diff --git a/channels/audin/client/alsa/audin_alsa.c b/channels/audin/client/alsa/audin_alsa.c
index e81ec63a5..bacdc3230 100644
--- a/channels/audin/client/alsa/audin_alsa.c
+++ b/channels/audin/client/alsa/audin_alsa.c
@@ -75,12 +75,6 @@ static snd_pcm_format_t audin_alsa_format(UINT32 wFormatTag, UINT32 bitPerChanne
 					return SND_PCM_FORMAT_UNKNOWN;
 			}
 
-		case WAVE_FORMAT_ALAW:
-			return SND_PCM_FORMAT_A_LAW;
-
-		case WAVE_FORMAT_MULAW:
-			return SND_PCM_FORMAT_MU_LAW;
-
 		default:
 			return SND_PCM_FORMAT_UNKNOWN;
 	}
diff --git a/channels/audin/client/oss/audin_oss.c b/channels/audin/client/oss/audin_oss.c
index 7ffc0cdbc..2da93a148 100644
--- a/channels/audin/client/oss/audin_oss.c
+++ b/channels/audin/client/oss/audin_oss.c
@@ -88,12 +88,6 @@ static UINT32 audin_oss_get_format(const AUDIO_FORMAT* format)
 			}
 
 			break;
-
-		case WAVE_FORMAT_ALAW:
-			return AFMT_A_LAW;
-
-		case WAVE_FORMAT_MULAW:
-			return AFMT_MU_LAW;
 	}
 
 	return 0;
diff --git a/channels/audin/client/pulse/audin_pulse.c b/channels/audin/client/pulse/audin_pulse.c
index 6855d757f..8977c420b 100644
--- a/channels/audin/client/pulse/audin_pulse.c
+++ b/channels/audin/client/pulse/audin_pulse.c
@@ -280,14 +280,6 @@ static UINT audin_pulse_set_format(IAudinDevice* device, const AUDIO_FORMAT* for
 
 			break;
 
-		case WAVE_FORMAT_ALAW: /* A-LAW */
-			sample_spec.format = PA_SAMPLE_ALAW;
-			break;
-
-		case WAVE_FORMAT_MULAW: /* U-LAW */
-			sample_spec.format = PA_SAMPLE_ULAW;
-			break;
-
 		default:
 			return ERROR_INTERNAL_ERROR;
 	}
@@ -300,6 +292,8 @@ static UINT audin_pulse_set_format(IAudinDevice* device, const AUDIO_FORMAT* for
 static void audin_pulse_stream_state_callback(pa_stream* stream, void* userdata)
 {
 	AudinPulseDevice* pulse = (AudinPulseDevice*)userdata;
+	WINPR_ASSERT(pulse);
+
 	pa_stream_state_t state = pa_stream_get_state(stream);
 
 	WLog_Print(pulse->log, WLOG_DEBUG, "stream state %s", pulse_stream_state_string(state));
@@ -416,7 +410,7 @@ static UINT audin_pulse_open(IAudinDevice* device, AudinReceive receive, void* u
 		return pa_context_errno(pulse->context);
 	}
 
-	for (;;)
+	while (pulse->stream)
 	{
 		state = pa_stream_get_state(pulse->stream);
 
diff --git a/channels/rdpsnd/client/oss/rdpsnd_oss.c b/channels/rdpsnd/client/oss/rdpsnd_oss.c
index a23849b72..91668812b 100644
--- a/channels/rdpsnd/client/oss/rdpsnd_oss.c
+++ b/channels/rdpsnd/client/oss/rdpsnd_oss.c
@@ -86,12 +86,6 @@ static int rdpsnd_oss_get_format(const AUDIO_FORMAT* format)
 			}
 
 			break;
-
-		case WAVE_FORMAT_ALAW:
-			return AFMT_A_LAW;
-
-		case WAVE_FORMAT_MULAW:
-			return AFMT_MU_LAW;
 	}
 
 	return 0;
diff --git a/channels/rdpsnd/client/pulse/rdpsnd_pulse.c b/channels/rdpsnd/client/pulse/rdpsnd_pulse.c
index 311bbf906..957948dff 100644
--- a/channels/rdpsnd/client/pulse/rdpsnd_pulse.c
+++ b/channels/rdpsnd/client/pulse/rdpsnd_pulse.c
@@ -327,14 +327,6 @@ static BOOL rdpsnd_pulse_set_format_spec(rdpsndPulsePlugin* pulse, const AUDIO_F
 
 			break;
 
-		case WAVE_FORMAT_ALAW:
-			sample_spec.format = PA_SAMPLE_ALAW;
-			break;
-
-		case WAVE_FORMAT_MULAW:
-			sample_spec.format = PA_SAMPLE_ULAW;
-			break;
-
 		default:
 			return FALSE;
 	}
@@ -422,7 +414,7 @@ static BOOL rdpsnd_pulse_open_stream(rdpsndDevicePlugin* device)
 		return FALSE;
 	}
 
-	for (;;)
+	while (pulse->stream)
 	{
 		state = pa_stream_get_state(pulse->stream);
 
diff --git a/channels/tsmf/client/pulse/tsmf_pulse.c b/channels/tsmf/client/pulse/tsmf_pulse.c
index 34a9fd55a..d580a243a 100644
--- a/channels/tsmf/client/pulse/tsmf_pulse.c
+++ b/channels/tsmf/client/pulse/tsmf_pulse.c
@@ -176,6 +176,8 @@ static void tsmf_pulse_wait_for_operation(TSMFPulseAudioDevice* pulse, pa_operat
 static void tsmf_pulse_stream_state_callback(pa_stream* stream, void* userdata)
 {
 	TSMFPulseAudioDevice* pulse = (TSMFPulseAudioDevice*)userdata;
+	WINPR_ASSERT(pulse);
+
 	pa_stream_state_t state = pa_stream_get_state(stream);
 
 	switch (state)
diff --git a/libfreerdp/codec/dsp.c b/libfreerdp/codec/dsp.c
index 913a886a5..26e44f4cf 100644
--- a/libfreerdp/codec/dsp.c
+++ b/libfreerdp/codec/dsp.c
@@ -1183,6 +1183,12 @@ fail:
 
 void freerdp_dsp_context_free(FREERDP_DSP_CONTEXT* context)
 {
+#if defined(WITH_FDK_AAC)
+	FREERDP_DSP_COMMON_CONTEXT* ctx = (FREERDP_DSP_COMMON_CONTEXT*)context;
+	WINPR_ASSERT(ctx);
+	fdk_aac_dsp_uninit(ctx);
+#endif
+
 #if defined(WITH_DSP_FFMPEG)
 	freerdp_dsp_ffmpeg_context_free(context);
 #else
diff --git a/libfreerdp/codec/dsp_ffmpeg.c b/libfreerdp/codec/dsp_ffmpeg.c
index 4f34d9209..52d33bfc2 100644
--- a/libfreerdp/codec/dsp_ffmpeg.c
+++ b/libfreerdp/codec/dsp_ffmpeg.c
@@ -70,19 +70,17 @@ static BOOL ffmpeg_codec_is_filtered(enum AVCodecID id, BOOL encoder)
 		case AV_CODEC_ID_MP3:
 		case AV_CODEC_ID_ADPCM_MS:
 		case AV_CODEC_ID_G723_1:
+		case AV_CODEC_ID_GSM_MS:
+		case AV_CODEC_ID_PCM_ALAW:
+		case AV_CODEC_ID_PCM_MULAW:
 			return TRUE;
 #endif
 
 		case AV_CODEC_ID_NONE:
 			return TRUE;
 
-		case AV_CODEC_ID_GSM_MS:
 		case AV_CODEC_ID_AAC:
 		case AV_CODEC_ID_AAC_LATM:
-#if !defined(WITH_DSP_EXPERIMENTAL)
-			if (encoder)
-				return TRUE;
-#endif
 			return FALSE;
 
 		default: