ;;
+esac
+
+case "$duration" in
+  ''|*[!0-9]*)
+    log_warn "Invalid duration '$duration' (must be numeric)"
+    echo "$TESTNAME SKIP" >"$RES_FILE"
+    exit 0
+    ;;
+  *)
+    if [ "$duration" -le 0 ] 2>/dev/null; then
+      log_warn "Duration must be positive (got '$duration')"
+      echo "$TESTNAME SKIP" >"$RES_FILE"
+      exit 0
+    fi
+    ;;
+esac
+
+# -------------------- Pre-checks --------------------
+check_dependencies "gst-launch-1.0 gst-inspect-1.0 awk grep head sed tr stat find curl tar" >/dev/null 2>&1 || {
+  log_skip "Missing required tools (gst-launch-1.0, gst-inspect-1.0, awk, grep, head, sed, tr, stat, find, curl, tar)"
+  echo "$TESTNAME SKIP" >"$RES_FILE"
+  exit 0
+}
+
+log_info "Checking dependencies: gst-launch-1.0 gst-inspect-1.0 awk grep head sed tr stat find curl tar"
+log_info "Test: $TESTNAME"
+log_info "Mode: $testMode"
+log_info "Formats: $formatList"
+log_info "Duration: ${duration}s"
+log_info "Backend: audiotestsrc + pulsesrc for recording, pulsesink for playback"
+log_info "Audio params: sample_rate=${SAMPLE_RATE}Hz, samples_per_buffer=${SAMPLES_PER_BUFFER}"
+log_info "Calculated num_buffers: $NUM_BUFFERS (for ${duration}s duration)"
+log_info "GST debug: GST_DEBUG=$gstDebugLevel"
+log_info "Logs: $OUTDIR"
+
+# -------------------- Required element validation --------------------
+check_required_elements() {
+  if ! has_element audiotestsrc; then
+    log_warn "audiotestsrc element not available"
+    return 1
+  fi
+  if ! has_element pulsesink; then
+    log_warn "pulsesink element not available"
+    return 1
+  fi
+  return 0
+}
+
+# -------------------- Record test function (audiotestsrc) --------------------
+run_record_test() {
+  fmt="$1"
+
+  testname="record_${fmt}"
+  log_info "=========================================="
+  log_info "Running: $testname"
+  log_info "=========================================="
+
+  # Check required plugins
+  case "$fmt" in
+    wav)
+      if ! has_element wavenc; then
+        log_warn "$testname: wavenc plugin not available"
+        return 1
+      fi
+      ;;
+    flac)
+      if ! has_element flacenc; then
+        log_warn "$testname: flacenc plugin not available"
+        return 1
+      fi
+      ;;
+  esac
+
+  ext="$fmt"
+  output_file="$RECORDED_DIR/${testname}.${ext}"
+  test_log="$OUTDIR/${testname}.log"
+
+  : >"$test_log"
+
+  pipeline="$(gstreamer_build_audio_record_pipeline "audiotestsrc" "$fmt" "$output_file" "$NUM_BUFFERS")"
+
+  if [ -z "$pipeline" ]; then
+    log_fail "$testname: FAIL (could not build record pipeline)"
+    fail_count=$((fail_count + 1))
+    return 1
+  fi
+
+  log_info "Pipeline: $pipeline"
+
+  # Run recording
+  if gstreamer_run_gstlaunch_timeout "$((duration + 10))" "$pipeline" >>"$test_log" 2>&1; then
+    gstRc=0
+  else
+    gstRc=$?
+  fi
+
+  log_info "Record exit code: $gstRc"
+
+  # Check for GStreamer errors in log
+  if ! gstreamer_validate_log "$test_log" "$testname"; then
+    log_fail "$testname: FAIL (GStreamer errors detected)"
+    fail_count=$((fail_count + 1))
+    return 1
+  fi
+
+  # Check if output file was created and has content
+  if [ -f "$output_file" ] && [ -s "$output_file" ]; then
+    file_size="$(gstreamer_file_size_bytes "$output_file")"
+    log_info "Recorded file: $output_file (size: $file_size bytes)"
+
+    if [ "$file_size" -gt 1000 ]; then
+      log_pass "$testname: PASS"
+      pass_count=$((pass_count + 1))
+      return 0
+    else
+      log_fail "$testname: FAIL (file too small: $file_size bytes)"
+      fail_count=$((fail_count + 1))
+      return 1
+    fi
+  else
+    log_fail "$testname: FAIL (no output file created)"
+    fail_count=$((fail_count + 1))
+    return 1
+  fi
+}
+
+# -------------------- Playback test function --------------------
+run_playback_test() {
+  fmt="$1"
+
+  testname="playback_${fmt}"
+  log_info "=========================================="
+  log_info "Running: $testname"
+  log_info "=========================================="
+
+  ext="$fmt"
+  input_file="$RECORDED_DIR/record_${fmt}.${ext}"
+
+  if [ ! -f "$input_file" ]; then
+    log_warn "$testname: SKIP - recorded file not found: $input_file (run record first)"
+    skip_count=$((skip_count + 1))
+    return 1
+  fi
+
+  test_log="$OUTDIR/${testname}.log"
+  : >"$test_log"
+
+  pipeline="$(gstreamer_build_audio_playback_pipeline "$fmt" "$input_file")"
+
+  if [ -z "$pipeline" ]; then
+    log_fail "$testname: FAIL (could not build playback pipeline)"
+    fail_count=$((fail_count + 1))
+    return 1
+  fi
+
+  log_info "Pipeline: $pipeline"
+
+  # Run playback
+  if gstreamer_run_gstlaunch_timeout "$((duration + 10))" "$pipeline" >>"$test_log" 2>&1; then
+    gstRc=0
+  else
+    gstRc=$?
+  fi
+
+  log_info "Playback exit code: $gstRc"
+
+  # Check for GStreamer errors in log
+  if ! gstreamer_validate_log "$test_log" "$testname"; then
+    log_fail "$testname: FAIL (GStreamer errors detected)"
+    fail_count=$((fail_count + 1))
+    return 1
+  fi
+
+  # Check for successful completion (rc=0 or timeout rc which means it played to end)
+  if [ "$gstRc" -eq 0 ] || [ "$gstRc" -eq 124 ] || [ "$gstRc" -eq 143 ]; then
+    log_pass "$testname: PASS"
+    pass_count=$((pass_count + 1))
+    return 0
+  else
+    log_fail "$testname: FAIL (rc=$gstRc)"
+    fail_count=$((fail_count + 1))
+    return 1
+  fi
+}
+
+# -------------------- PulseSrc Record test function --------------------
+run_record_pulsesrc_test() {
+  fmt="$1"
+
+  testname="record_pulsesrc_${fmt}"
+  log_info "=========================================="
+  log_info "Running: $testname"
+  log_info "=========================================="
+
+  # Check required plugins
+  if ! has_element pulsesrc; then
+    log_warn "$testname: pulsesrc plugin not available"
+    skip_count=$((skip_count + 1))
+    return 1
+  fi
+
+  case "$fmt" in
+    wav)
+      if ! has_element wavenc; then
+        log_warn "$testname: wavenc plugin not available"
+        skip_count=$((skip_count + 1))
+        return 1
+      fi
+      ;;
+    flac)
+      if ! has_element flacenc; then
+        log_warn "$testname: flacenc plugin not available"
+        skip_count=$((skip_count + 1))
+        return 1
+      fi
+      ;;
+  esac
+
+  ext="$fmt"
+  output_file="$RECORDED_DIR/${testname}.${ext}"
+  test_log="$OUTDIR/${testname}.log"
+
+  : >"$test_log"
+
+  pipeline="$(gstreamer_build_audio_record_pipeline "pulsesrc" "$fmt" "$output_file")"
+
+  if [ -z "$pipeline" ]; then
+    log_fail "$testname: FAIL (could not build pulsesrc record pipeline)"
+    fail_count=$((fail_count + 1))
+    return 1
+  fi
+
+  log_info "Pipeline: $pipeline"
+
+  # Run recording with timeout
+  if gstreamer_run_gstlaunch_timeout "$duration" "$pipeline" >>"$test_log" 2>&1; then
+    gstRc=0
+  else
+    gstRc=$?
+  fi
+
+  log_info "Record exit code: $gstRc"
+
+  # Check for GStreamer errors in log
+  if ! gstreamer_validate_log "$test_log" "$testname"; then
+    log_fail "$testname: FAIL (GStreamer errors detected)"
+    fail_count=$((fail_count + 1))
+    return 1
+  fi
+
+  # Check if output file was created and has content
+  if [ -f "$output_file" ] && [ -s "$output_file" ]; then
+    file_size="$(gstreamer_file_size_bytes "$output_file")"
+    log_info "Recorded file: $output_file (size: $file_size bytes)"
+
+    if [ "$file_size" -gt 1000 ]; then
+      log_pass "$testname: PASS"
+      pass_count=$((pass_count + 1))
+      return 0
+    else
+      log_fail "$testname: FAIL (file too small: $file_size bytes)"
+      fail_count=$((fail_count + 1))
+      return 1
+    fi
+  else
+    log_fail "$testname: FAIL (no output file created)"
+    fail_count=$((fail_count + 1))
+    return 1
+  fi
+}
+
+# -------------------- PulseSrc Playback test function --------------------
+run_playback_pulsesrc_test() {
+  fmt="$1"
+
+  testname="playback_pulsesrc_${fmt}"
+  log_info "=========================================="
+  log_info "Running: $testname"
+  log_info "=========================================="
+
+  ext="$fmt"
+  input_file="$RECORDED_DIR/record_pulsesrc_${fmt}.${ext}"
+
+  if [ ! -f "$input_file" ]; then
+    log_warn "$testname: SKIP - recorded file not found: $input_file (run pulsesrc record first)"
+    skip_count=$((skip_count + 1))
+    return 1
+  fi
+
+  test_log="$OUTDIR/${testname}.log"
+  : >"$test_log"
+
+  pipeline="$(gstreamer_build_audio_playback_pipeline "$fmt" "$input_file")"
+
+  if [ -z "$pipeline" ]; then
+    log_fail "$testname: FAIL (could not build playback pipeline)"
+    fail_count=$((fail_count + 1))
+    return 1
+  fi
+
+  log_info "Pipeline: $pipeline"
+
+  # Run playback
+  if gstreamer_run_gstlaunch_timeout "$((duration + 10))" "$pipeline" >>"$test_log" 2>&1; then
+    gstRc=0
+  else
+    gstRc=$?
+  fi
+
+  log_info "Playback exit code: $gstRc"
+
+  # Check for GStreamer errors in log
+  if ! gstreamer_validate_log "$test_log" "$testname"; then
+    log_fail "$testname: FAIL (GStreamer errors detected)"
+    fail_count=$((fail_count + 1))
+    return 1
+  fi
+
+  # Check for successful completion (rc=0 or timeout rc which means it played to end)
+  if [ "$gstRc" -eq 0 ] || [ "$gstRc" -eq 124 ] || [ "$gstRc" -eq 143 ]; then
+    log_pass "$testname: PASS"
+    pass_count=$((pass_count + 1))
+    return 0
+  else
+    log_fail "$testname: FAIL (rc=$gstRc)"
+    fail_count=$((fail_count + 1))
+    return 1
+  fi
+}
+
+# -------------------- GStreamer debug capture --------------------
+export GST_DEBUG_NO_COLOR=1
+export GST_DEBUG="$gstDebugLevel"
+export GST_DEBUG_FILE="$GST_LOG"
+
+# -------------------- Main test execution --------------------
+log_info "Starting audio record/playback tests..."
+
+# Check required elements
+if ! check_required_elements; then
+  log_warn "Required GStreamer elements (audiotestsrc/pulsesink) not available"
+  echo "$TESTNAME SKIP" >"$RES_FILE"
+  exit 0
+fi
+log_info "Required GStreamer elements verified"
+
+# Parse format list
+formats=$(printf '%s' "$formatList" | tr ',' ' ')
+
+# Run ALL record/encode tests first (4 tests total)
+if [ "$testMode" = "all" ] || [ "$testMode" = "record" ]; then
+  log_info "=========================================="
+  log_info "RECORD TESTS"
+  log_info "=========================================="
+
+  # 1. Record with audiotestsrc (2 tests: wav, flac)
+  log_info "Recording with audiotestsrc..."
+  for fmt in $formats; do
+    total_tests=$((total_tests + 1))
+    run_record_test "$fmt" || true
+  done
+
+  # 2. Record with pulsesrc HW (2 tests: wav, flac)
+  log_info "Recording with pulsesrc HW..."
+  for fmt in $formats; do
+    total_tests=$((total_tests + 1))
+    run_record_pulsesrc_test "$fmt" || true
+  done
+fi
+
+# Run ALL playback/decode tests after recording (4 tests total)
+if [ "$testMode" = "all" ] || [ "$testMode" = "playback" ]; then
+  log_info "=========================================="
+  log_info "PLAYBACK TESTS"
+  log_info "=========================================="
+
+  # 3. Playback audiotestsrc recordings (2 tests: wav, flac)
+  log_info "Playing back audiotestsrc recordings..."
+  for fmt in $formats; do
+    total_tests=$((total_tests + 1))
+    run_playback_test "$fmt" || true
+  done
+
+  # 4. Playback pulsesrc recordings (2 tests: wav, flac)
+  log_info "Playing back pulsesrc recordings..."
+  for fmt in $formats; do
+    total_tests=$((total_tests + 1))
+    run_playback_pulsesrc_test "$fmt" || true
+  done
+fi
+
+# -------------------- Dmesg error scan --------------------
+log_info "=========================================="
+log_info "DMESG ERROR SCAN"
+log_info "=========================================="
+
+module_regex="audio|sound|pulse|codec|dsp"
+exclude_regex="dummy regulator|supply [^ ]+ not found|using dummy regulator"
+
+if command -v scan_dmesg_errors >/dev/null 2>&1; then
+  scan_dmesg_errors "$DMESG_DIR" "$module_regex" "$exclude_regex" || true
+
+  if [ -s "$DMESG_DIR/dmesg_errors.log" ]; then
+    log_warn "dmesg scan found audio-related warnings or errors in $DMESG_DIR/dmesg_errors.log"
+  else
+    log_info "No relevant audio-related errors found in dmesg"
+  fi
+else
+  log_info "scan_dmesg_errors not available, skipping dmesg scan"
+fi
+
+# -------------------- Summary --------------------
+log_info "=========================================="
+log_info "TEST SUMMARY"
+log_info "=========================================="
+actual_total=$((pass_count + fail_count + skip_count))
+log_info "Total testcases: $actual_total"
+log_info "Passed: $pass_count"
+log_info "Failed: $fail_count"
+log_info "Skipped: $skip_count"
+
+# -------------------- Emit result --------------------
+if [ "$fail_count" -eq 0 ] && [ "$pass_count" -gt 0 ]; then
+  result="PASS"
+  if [ "$skip_count" -gt 0 ]; then
+    reason="No failures (passed: $pass_count, failed: $fail_count, skipped: $skip_count, total: $actual_total)"
+  else
+    reason="All tests passed ($pass_count/$actual_total)"
+  fi
+elif [ "$fail_count" -gt 0 ]; then
+  result="FAIL"
+  reason="Some tests failed (passed: $pass_count, failed: $fail_count, skipped: $skip_count, total: $actual_total)"
+else
+  result="SKIP"
+  reason="No tests passed (skipped: $skip_count, total: $actual_total)"
+fi
+
+case "$result" in
+  PASS)
+    log_pass "$TESTNAME $result: $reason"
+    echo "$TESTNAME PASS" >"$RES_FILE"
+    ;;
+  FAIL)
+    log_fail "$TESTNAME $result: $reason"
+    echo "$TESTNAME FAIL" >"$RES_FILE"
+    ;;
+  *)
+    log_warn "$TESTNAME $result: $reason"
+    echo "$TESTNAME SKIP" >"$RES_FILE"
+    ;;
+esac
+
+exit 0
diff --git a/Runner/utils/lib_gstreamer.sh b/Runner/utils/lib_gstreamer.sh
index 2af696496..56166aa03 100755
--- a/Runner/utils/lib_gstreamer.sh
+++ b/Runner/utils/lib_gstreamer.sh
@@ -471,30 +471,80 @@ gstreamer_run_gstlaunch_timeout() {
   return $?
 }
 
-# -------------------- Playback pipeline builder (backend-aware) --------------------
-# gstreamer_build_playback_pipeline <backend> <format> <file> <capsStrOrEmpty> <alsadev>
-gstreamer_build_playback_pipeline() {
-  backend="$1"
-  format="$2"
-  file="$3"
-  capsStr="$4"
-  alsadev="$5"
+# -------------------- Audio Record/Playback pipeline builders --------------------
+# gstreamer_build_audio_record_pipeline <source_type> <format> <output_file> [num_buffers]
+# Builds audio recording pipeline with specified source
+# Parameters:
+#   source_type: "audiotestsrc" or "pulsesrc"
+#   format: "wav" or "flac"
+#   output_file: path to output file
+#   num_buffers: (optional) number of buffers for audiotestsrc (ignored for pulsesrc)
+# Prints: pipeline string or empty if format/source not supported
+gstreamer_build_audio_record_pipeline() {
+  source_type="$1"
+  fmt="$2"
+  output_file="$3"
+  num_buffers="${4:-}"
+
+  # Build source element
+  case "$source_type" in
+    audiotestsrc)
+      # num_buffers is required for audiotestsrc
+      if [ -z "$num_buffers" ]; then
+        printf '%s\n' ""
+        return 1
+      fi
+      source_elem="audiotestsrc wave=sine freq=440 volume=1.0 num-buffers=${num_buffers}"
+      ;;
+    pulsesrc)
+      # pulsesrc doesn't use num_buffers (continuous capture until timeout)
+      source_elem="pulsesrc volume=10"
+      ;;
+    *)
+      printf '%s\n' ""
+      return 1
+      ;;
+  esac
 
-  [ -n "$alsadev" ] || alsadev="default"
+  # Build encoder element
+  case "$fmt" in
+    wav)
+      encoder_elem="wavenc"
+      ;;
+    flac)
+      encoder_elem="flacenc"
+      ;;
+    *)
+      printf '%s\n' ""
+      return 1
+      ;;
+  esac
 
-  dec="$(gstreamer_pick_decode_chain "$format")"
-  sinkElem="$(gstreamer_pick_sink_element "$backend" "$alsadev")"
-  if [ -z "$sinkElem" ]; then
-    printf '%s\n' ""
-    return 0
-  fi
+  # Construct complete pipeline
+  printf '%s\n' "${source_elem} ! audioconvert ! ${encoder_elem} ! filesink location=${output_file}"
+  return 0
+}
 
-  if [ -n "$capsStr" ]; then
-    printf '%s\n' "filesrc location=${file} ! ${dec} ! audioconvert ! audioresample ! ${capsStr} ! ${sinkElem}"
-    return 0
-  fi
+# gstreamer_build_audio_playback_pipeline <format> <input_file>
+# Builds audio playback pipeline using pulsesink
+# Supports: wav, flac formats
+# Prints: pipeline string or empty if format not supported
+gstreamer_build_audio_playback_pipeline() {
+  fmt="$1"
+  input_file="$2"
 
-  printf '%s\n' "filesrc location=${file} ! ${dec} ! audioconvert ! audioresample ! ${sinkElem}"
+  case "$fmt" in
+    wav)
+      printf '%s\n' "filesrc location=${input_file} ! wavparse ! audioconvert ! pulsesink volume=10"
+      ;;
+    flac)
+      printf '%s\n' "filesrc location=${input_file} ! flacparse ! flacdec ! pulsesink volume=10"
+      ;;
+    *)
+      printf '%s\n' ""
+      return 1
+      ;;
+  esac
   return 0
 }
 

From 7c601700acb1ec08da2d58fcbed20e1c2a3213da Mon Sep 17 00:00:00 2001
From: Nitin Nakka <nitinn@qti.qualcomm.com>
Date: Sat, 7 Mar 2026 01:36:13 +0530
Subject: [PATCH 2/7] Modified video cleanup to better handle killing of
 gst-launch, in audio script added handle to check file size to be minimum
 1000 bytes before initiating playback or else to skip it

Signed-off-by: Nitin Nakka <nitinn@qti.qualcomm.com>
---
 .../Audio/Audio_Record_Playback/run.sh        | 21 ++++++++++++++++++-
 .../Video/Video_Encode_Decode/run.sh          |  5 ++++-
 2 files changed, 24 insertions(+), 2 deletions(-)

diff --git a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh
index b42759979..076f0aa49 100644
--- a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh
+++ b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh
@@ -130,7 +130,10 @@ for param in AUDIO_DURATION AUDIO_GST_DEBUG GST_DEBUG_LEVEL; do
 done
 
 cleanup() {
-  pkill -x gst-launch-1.0 >/dev/null 2>&1 || true
+  # Best-effort: try to kill only children first; fall back to name-based kill
+  if ! pkill -P "$$" -x gst-launch-1.0 >/dev/null 2>&1; then
+    pkill -x gst-launch-1.0 >/dev/null 2>&1 || true
+  fi
 }
 trap cleanup INT TERM EXIT
 
@@ -358,6 +361,14 @@ run_playback_test() {
     return 1
   fi
 
+  # Check if file has minimum content (same threshold as recording: 1000 bytes)
+  file_size="$(gstreamer_file_size_bytes "$input_file")"
+  if [ "$file_size" -le 1000 ]; then
+    log_warn "$testname: SKIP - recorded file too small: $file_size bytes (recording likely failed)"
+    skip_count=$((skip_count + 1))
+    return 1
+  fi
+
   test_log="$OUTDIR/${testname}.log"
   : >"$test_log"
 
@@ -503,6 +514,14 @@ run_playback_pulsesrc_test() {
     return 1
   fi
 
+  # Check if file has minimum content (same threshold as recording: 1000 bytes)
+  file_size="$(gstreamer_file_size_bytes "$input_file")"
+  if [ "$file_size" -le 1000 ]; then
+    log_warn "$testname: SKIP - recorded file too small: $file_size bytes (pulsesrc recording likely failed)"
+    skip_count=$((skip_count + 1))
+    return 1
+  fi
+
   test_log="$OUTDIR/${testname}.log"
   : >"$test_log"
 
diff --git a/Runner/suites/Multimedia/GSTreamer/Video/Video_Encode_Decode/run.sh b/Runner/suites/Multimedia/GSTreamer/Video/Video_Encode_Decode/run.sh
index 318136a74..245c20a35 100755
--- a/Runner/suites/Multimedia/GSTreamer/Video/Video_Encode_Decode/run.sh
+++ b/Runner/suites/Multimedia/GSTreamer/Video/Video_Encode_Decode/run.sh
@@ -111,7 +111,10 @@ for param in VIDEO_DURATION RUNTIMESEC VIDEO_FRAMERATE VIDEO_GST_DEBUG GST_DEBUG
 done
 
 cleanup() {
-  pkill -x gst-launch-1.0 >/dev/null 2>&1 || true
+  # Best-effort: try to kill only children first; fall back to name-based kill
+  if ! pkill -P "$$" -x gst-launch-1.0 >/dev/null 2>&1; then
+    pkill -x gst-launch-1.0 >/dev/null 2>&1 || true
+  fi
 }
 trap cleanup INT TERM EXIT
 

From afa337cc5d15ae34e1b4db276ea42a8f7a1881e6 Mon Sep 17 00:00:00 2001
From: Nitin Nakka <nitinn@qti.qualcomm.com>
Date: Sat, 7 Mar 2026 08:16:49 +0530
Subject: [PATCH 3/7] Fix: restore executable permission for
 Audio_Record_Playback run.sh

Signed-off-by: Nitin Nakka <nitinn@qti.qualcomm.com>
---
 .../Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh       | 0
 1 file changed, 0 insertions(+), 0 deletions(-)
 mode change 100644 => 100755 Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh

diff --git a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh
old mode 100644
new mode 100755

From bec6a348ef80d2f7d35b97d62dea133d85969f7b Mon Sep 17 00:00:00 2001
From: Nitin Nakka <nitinn@qti.qualcomm.com>
Date: Sun, 8 Mar 2026 00:54:57 +0530
Subject: [PATCH 4/7] Added two new playback support tests in audio for ogg and
 mp3 formats incl corresponding lib gstreamer

Signed-off-by: Nitin Nakka <nitinn@qti.qualcomm.com>
---
 .../Audio/Audio_Record_Playback/run.sh        | 182 +++++++++++++++++-
 Runner/utils/lib_gstreamer.sh                 |  23 ++-
 2 files changed, 195 insertions(+), 10 deletions(-)

diff --git a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh
index 076f0aa49..623a14816 100755
--- a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh
+++ b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh
@@ -2,7 +2,7 @@
 # Copyright (c) Qualcomm Technologies, Inc. and/or its subsidiaries.
 # SPDX-License-Identifier: BSD-3-Clause-Clear
 #
-# Audio Record/Playback Validation using GStreamer (8 tests total)
+# Audio Record/Playback Validation using GStreamer (10 tests total)
 #
 # Test Sequence:
 #   ENCODE PHASE (4 tests):
@@ -11,17 +11,20 @@
 #     3. record_pulsesrc_wav - pulsesrc HW → wavenc → file
 #     4. record_pulsesrc_flac- pulsesrc HW → flacenc → file
 #
-#   DECODE PHASE (4 tests):
+#   DECODE PHASE (6 tests):
 #     5. playback_wav             - file → wavparse → pulsesink
 #     6. playback_flac            - file → flacparse → flacdec → pulsesink
 #     7. playback_pulsesrc_wav    - file → wavparse → pulsesink
 #     8. playback_pulsesrc_flac   - file → flacparse → flacdec → pulsesink
+#     9. playback_sample_ogg      - file → oggdemux → vorbisdec → pulsesink
+#    10. playback_sample_mp3      - file → mpegaudioparse → mpg123audiodec → pulsesink
 #
 # Features:
 #   - audiotestsrc: Synthetic audio generation (uses num-buffers for duration control)
 #   - pulsesrc: Hardware audio capture (uses timeout for duration control)
 #   - pulsesink: Audio playback
-#   - Formats: WAV (wavenc/wavparse), FLAC (flacenc/flacparse/flacdec)
+#   - Formats: WAV (wavenc/wavparse), FLAC (flacenc/flacparse/flacdec), OGG (oggdemux/vorbisdec), MP3 (mpegaudioparse/mpg123audiodec)
+#   - Test file provisioning via URL download or local path
 #   - Duration control via AUDIO_DURATION env var (default: 10 seconds)
 #
 # Logs everything to console and local log files.
@@ -96,6 +99,8 @@ testMode="${AUDIO_TEST_MODE:-all}"
 formatList="${AUDIO_FORMATS:-wav,flac}"
 duration="${AUDIO_DURATION:-${RUNTIMESEC:-10}}"
 gstDebugLevel="${AUDIO_GST_DEBUG:-${GST_DEBUG_LEVEL:-2}}"
+clipUrl="${AUDIO_CLIP_URL:-https://github.com/qualcomm-linux/qcom-linux-testkit/releases/download/GST-Audio-Files-v1.0/audio_clips_gst.tar.gz}"
+clipPath="${AUDIO_CLIP_PATH:-}"
 
 # Calculate num_buffers based on duration
 # Formula: num_buffers = (sample_rate * duration) / samples_per_buffer
@@ -191,6 +196,26 @@ while [ $# -gt 0 ]; do
       shift 2
       ;;
 
+    --clip-url)
+      if [ $# -lt 2 ] || [ "${2#--}" != "$2" ]; then
+        log_warn "Missing/invalid value for --clip-url"
+        echo "$TESTNAME SKIP" >"$RES_FILE"
+        exit 0
+      fi
+      [ -n "$2" ] && clipUrl="$2"
+      shift 2
+      ;;
+
+    --clip-path)
+      if [ $# -lt 2 ] || [ "${2#--}" != "$2" ]; then
+        log_warn "Missing/invalid value for --clip-path"
+        echo "$TESTNAME SKIP" >"$RES_FILE"
+        exit 0
+      fi
+      [ -n "$2" ] && clipPath="$2"
+      shift 2
+      ;;
+
     -h|--help)
       echo "$TESTNAME SKIP" >"$RES_FILE"
       exit 0
@@ -563,11 +588,152 @@ run_playback_pulsesrc_test() {
   fi
 }
 
+# -------------------- Test file playback test function (OGG/MP3) --------------------
+run_playback_ogg_mp3_test() {
+  fmt="$1"
+  
+  testname="playback_sample_${fmt}"
+  log_info "=========================================="
+  log_info "Running: $testname"
+  log_info "=========================================="
+  
+  # Determine input file based on format
+  case "$fmt" in
+    ogg)
+      input_file="$OUTDIR/sample_audio.ogg"
+      ;;
+    mp3)
+      input_file="$OUTDIR/sample_audio.mp3"
+      ;;
+    *)
+      log_warn "$testname: SKIP - unsupported format: $fmt"
+      skip_count=$((skip_count + 1))
+      return 1
+      ;;
+  esac
+  
+  if [ ! -f "$input_file" ]; then
+    log_warn "$testname: SKIP - Test file not found: $input_file"
+    skip_count=$((skip_count + 1))
+    return 1
+  fi
+  
+  # Check if file has minimum content
+  file_size="$(gstreamer_file_size_bytes "$input_file")"
+  if [ "$file_size" -le 1000 ]; then
+    log_warn "$testname: SKIP - Test file too small: $file_size bytes"
+    skip_count=$((skip_count + 1))
+    return 1
+  fi
+  
+  test_log="$OUTDIR/${testname}.log"
+  : >"$test_log"
+  
+  pipeline="$(gstreamer_build_audio_playback_pipeline "$fmt" "$input_file")"
+  
+  if [ -z "$pipeline" ]; then
+    log_fail "$testname: FAIL (could not build playback pipeline - format not supported or elements missing)"
+    fail_count=$((fail_count + 1))
+    return 1
+  fi
+  
+  log_info "Pipeline: $pipeline"
+  
+  # Run playback
+  if gstreamer_run_gstlaunch_timeout "$((duration + 10))" "$pipeline" >>"$test_log" 2>&1; then
+    gstRc=0
+  else
+    gstRc=$?
+  fi
+  
+  log_info "Playback exit code: $gstRc"
+  
+  # Check for GStreamer errors in log
+  if ! gstreamer_validate_log "$test_log" "$testname"; then
+    log_fail "$testname: FAIL (GStreamer errors detected)"
+    fail_count=$((fail_count + 1))
+    return 1
+  fi
+  
+  # Check for successful completion
+  if [ "$gstRc" -eq 0 ] || [ "$gstRc" -eq 124 ] || [ "$gstRc" -eq 143 ]; then
+    log_pass "$testname: PASS"
+    pass_count=$((pass_count + 1))
+    return 0
+  else
+    log_fail "$testname: FAIL (rc=$gstRc)"
+    fail_count=$((fail_count + 1))
+    return 1
+  fi
+}
+
 # -------------------- GStreamer debug capture --------------------
 export GST_DEBUG_NO_COLOR=1
 export GST_DEBUG="$gstDebugLevel"
 export GST_DEBUG_FILE="$GST_LOG"
 
+# -------------------- Test file provisioning (OGG/MP3) --------------------
+provision_test_files() {
+  log_info "=========================================="
+  log_info "TEST FILE PROVISIONING"
+  log_info "=========================================="
+  
+  sample_ogg="$OUTDIR/sample_audio.ogg"
+  sample_mp3="$OUTDIR/sample_audio.mp3"
+  
+  # Check if Test files already exist
+  if [ -f "$sample_ogg" ] && [ -f "$sample_mp3" ]; then
+    log_info "Test files already exist"
+  else
+    # Try to get Test files from provided path or URL
+    if [ -n "$clipPath" ]; then
+      log_info "Attempting to get Test files from local path: $clipPath"
+      if [ -f "$clipPath/sample_audio.ogg" ]; then
+        cp "$clipPath/sample_audio.ogg" "$sample_ogg" 2>/dev/null || true
+        log_info "Sample OGG file copied from local path"
+      fi
+      if [ -f "$clipPath/sample_audio.mp3" ]; then
+        cp "$clipPath/sample_audio.mp3" "$sample_mp3" 2>/dev/null || true
+        log_info "Sample MP3 file copied from local path"
+      fi
+    fi
+    
+    # If not found locally, try URL download
+    if [ ! -f "$sample_ogg" ] || [ ! -f "$sample_mp3" ]; then
+      log_info "Test files not found locally; attempting download from URL..."
+      if extract_tar_from_url "$clipUrl" "$OUTDIR"; then
+        log_pass "Test files downloaded and extracted successfully"
+      else
+        log_warn "Test file download failed (offline or URL issue)"
+      fi
+    fi
+  fi
+  
+  # Check what we have
+  have_ogg=0
+  have_mp3=0
+  
+  if [ -f "$sample_ogg" ]; then
+    size=$(gstreamer_file_size_bytes "$sample_ogg")
+    if [ "$size" -gt 1000 ]; then
+      have_ogg=1
+      log_info "OGG Test file available (size: $size bytes)"
+    fi
+  fi
+  
+  if [ -f "$sample_mp3" ]; then
+    size=$(gstreamer_file_size_bytes "$sample_mp3")
+    if [ "$size" -gt 1000 ]; then
+      have_mp3=1
+      log_info "MP3 Test file available (size: $size bytes)"
+    fi
+  fi
+  
+  if [ "$have_ogg" -eq 0 ] && [ "$have_mp3" -eq 0 ]; then
+    log_warn "No Test files (OGG/MP3) available for playback tests"
+  fi
+}
+
 # -------------------- Main test execution --------------------
 log_info "Starting audio record/playback tests..."
 
@@ -579,6 +745,9 @@ if ! check_required_elements; then
 fi
 log_info "Required GStreamer elements verified"
 
+# Provision Test files for OGG/MP3 playback tests
+provision_test_files
+
 # Parse format list
 formats=$(printf '%s' "$formatList" | tr ',' ' ')
 
@@ -622,6 +791,13 @@ if [ "$testMode" = "all" ] || [ "$testMode" = "playback" ]; then
     total_tests=$((total_tests + 1))
     run_playback_pulsesrc_test "$fmt" || true
   done
+  
+  # 5. Playback Test files (2 tests: ogg, mp3)
+  log_info "Playing back Test files (OGG/MP3)..."
+  for fmt in ogg mp3; do
+    total_tests=$((total_tests + 1))
+    run_playback_ogg_mp3_test "$fmt" || true
+  done
 fi
 
 # -------------------- Dmesg error scan --------------------
diff --git a/Runner/utils/lib_gstreamer.sh b/Runner/utils/lib_gstreamer.sh
index 56166aa03..9d5b622a9 100755
--- a/Runner/utils/lib_gstreamer.sh
+++ b/Runner/utils/lib_gstreamer.sh
@@ -527,25 +527,34 @@ gstreamer_build_audio_record_pipeline() {
 
 # gstreamer_build_audio_playback_pipeline <format> <input_file>
 # Builds audio playback pipeline using pulsesink
-# Supports: wav, flac formats
+# Supports: wav, flac, ogg, mp3 formats
 # Prints: pipeline string or empty if format not supported
 gstreamer_build_audio_playback_pipeline() {
-  fmt="$1"
-  input_file="$2"
+  _fmt="$1"
+  _input_file="$2"
 
-  case "$fmt" in
+  case "$_fmt" in
     wav)
-      printf '%s\n' "filesrc location=${input_file} ! wavparse ! audioconvert ! pulsesink volume=10"
+      printf '%s\n' "filesrc location=${_input_file} ! wavparse ! audioconvert ! pulsesink volume=10"
+      return 0
       ;;
     flac)
-      printf '%s\n' "filesrc location=${input_file} ! flacparse ! flacdec ! pulsesink volume=10"
+      printf '%s\n' "filesrc location=${_input_file} ! flacparse ! flacdec ! audioconvert ! pulsesink volume=10"
+      return 0
+      ;;
+    ogg)
+      printf '%s\n' "filesrc location=${_input_file} ! oggdemux ! vorbisdec ! audioconvert ! pulsesink volume=10"
+      return 0
+      ;;
+    mp3)
+      printf '%s\n' "filesrc location=${_input_file} ! mpegaudioparse ! mpg123audiodec ! audioconvert ! pulsesink volume=10"
+      return 0
       ;;
     *)
       printf '%s\n' ""
       return 1
       ;;
   esac
-  return 0
 }
 
 # -------------------- GStreamer error log checker --------------------

From 9d7004f458eb133dd5ec30a61c1d21d0b149b861 Mon Sep 17 00:00:00 2001
From: Nitin Nakka <nitinn@qti.qualcomm.com>
Date: Tue, 10 Mar 2026 01:46:51 +0530
Subject: [PATCH 5/7] Comments on audio test addressed, back ported some of
 these items to existing video and display tests as well. Yaml files updated
 for all tests by removing || true in res line

Signed-off-by: Nitin Nakka <nitinn@qti.qualcomm.com>
---
 .../Audio_Record_Playback.yaml                |  2 +-
 .../Audio/Audio_Record_Playback/run.sh        | 97 +++++++++++++++----
 .../Waylandsink_Playback.yaml                 |  2 +-
 .../Display/Waylandsink_Playback/run.sh       |  2 -
 .../Video_Encode_Decode.yaml                  |  2 +-
 .../Video/Video_Encode_Decode/run.sh          | 95 +++++++++++++++---
 Runner/utils/lib_gstreamer.sh                 | 29 ++++++
 7 files changed, 191 insertions(+), 38 deletions(-)

diff --git a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/Audio_Record_Playback.yaml b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/Audio_Record_Playback.yaml
index 3bdc919ee..4aadc832f 100644
--- a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/Audio_Record_Playback.yaml
+++ b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/Audio_Record_Playback.yaml
@@ -23,4 +23,4 @@ run:
     - cd Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/
     - export AUDIO_TEST_MODE AUDIO_FORMATS AUDIO_DURATION AUDIO_GST_DEBUG
     - ./run.sh --mode "${AUDIO_TEST_MODE}" --formats "${AUDIO_FORMATS}" --duration "${AUDIO_DURATION}" --gst-debug "${AUDIO_GST_DEBUG}" || true
-    - $REPO_PATH/Runner/utils/send-to-lava.sh Audio_Record_Playback.res || true
+    - $REPO_PATH/Runner/utils/send-to-lava.sh Audio_Record_Playback.res
diff --git a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh
index 623a14816..7b600e858 100755
--- a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh
+++ b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh
@@ -51,11 +51,6 @@ RECORDED_DIR="$OUTDIR/recorded"
 mkdir -p "$OUTDIR" "$DMESG_DIR" "$RECORDED_DIR" >/dev/null 2>&1 || true
 : >"$RES_FILE"
 : >"$GST_LOG"
-
-SCRIPT_DIR="$(
-  cd "$(dirname "$0")" || exit 1
-  pwd
-)"
  
 INIT_ENV=""
 SEARCH="$SCRIPT_DIR"
@@ -67,8 +62,6 @@ while [ "$SEARCH" != "/" ]; do
   SEARCH=$(dirname "$SEARCH")
 done
 
-RES_FILE="$SCRIPT_DIR/${TESTNAME}.res"
-
 if [ -z "${INIT_ENV:-}" ]; then
   echo "[ERROR] Could not find init_env (starting at $SCRIPT_DIR)" >&2
   echo "$TESTNAME SKIP" >"$RES_FILE" 2>/dev/null || true
@@ -217,7 +210,76 @@ while [ $# -gt 0 ]; do
       ;;
 
     -h|--help)
-      echo "$TESTNAME SKIP" >"$RES_FILE"
+      cat <<EOF
+Usage: $0 [OPTIONS]
+
+Audio Record/Playback Validation using GStreamer
+
+OPTIONS:
+  --mode <all|record|playback>
+                        Test mode (default: all)
+                        - all: Run both record and playback tests
+                        - record: Run only recording tests
+                        - playback: Run only playback tests
+
+  --formats <format1,format2,...>
+                        Comma-separated list of audio formats to test
+                        (default: wav,flac)
+                        Supported: wav, flac
+
+  --duration <seconds>  Duration for recording/playback in seconds
+                        (default: 10)
+
+  --gst-debug <level>   GStreamer debug level (1-9)
+                        (default: 2)
+
+  --clip-url <url>      URL to download test audio files (OGG/MP3)
+                        (default: GitHub release URL)
+
+  --clip-path <path>    Local path to test audio files
+                        (overrides --clip-url if files exist)
+
+  -h, --help            Display this help message
+
+ENVIRONMENT VARIABLES:
+  AUDIO_TEST_MODE       Same as --mode
+  AUDIO_FORMATS         Same as --formats
+  AUDIO_DURATION        Same as --duration
+  AUDIO_GST_DEBUG       Same as --gst-debug
+  AUDIO_CLIP_URL        Same as --clip-url
+  AUDIO_CLIP_PATH       Same as --clip-path
+  GST_DEBUG_LEVEL       Alternative to AUDIO_GST_DEBUG
+  RUNTIMESEC            Alternative to AUDIO_DURATION
+
+EXAMPLES:
+  # Run all tests with default settings
+  $0
+
+  # Run only recording tests for 5 seconds
+  $0 --mode record --duration 5
+
+  # Test only WAV format
+  $0 --formats wav
+
+  # Use local test files
+  $0 --clip-path /path/to/audio/files
+
+TEST SEQUENCE:
+  ENCODE PHASE (4 tests):
+    1. record_wav          - audiotestsrc → wavenc → file
+    2. record_flac         - audiotestsrc → flacenc → file
+    3. record_pulsesrc_wav - pulsesrc HW → wavenc → file
+    4. record_pulsesrc_flac- pulsesrc HW → flacenc → file
+
+  DECODE PHASE (6 tests):
+    5. playback_wav             - file → wavparse → pulsesink
+    6. playback_flac            - file → flacparse → flacdec → pulsesink
+    7. playback_pulsesrc_wav    - file → wavparse → pulsesink
+    8. playback_pulsesrc_flac   - file → flacparse → flacdec → pulsesink
+    9. playback_sample_ogg      - file → oggdemux → vorbisdec → pulsesink
+   10. playback_sample_mp3      - file → mpegaudioparse → mpg123audiodec → pulsesink
+
+EOF
       exit 0
       ;;
 
@@ -423,8 +485,8 @@ run_playback_test() {
     return 1
   fi
 
-  # Check for successful completion (rc=0 or timeout rc which means it played to end)
-  if [ "$gstRc" -eq 0 ] || [ "$gstRc" -eq 124 ] || [ "$gstRc" -eq 143 ]; then
+  # Check for successful completion
+  if [ "$gstRc" -eq 0 ]; then
     log_pass "$testname: PASS"
     pass_count=$((pass_count + 1))
     return 0
@@ -576,8 +638,8 @@ run_playback_pulsesrc_test() {
     return 1
   fi
 
-  # Check for successful completion (rc=0 or timeout rc which means it played to end)
-  if [ "$gstRc" -eq 0 ] || [ "$gstRc" -eq 124 ] || [ "$gstRc" -eq 143 ]; then
+  # Check for successful completion
+  if [ "$gstRc" -eq 0 ]; then
     log_pass "$testname: PASS"
     pass_count=$((pass_count + 1))
     return 0
@@ -824,8 +886,7 @@ fi
 log_info "=========================================="
 log_info "TEST SUMMARY"
 log_info "=========================================="
-actual_total=$((pass_count + fail_count + skip_count))
-log_info "Total testcases: $actual_total"
+log_info "Total testcases: $total_tests"
 log_info "Passed: $pass_count"
 log_info "Failed: $fail_count"
 log_info "Skipped: $skip_count"
@@ -834,16 +895,16 @@ log_info "Skipped: $skip_count"
 if [ "$fail_count" -eq 0 ] && [ "$pass_count" -gt 0 ]; then
   result="PASS"
   if [ "$skip_count" -gt 0 ]; then
-    reason="No failures (passed: $pass_count, failed: $fail_count, skipped: $skip_count, total: $actual_total)"
+    reason="No failures (passed: $pass_count, failed: $fail_count, skipped: $skip_count, total: $total_tests)"
   else
-    reason="All tests passed ($pass_count/$actual_total)"
+    reason="All tests passed ($pass_count/$total_tests)"
   fi
 elif [ "$fail_count" -gt 0 ]; then
   result="FAIL"
-  reason="Some tests failed (passed: $pass_count, failed: $fail_count, skipped: $skip_count, total: $actual_total)"
+  reason="Some tests failed (passed: $pass_count, failed: $fail_count, skipped: $skip_count, total: $total_tests)"
 else
   result="SKIP"
-  reason="No tests passed (skipped: $skip_count, total: $actual_total)"
+  reason="No tests passed (skipped: $skip_count, total: $total_tests)"
 fi
 
 case "$result" in
diff --git a/Runner/suites/Multimedia/GSTreamer/Display/Waylandsink_Playback/Waylandsink_Playback.yaml b/Runner/suites/Multimedia/GSTreamer/Display/Waylandsink_Playback/Waylandsink_Playback.yaml
index fcadd9b8a..e25a435be 100644
--- a/Runner/suites/Multimedia/GSTreamer/Display/Waylandsink_Playback/Waylandsink_Playback.yaml
+++ b/Runner/suites/Multimedia/GSTreamer/Display/Waylandsink_Playback/Waylandsink_Playback.yaml
@@ -25,4 +25,4 @@ run:
     - cd Runner/suites/Multimedia/GSTreamer/Display/Waylandsink_Playback/
     - export VIDEO_DURATION VIDEO_PATTERN VIDEO_WIDTH VIDEO_HEIGHT VIDEO_FRAMERATE VIDEO_GST_DEBUG
     - ./run.sh --resolution "${VIDEO_WIDTH}x${VIDEO_HEIGHT}" --pattern "${VIDEO_PATTERN}" --duration "${VIDEO_DURATION}" --framerate "${VIDEO_FRAMERATE}" --gst-debug "${VIDEO_GST_DEBUG}" || true
-    - $REPO_PATH/Runner/utils/send-to-lava.sh Waylandsink_Playback.res || true
+    - $REPO_PATH/Runner/utils/send-to-lava.sh Waylandsink_Playback.res
diff --git a/Runner/suites/Multimedia/GSTreamer/Display/Waylandsink_Playback/run.sh b/Runner/suites/Multimedia/GSTreamer/Display/Waylandsink_Playback/run.sh
index 8c18ee1ba..6fdbb1327 100755
--- a/Runner/suites/Multimedia/GSTreamer/Display/Waylandsink_Playback/run.sh
+++ b/Runner/suites/Multimedia/GSTreamer/Display/Waylandsink_Playback/run.sh
@@ -38,8 +38,6 @@ while [ "$SEARCH" != "/" ]; do
   SEARCH=$(dirname "$SEARCH")
 done
  
-RES_FILE="$SCRIPT_DIR/${TESTNAME}.res"
- 
 if [ -z "${INIT_ENV:-}" ]; then
   echo "[ERROR] Could not find init_env (starting at $SCRIPT_DIR)" >&2
   echo "$TESTNAME SKIP" >"$RES_FILE" 2>/dev/null || true
diff --git a/Runner/suites/Multimedia/GSTreamer/Video/Video_Encode_Decode/Video_Encode_Decode.yaml b/Runner/suites/Multimedia/GSTreamer/Video/Video_Encode_Decode/Video_Encode_Decode.yaml
index ac8260ac6..2c33cb45a 100644
--- a/Runner/suites/Multimedia/GSTreamer/Video/Video_Encode_Decode/Video_Encode_Decode.yaml
+++ b/Runner/suites/Multimedia/GSTreamer/Video/Video_Encode_Decode/Video_Encode_Decode.yaml
@@ -27,4 +27,4 @@ run:
     - export VIDEO_TEST_MODE VIDEO_CODECS VIDEO_RESOLUTIONS VIDEO_DURATION VIDEO_FRAMERATE
     - export VIDEO_STACK VIDEO_GST_DEBUG VIDEO_CLIP_URL VIDEO_CLIP_PATH
     - ./run.sh --mode "${VIDEO_TEST_MODE}" --codecs "${VIDEO_CODECS}" --resolutions "${VIDEO_RESOLUTIONS}" --duration "${VIDEO_DURATION}" --framerate "${VIDEO_FRAMERATE}" --stack "${VIDEO_STACK}" --gst-debug "${VIDEO_GST_DEBUG}" --clip-url "${VIDEO_CLIP_URL}" --clip-path "${VIDEO_CLIP_PATH}" || true
-    - $REPO_PATH/Runner/utils/send-to-lava.sh Video_Encode_Decode.res || true
+    - $REPO_PATH/Runner/utils/send-to-lava.sh Video_Encode_Decode.res
diff --git a/Runner/suites/Multimedia/GSTreamer/Video/Video_Encode_Decode/run.sh b/Runner/suites/Multimedia/GSTreamer/Video/Video_Encode_Decode/run.sh
index 245c20a35..47de483b9 100755
--- a/Runner/suites/Multimedia/GSTreamer/Video/Video_Encode_Decode/run.sh
+++ b/Runner/suites/Multimedia/GSTreamer/Video/Video_Encode_Decode/run.sh
@@ -23,11 +23,6 @@ ENCODED_DIR="$OUTDIR/encoded"
 mkdir -p "$OUTDIR" "$DMESG_DIR" "$ENCODED_DIR" >/dev/null 2>&1 || true
 : >"$RES_FILE"
 : >"$GST_LOG"
-
-SCRIPT_DIR="$(
-  cd "$(dirname "$0")" || exit 1
-  pwd
-)"
  
 INIT_ENV=""
 SEARCH="$SCRIPT_DIR"
@@ -39,8 +34,6 @@ while [ "$SEARCH" != "/" ]; do
   SEARCH=$(dirname "$SEARCH")
 done
  
-RES_FILE="$SCRIPT_DIR/${TESTNAME}.res"
- 
 if [ -z "${INIT_ENV:-}" ]; then
   echo "[ERROR] Could not find init_env (starting at $SCRIPT_DIR)" >&2
   echo "$TESTNAME SKIP" >"$RES_FILE" 2>/dev/null || true
@@ -245,7 +238,81 @@ while [ $# -gt 0 ]; do
       shift 2
       ;;
     -h|--help)
-      echo "$TESTNAME SKIP" >"$RES_FILE"
+      cat <<EOF
+Usage: $0 [OPTIONS]
+
+Video Encode/Decode Validation using GStreamer with V4L2 hardware accelerated codecs
+
+OPTIONS:
+  --mode <all|encode|decode>
+                        Test mode (default: all)
+                        - all: Run both encode and decode tests
+                        - encode: Run only encoding tests
+                        - decode: Run only decoding tests
+
+  --codecs <codec1,codec2,...>
+                        Comma-separated list of codecs to test
+                        (default: h264,h265,vp9)
+                        Supported: h264, h265, vp9
+
+  --resolutions <res1,res2,...>
+                        Comma-separated list of resolutions to test
+                        (default: 480p)
+                        Supported: 480p, 720p, 1080p, 4k
+
+  --duration <seconds>  Duration for encoding/decoding in seconds
+                        (default: 30)
+
+  --framerate <fps>     Framerate for video encoding
+                        (default: 30)
+
+  --stack <auto|upstream|downstream>
+                        Video stack to use
+                        (default: auto)
+
+  --gst-debug <level>   GStreamer debug level (1-9)
+                        (default: 2)
+
+  --clip-url <url>      URL to download test video files (VP9)
+                        (default: GitHub release URL)
+
+  --clip-path <path>    Local path to test video files
+                        (overrides --clip-url if files exist)
+
+  -h, --help            Display this help message
+
+ENVIRONMENT VARIABLES:
+  VIDEO_TEST_MODE       Same as --mode
+  VIDEO_CODECS          Same as --codecs
+  VIDEO_RESOLUTIONS     Same as --resolutions
+  VIDEO_DURATION        Same as --duration
+  VIDEO_FRAMERATE       Same as --framerate
+  VIDEO_STACK           Same as --stack
+  VIDEO_GST_DEBUG       Same as --gst-debug
+  VIDEO_CLIP_URL        Same as --clip-url
+  VIDEO_CLIP_PATH       Same as --clip-path
+  GST_DEBUG_LEVEL       Alternative to VIDEO_GST_DEBUG
+  RUNTIMESEC            Alternative to VIDEO_DURATION
+
+EXAMPLES:
+  # Run all tests with default settings
+  $0
+
+  # Run only encoding tests for H.264 at 720p
+  $0 --mode encode --codecs h264 --resolutions 720p
+
+  # Test multiple codecs and resolutions
+  $0 --codecs h264,h265 --resolutions 480p,720p
+
+  # Use upstream video stack
+  $0 --stack upstream
+
+SUPPORTED CODECS:
+  - h264: H.264/AVC encoding and decoding (v4l2h264enc, v4l2h264dec)
+  - h265: H.265/HEVC encoding and decoding (v4l2h265enc, v4l2h265dec)
+  - vp9:  VP9 decoding only (v4l2vp9dec) - uses pre-recorded WebM clip
+
+EOF
       exit 0
       ;;
     *)
@@ -630,9 +697,7 @@ fi
 log_info "=========================================="
 log_info "TEST SUMMARY"
 log_info "=========================================="
-# Calculate actual total for display (sum of pass/fail/skip)
-actual_total=$((pass_count + fail_count + skip_count))
-log_info "Total testcases: $actual_total"
+log_info "Total testcases: $total_tests"
 log_info "Passed: $pass_count"
 log_info "Failed: $fail_count"
 log_info "Skipped: $skip_count"
@@ -641,16 +706,16 @@ log_info "Skipped: $skip_count"
 if [ "$fail_count" -eq 0 ] && [ "$pass_count" -gt 0 ]; then
   result="PASS"
   if [ "$skip_count" -gt 0 ]; then
-    reason="No failures (passed: $pass_count, failed: $fail_count, skipped: $skip_count, total: $actual_total)"
+    reason="No failures (passed: $pass_count, failed: $fail_count, skipped: $skip_count, total: $total_tests)"
   else
-    reason="All tests passed ($pass_count/$actual_total)"
+    reason="All tests passed ($pass_count/$total_tests)"
   fi
 elif [ "$fail_count" -gt 0 ]; then
   result="FAIL"
-  reason="Some tests failed (passed: $pass_count, failed: $fail_count, skipped: $skip_count, total: $actual_total)"
+  reason="Some tests failed (passed: $pass_count, failed: $fail_count, skipped: $skip_count, total: $total_tests)"
 else
   result="SKIP"
-  reason="No tests passed (skipped: $skip_count, total: $actual_total)"
+  reason="No tests passed (skipped: $skip_count, total: $total_tests)"
 fi
 
 case "$result" in
diff --git a/Runner/utils/lib_gstreamer.sh b/Runner/utils/lib_gstreamer.sh
index 9d5b622a9..c5c5514a8 100755
--- a/Runner/utils/lib_gstreamer.sh
+++ b/Runner/utils/lib_gstreamer.sh
@@ -525,6 +525,35 @@ gstreamer_build_audio_record_pipeline() {
   return 0
 }
 
+
+# -------------------- Playback pipeline builder (backend-aware) --------------------
+# gstreamer_build_playback_pipeline <backend> <format> <file> <capsStrOrEmpty> <alsadev>
+gstreamer_build_playback_pipeline() {
+  backend="$1"
+  format="$2"
+  file="$3"
+  capsStr="$4"
+  alsadev="$5"
+
+  [ -n "$alsadev" ] || alsadev="default"
+
+  dec="$(gstreamer_pick_decode_chain "$format")"
+  sinkElem="$(gstreamer_pick_sink_element "$backend" "$alsadev")"
+  if [ -z "$sinkElem" ]; then
+    printf '%s\n' ""
+    return 0
+  fi
+
+  if [ -n "$capsStr" ]; then
+    printf '%s\n' "filesrc location=${file} ! ${dec} ! audioconvert ! audioresample ! ${capsStr} ! ${sinkElem}"
+    return 0
+  fi
+
+  printf '%s\n' "filesrc location=${file} ! ${dec} ! audioconvert ! audioresample ! ${sinkElem}"
+  return 0
+}
+
+
 # gstreamer_build_audio_playback_pipeline <format> <input_file>
 # Builds audio playback pipeline using pulsesink
 # Supports: wav, flac, ogg, mp3 formats

From 903ad59f562450a9b1414c6430b8f04a247ac059 Mon Sep 17 00:00:00 2001
From: Nitin Nakka <nitinn@qti.qualcomm.com>
Date: Tue, 10 Mar 2026 15:55:08 +0530
Subject: [PATCH 6/7] Added shellcheck disable=SC2317 for the cleanup function
 in all gst mm runner scripts, added help section and updated readme files for
 audio

Signed-off-by: Nitin Nakka <nitinn@qti.qualcomm.com>
---
 .../Audio/Audio_Record_Playback/README.md      | 18 +++++++++++++-----
 .../Audio/Audio_Record_Playback/run.sh         |  7 +++++--
 .../Display/Waylandsink_Playback/run.sh        |  1 +
 .../GSTreamer/Video/Video_Encode_Decode/run.sh |  1 +
 4 files changed, 20 insertions(+), 7 deletions(-)

diff --git a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/README.md b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/README.md
index 60ce90f61..5aa70a503 100644
--- a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/README.md
+++ b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/README.md
@@ -58,7 +58,7 @@ At a high level, the test:
 
 ## Test Cases
 
-By default, the test runs **8 test cases** with 10 second duration:
+By default, the test runs **10 test cases** with 10 second duration:
 
 ### ENCODE PHASE (4 tests)
 
@@ -70,7 +70,7 @@ By default, the test runs **8 test cases** with 10 second duration:
 3. **record_pulsesrc_wav** - Record WAV format using pulsesrc → audioconvert → wavenc
 4. **record_pulsesrc_flac** - Record FLAC format using pulsesrc → audioconvert → flacenc
 
-### DECODE PHASE (4 tests)
+### DECODE PHASE (6 tests)
 
 #### Playback Tests - audiotestsrc recordings
 5. **playback_wav** - Playback WAV file using filesrc → wavparse → audioconvert → pulsesink
@@ -80,7 +80,13 @@ By default, the test runs **8 test cases** with 10 second duration:
 7. **playback_pulsesrc_wav** - Playback WAV file using filesrc → wavparse → audioconvert → pulsesink
 8. **playback_pulsesrc_flac** - Playback FLAC file using filesrc → flacparse → flacdec → pulsesink
 
-**Total: 8 test cases** (4 encode + 4 decode)
+#### Playback Tests - External Test Files (OGG/MP3)
+9. **playback_sample_ogg** - Playback OGG file using filesrc → oggdemux → vorbisdec → pulsesink
+10. **playback_sample_mp3** - Playback MP3 file using filesrc → mpegaudioparse → mpg123audiodec → pulsesink
+
+**Total: 10 test cases** (4 encode + 6 decode)
+
+**Note:** OGG/MP3 playback tests require external test files (downloaded from URL or provided via `--clip-path`). If test files are not available, these tests will be skipped.
 
 ---
 
@@ -88,12 +94,14 @@ By default, the test runs **8 test cases** with 10 second duration:
 
 ### PASS
 - **Recording**: Output file is created and has size > 1000 bytes
-- **Playback**: Pipeline completes successfully (exit code 0, 124, or 143)
+- **Playback**: Pipeline completes successfully with exit code 0 (clean completion)
+  - **Note**: Timeout (exit code 124) or termination (exit code 143) are treated as **FAIL** for controlled-duration recordings
+  - This ensures playback completed successfully rather than being interrupted
 - **Overall**: At least one test passes and no tests fail
 
 ### FAIL
 - **Recording**: No output file created or file size too small
-- **Playback**: Pipeline fails or GStreamer errors detected
+- **Playback**: Pipeline exits with non-zero code (including timeout 124 or termination 143) or GStreamer errors detected
 - **Overall**: One or more tests fail
 
 ### SKIP
diff --git a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh
index 7b600e858..7c12eedf3 100755
--- a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh
+++ b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/run.sh
@@ -127,6 +127,7 @@ for param in AUDIO_DURATION AUDIO_GST_DEBUG GST_DEBUG_LEVEL; do
   fi
 done
 
+# shellcheck disable=SC2317
 cleanup() {
   # Best-effort: try to kill only children first; fall back to name-based kill
   if ! pkill -P "$$" -x gst-launch-1.0 >/dev/null 2>&1; then
@@ -718,7 +719,7 @@ run_playback_ogg_mp3_test() {
   fi
   
   # Check for successful completion
-  if [ "$gstRc" -eq 0 ] || [ "$gstRc" -eq 124 ] || [ "$gstRc" -eq 143 ]; then
+  if [ "$gstRc" -eq 0 ]; then
     log_pass "$testname: PASS"
     pass_count=$((pass_count + 1))
     return 0
@@ -769,9 +770,10 @@ provision_test_files() {
         log_warn "Test file download failed (offline or URL issue)"
       fi
     fi
+    
   fi
   
-  # Check what we have
+  # Check what we have AFTER all provisioning attempts
   have_ogg=0
   have_mp3=0
   
@@ -791,6 +793,7 @@ provision_test_files() {
     fi
   fi
   
+  # Only warn if BOTH files are missing after all attempts
   if [ "$have_ogg" -eq 0 ] && [ "$have_mp3" -eq 0 ]; then
     log_warn "No Test files (OGG/MP3) available for playback tests"
   fi
diff --git a/Runner/suites/Multimedia/GSTreamer/Display/Waylandsink_Playback/run.sh b/Runner/suites/Multimedia/GSTreamer/Display/Waylandsink_Playback/run.sh
index 6fdbb1327..435b2091a 100755
--- a/Runner/suites/Multimedia/GSTreamer/Display/Waylandsink_Playback/run.sh
+++ b/Runner/suites/Multimedia/GSTreamer/Display/Waylandsink_Playback/run.sh
@@ -92,6 +92,7 @@ height="${VIDEO_HEIGHT:-1080}"
 framerate="${VIDEO_FRAMERATE:-30}"
 gstDebugLevel="${VIDEO_GST_DEBUG:-${GST_DEBUG_LEVEL:-2}}"
 
+# shellcheck disable=SC2317
 cleanup() {
   # Best-effort: try to kill only children first; fall back to name-based kill
   if ! pkill -P "$$" -x gst-launch-1.0 >/dev/null 2>&1; then
diff --git a/Runner/suites/Multimedia/GSTreamer/Video/Video_Encode_Decode/run.sh b/Runner/suites/Multimedia/GSTreamer/Video/Video_Encode_Decode/run.sh
index 47de483b9..e0bb4bc83 100755
--- a/Runner/suites/Multimedia/GSTreamer/Video/Video_Encode_Decode/run.sh
+++ b/Runner/suites/Multimedia/GSTreamer/Video/Video_Encode_Decode/run.sh
@@ -103,6 +103,7 @@ for param in VIDEO_DURATION RUNTIMESEC VIDEO_FRAMERATE VIDEO_GST_DEBUG GST_DEBUG
   fi
 done
 
+# shellcheck disable=SC2317
 cleanup() {
   # Best-effort: try to kill only children first; fall back to name-based kill
   if ! pkill -P "$$" -x gst-launch-1.0 >/dev/null 2>&1; then

From 66487b241b5f2dfc8bc4bd51ed06e418fcceaa36 Mon Sep 17 00:00:00 2001
From: Nitin Nakka <nitinn@qti.qualcomm.com>
Date: Sun, 15 Mar 2026 12:18:07 +0530
Subject: [PATCH 7/7] Added 10 tests related changes in the readme and the yaml
 files

Signed-off-by: Nitin Nakka <nitinn@qti.qualcomm.com>
---
 .../Audio_Record_Playback.yaml                 |  4 ++--
 .../Audio/Audio_Record_Playback/README.md      | 18 ++++++++++--------
 2 files changed, 12 insertions(+), 10 deletions(-)

diff --git a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/Audio_Record_Playback.yaml b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/Audio_Record_Playback.yaml
index 4aadc832f..cf1fbd372 100644
--- a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/Audio_Record_Playback.yaml
+++ b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/Audio_Record_Playback.yaml
@@ -3,8 +3,8 @@ metadata:
   format: "Lava-Test Test Definition 1.0"
   description: >
     Audio record/playback validation using GStreamer (gst-launch-1.0).
-    Tests 8 scenarios: 4 encode (2 audiotestsrc + 2 pulsesrc) + 4 decode (playback).
-    Supports WAV and FLAC formats with both synthetic audio (audiotestsrc) and hardware capture (pulsesrc).
+    Tests 10 scenarios: 4 encode (2 audiotestsrc + 2 pulsesrc) + 6 decode (4 playback + 2 OGG/MP3).
+    Supports WAV, FLAC, OGG, and MP3 formats with both synthetic audio (audiotestsrc) and hardware capture (pulsesrc).
     Duration controlled via AUDIO_DURATION (default 10s): audiotestsrc uses num-buffers, pulsesrc uses timeout.
   os:
     - linux
diff --git a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/README.md b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/README.md
index 5aa70a503..c1247e42c 100644
--- a/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/README.md
+++ b/Runner/suites/Multimedia/GSTreamer/Audio/Audio_Record_Playback/README.md
@@ -49,9 +49,10 @@ At a high level, the test:
    - Duration control:
      - audiotestsrc: Buffer count calculated dynamically: `(44100 * duration) / 1024`
      - pulsesrc: Uses timeout command to stop recording after specified duration
-5. **Playback phase (DECODE - 4 tests)**:
+5. **Playback phase (DECODE - 6 tests)**:
    - Reads the previously recorded files (both audiotestsrc and pulsesrc recordings)
    - Decodes and plays back using pulsesink
+   - Plays back external test files (OGG/MP3 formats)
 6. Collects test results and emits PASS/FAIL/SKIP
 
 ---
@@ -219,7 +220,7 @@ Help:
 
 ## Examples
 
-### 1) Run all tests (default - 8 tests: 4 encode + 4 decode for WAV and FLAC with 10 second duration)
+### 1) Run all tests (default - 10 tests: 4 encode + 6 decode for WAV, FLAC, OGG, and MP3 with 10 second duration)
 
 ```bash
 ./run.sh
@@ -231,19 +232,19 @@ Help:
 ./run.sh --mode record
 ```
 
-### 3) Run only playback tests (4 decode tests - requires recorded files from previous run)
+### 3) Run only playback tests (6 decode tests - requires recorded files from previous run, plus OGG/MP3 test files)
 
 ```bash
 ./run.sh --mode playback
 ```
 
-### 4) Test only WAV format (4 tests: 2 encode + 2 decode)
+### 4) Test only WAV format (6 tests total: 2 encode + 2 decode + 2 OGG/MP3 playback)
 
 ```bash
 ./run.sh --formats wav
 ```
 
-### 5) Test only FLAC format (4 tests: 2 encode + 2 decode)
+### 5) Test only FLAC format (6 tests total: 2 encode + 2 decode + 2 OGG/MP3 playback)
 
 ```bash
 ./run.sh --formats flac
@@ -255,7 +256,7 @@ Help:
 ./run.sh --duration 30
 ```
 
-### 7) Quick test - WAV only with 3 second duration (4 tests)
+### 7) Quick test - WAV only with 3 second duration (6 tests total)
 
 ```bash
 ./run.sh --formats wav --duration 3
@@ -546,11 +547,12 @@ The test supports these environment variables (can be set in LAVA job definition
 
 ### Test Counting
 
-- **Total tests**: 8 (when running with default wav,flac formats in all mode)
+- **Total tests**: 10 (when running with default wav,flac formats in all mode)
   - 4 encode tests (2 audiotestsrc + 2 pulsesrc)
-  - 4 decode tests (2 audiotestsrc playback + 2 pulsesrc playback)
+  - 6 decode tests (2 audiotestsrc playback + 2 pulsesrc playback + 2 OGG/MP3 playback)
 - **Pass/Fail/Skip are mutually exclusive**: Each test increments exactly one counter
 - **Plugin unavailability**: pulsesrc tests will skip if pulsesrc plugin is not available, but audiotestsrc tests will still run
+- **Test file availability**: OGG/MP3 playback tests will skip if test files are not available
 
 ---