diff --git a/src/audio/alsa.c b/src/audio/alsa.c
index 89a593c..d95bb3d 100644
--- a/src/audio/alsa.c
+++ b/src/audio/alsa.c
@@ -29,7 +29,8 @@
 
 static snd_pcm_t *handle;
 static OpusMSDecoder* decoder;
-static short pcmBuffer[FRAME_SIZE * AUDIO_CONFIGURATION_MAX_CHANNEL_COUNT];
+static short* pcmBuffer;
+static int samplesPerFrame;
 
 static int alsa_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig, void* context, int arFlags) {
   int rc;
@@ -47,11 +48,16 @@ static int alsa_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGUR
     alsaMapping[5] = opusConfig->mapping[3];
   }
 
+  samplesPerFrame = opusConfig->samplesPerFrame;
+  pcmBuffer = malloc(sizeof(short) * opusConfig->channelCount * samplesPerFrame);
+  if (pcmBuffer == NULL)
+    return -1;
+
   decoder = opus_multistream_decoder_create(opusConfig->sampleRate, opusConfig->channelCount, opusConfig->streams, opusConfig->coupledStreams, alsaMapping, &rc);
 
   snd_pcm_hw_params_t *hw_params;
   snd_pcm_sw_params_t *sw_params;
-  snd_pcm_uframes_t period_size = FRAME_SIZE * FRAME_BUFFER;
+  snd_pcm_uframes_t period_size = samplesPerFrame * FRAME_BUFFER;
   snd_pcm_uframes_t buffer_size = 2 * period_size;
   unsigned int sampleRate = opusConfig->sampleRate;
 
@@ -88,17 +94,25 @@ static int alsa_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGUR
 }
 
 static void alsa_renderer_cleanup() {
-  if (decoder != NULL)
+  if (decoder != NULL) {
     opus_multistream_decoder_destroy(decoder);
+    decoder = NULL;
+  }
 
   if (handle != NULL) {
     snd_pcm_drain(handle);
     snd_pcm_close(handle);
+    handle = NULL;
+  }
+
+  if (pcmBuffer != NULL) {
+    free(pcmBuffer);
+    pcmBuffer = NULL;
   }
 }
 
 static void alsa_renderer_decode_and_play_sample(char* data, int length) {
-  int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
+  int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, samplesPerFrame, 0);
   if (decodeLen > 0) {
     int rc = snd_pcm_writei(handle, pcmBuffer, decodeLen);
     if (rc == -EPIPE)
@@ -117,5 +131,5 @@ AUDIO_RENDERER_CALLBACKS audio_callbacks_alsa = {
   .init = alsa_renderer_init,
   .cleanup = alsa_renderer_cleanup,
   .decodeAndPlaySample = alsa_renderer_decode_and_play_sample,
-  .capabilities = CAPABILITY_DIRECT_SUBMIT,
+  .capabilities = CAPABILITY_DIRECT_SUBMIT | CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION,
 };
diff --git a/src/audio/audio.h b/src/audio/audio.h
index 470c307..fe1fbfa 100644
--- a/src/audio/audio.h
+++ b/src/audio/audio.h
@@ -21,7 +21,6 @@
 
 #include <Limelight.h>
 
-#define FRAME_SIZE 240
 #define FRAME_BUFFER 12
 
 #ifdef HAVE_ALSA
diff --git a/src/audio/omx.c b/src/audio/omx.c
index 21cec30..65deaad 100644
--- a/src/audio/omx.c
+++ b/src/audio/omx.c
@@ -29,8 +29,9 @@ static OpusMSDecoder* decoder;
 ILCLIENT_T* handle;
 COMPONENT_T* component;
 static OMX_BUFFERHEADERTYPE *buf;
-static short pcmBuffer[FRAME_SIZE * AUDIO_CONFIGURATION_MAX_CHANNEL_COUNT];
+static short* pcmBuffer;
 static int channelCount;
+static int samplesPerFrame;
 
 static int omx_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig, void* context, int arFlags) {
   int rc, error;
@@ -39,6 +40,11 @@ static int omx_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURA
   char* componentName = "audio_render";
 
   channelCount = opusConfig->channelCount;
+  samplesPerFrame = opusConfig->samplesPerFrame;
+  pcmBuffer = malloc(sizeof(short) * channelCount * samplesPerFrame);
+  if (pcmBuffer == NULL)
+    return -1;
+
   /* The supplied mapping array has order: FL-FR-C-LFE-RL-RR-SL-SR
    * OMX expects the order: FL-FR-LFE-C-RL-RR-SL-SR
    * We need copy the mapping locally and swap the channels around.
@@ -155,8 +161,10 @@ static int omx_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURA
 }
 
 static void omx_renderer_cleanup() {
-  if (decoder != NULL)
+  if (decoder != NULL) {
     opus_multistream_decoder_destroy(decoder);
+    decoder = NULL;
+  }
   if (handle != NULL) {
     if((buf = ilclient_get_input_buffer(component, 100, 1)) == NULL){
       fprintf(stderr, "Can't get audio buffer\n");
@@ -174,11 +182,16 @@ static void omx_renderer_cleanup() {
     ilclient_disable_port_buffers(component, 100, NULL, NULL, NULL);
     ilclient_change_component_state(component, OMX_StateIdle);
     ilclient_change_component_state(component, OMX_StateLoaded);
+    handle = NULL;
+  }
+  if (pcmBuffer != NULL) {
+    free(pcmBuffer);
+    pcmBuffer = NULL;
   }
 }
 
 static void omx_renderer_decode_and_play_sample(char* data, int length) {
-  int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
+  int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, samplesPerFrame, 0);
   if (decodeLen > 0) {
     buf = ilclient_get_input_buffer(component, 100, 1);
     buf->nOffset = 0;
@@ -199,5 +212,5 @@ AUDIO_RENDERER_CALLBACKS audio_callbacks_omx = {
   .init = omx_renderer_init,
   .cleanup = omx_renderer_cleanup,
   .decodeAndPlaySample = omx_renderer_decode_and_play_sample,
-  .capabilities = CAPABILITY_DIRECT_SUBMIT,
+  .capabilities = CAPABILITY_DIRECT_SUBMIT | CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION,
 };
diff --git a/src/audio/pulse.c b/src/audio/pulse.c
index c10c57b..596cde2 100644
--- a/src/audio/pulse.c
+++ b/src/audio/pulse.c
@@ -29,7 +29,8 @@
 
 static OpusMSDecoder* decoder;
 static pa_simple *dev = NULL;
-static short pcmBuffer[FRAME_SIZE * AUDIO_CONFIGURATION_MAX_CHANNEL_COUNT];
+static short* pcmBuffer;
+static int samplesPerFrame;
 static int channelCount;
 
 bool audio_pulse_init(char* audio_device) {
@@ -53,6 +54,10 @@ static int pulse_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGU
   unsigned char alsaMapping[AUDIO_CONFIGURATION_MAX_CHANNEL_COUNT];
 
   channelCount = opusConfig->channelCount;
+  samplesPerFrame = opusConfig->samplesPerFrame;
+  pcmBuffer = malloc(sizeof(short) * channelCount * samplesPerFrame);
+  if (pcmBuffer == NULL)
+    return -1;
 
   /* The supplied mapping array has order: FL-FR-C-LFE-RL-RR-SL-SR
    * ALSA expects the order: FL-FR-RL-RR-C-LFE-SL-SR
@@ -86,7 +91,7 @@ static int pulse_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGU
 }
 
 static void pulse_renderer_decode_and_play_sample(char* data, int length) {
-  int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
+  int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, samplesPerFrame, 0);
   if (decodeLen > 0) {
     int error;
     int rc = pa_simple_write(dev, pcmBuffer, decodeLen * sizeof(short) * channelCount, &error);
@@ -99,12 +104,23 @@ static void pulse_renderer_decode_and_play_sample(char* data, int length) {
 }
 
 static void pulse_renderer_cleanup() {
-  pa_simple_free(dev);
+  if (decoder != NULL) {
+    opus_multistream_decoder_destroy(decoder);
+    decoder = NULL;
+  }
+  if (dev != NULL) {
+    pa_simple_free(dev);
+    dev = NULL;
+  }
+  if (pcmBuffer != NULL) {
+    free(pcmBuffer);
+    pcmBuffer = NULL;
+  }
 }
 
 AUDIO_RENDERER_CALLBACKS audio_callbacks_pulse = {
   .init = pulse_renderer_init,
   .cleanup = pulse_renderer_cleanup,
   .decodeAndPlaySample = pulse_renderer_decode_and_play_sample,
-  .capabilities = CAPABILITY_DIRECT_SUBMIT,
+  .capabilities = CAPABILITY_DIRECT_SUBMIT | CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION,
 };
diff --git a/src/audio/sdl.c b/src/audio/sdl.c
index 34633ea..96a30cd 100644
--- a/src/audio/sdl.c
+++ b/src/audio/sdl.c
@@ -26,7 +26,8 @@
 #include <opus_multistream.h>
 
 static OpusMSDecoder* decoder;
-static short pcmBuffer[FRAME_SIZE * AUDIO_CONFIGURATION_MAX_CHANNEL_COUNT];
+static short* pcmBuffer;
+static int samplesPerFrame;
 static SDL_AudioDeviceID dev;
 static int channelCount;
 
@@ -35,6 +36,10 @@ static int sdl_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURA
   decoder = opus_multistream_decoder_create(opusConfig->sampleRate, opusConfig->channelCount, opusConfig->streams, opusConfig->coupledStreams, opusConfig->mapping, &rc);
 
   channelCount = opusConfig->channelCount;
+  samplesPerFrame = opusConfig->samplesPerFrame;
+  pcmBuffer = malloc(sizeof(short) * channelCount * samplesPerFrame);
+  if (pcmBuffer == NULL)
+    return -1;
 
   SDL_InitSubSystem(SDL_INIT_AUDIO);
 
@@ -45,13 +50,11 @@ static int sdl_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURA
   want.channels = opusConfig->channelCount;
   want.samples = 4096;
 
-  dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, SDL_AUDIO_ALLOW_FORMAT_CHANGE);
+  dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0);
   if (dev == 0) {
     printf("Failed to open audio: %s\n", SDL_GetError());
     return -1;
   } else {
-    if (have.format != want.format)  // we let this one thing change.
-      printf("We didn't get requested audio format.\n");
     SDL_PauseAudioDevice(dev, 0);  // start audio playing.
   }
 
@@ -59,14 +62,24 @@ static int sdl_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURA
 }
 
 static void sdl_renderer_cleanup() {
-  if (decoder != NULL)
+  if (decoder != NULL) {
     opus_multistream_decoder_destroy(decoder);
+    decoder = NULL;
+  }
 
-  SDL_CloseAudioDevice(dev);
+  if (pcmBuffer != NULL) {
+    free(pcmBuffer);
+    pcmBuffer = NULL;
+  }
+
+  if (dev != 0) {
+    SDL_CloseAudioDevice(dev);
+    dev = 0;
+  }
 }
 
 static void sdl_renderer_decode_and_play_sample(char* data, int length) {
-  int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
+  int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, samplesPerFrame, 0);
   if (decodeLen > 0) {
     SDL_QueueAudio(dev, pcmBuffer, decodeLen * channelCount * sizeof(short));
   } else {
@@ -78,5 +91,5 @@ AUDIO_RENDERER_CALLBACKS audio_callbacks_sdl = {
   .init = sdl_renderer_init,
   .cleanup = sdl_renderer_cleanup,
   .decodeAndPlaySample = sdl_renderer_decode_and_play_sample,
-  .capabilities = CAPABILITY_DIRECT_SUBMIT,
+  .capabilities = CAPABILITY_DIRECT_SUBMIT | CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION,
 };