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audiortppay: more refactoring

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Unify the sample/frame buffer handling code by making the functions plugable.
This commit is contained in:
Wim Taymans 2009-09-02 12:24:22 +02:00
parent fb5037f727
commit 299ab7be0e
1 changed files with 49 additions and 146 deletions
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195
gst-libs/gst/rtp/gstbasertpaudiopayload.c
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@ -70,18 +70,19 @@
GST_DEBUG_CATEGORY_STATIC (basertpaudiopayload_debug); GST_DEBUG_CATEGORY_STATIC (basertpaudiopayload_debug);
#define GST_CAT_DEFAULT (basertpaudiopayload_debug) #define GST_CAT_DEFAULT (basertpaudiopayload_debug)
typedef enum /* function to calculate the min/max length and alignment of a packet */
{ typedef gboolean (*GetLengthsFunc) (GstBaseRTPPayload * basepayload,
AUDIO_CODEC_TYPE_NONE, guint * min_payload_len, guint * max_payload_len, guint * align);
AUDIO_CODEC_TYPE_FRAME_BASED, /* function to convert bytes to a duration */
AUDIO_CODEC_TYPE_SAMPLE_BASED typedef GstClockTime (*GetDurationFunc) (GstBaseRTPAudioPayload * payload,
} AudioCodecType; guint bytes);
struct _GstBaseRTPAudioPayloadPrivate struct _GstBaseRTPAudioPayloadPrivate
{ {
AudioCodecType type; GetLengthsFunc get_lengths;
GetDurationFunc get_duration;
GstAdapter *adapter; GstAdapter *adapter;
guint64 min_ptime;
guint fragment_size; guint fragment_size;
}; };
@ -92,22 +93,30 @@ struct _GstBaseRTPAudioPayloadPrivate
static void gst_base_rtp_audio_payload_finalize (GObject * object); static void gst_base_rtp_audio_payload_finalize (GObject * object);
/* length functions */
static gboolean gst_base_rtp_audio_payload_get_frame_lengths (GstBaseRTPPayload
* basepayload, guint * min_payload_len, guint * max_payload_len,
guint * align);
static gboolean gst_base_rtp_audio_payload_get_sample_lengths (GstBaseRTPPayload
* basepayload, guint * min_payload_len, guint * max_payload_len,
guint * align);
/* duration functions */
static GstClockTime
gst_base_rtp_audio_payload_get_frame_duration (GstBaseRTPAudioPayload * payload,
guint bytes);
static GstClockTime
gst_base_rtp_audio_payload_get_sample_duration (GstBaseRTPAudioPayload *
payload, guint bytes);
static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload
* payload, GstBuffer * buffer); * payload, GstBuffer * buffer);
static GstFlowReturn static GstStateChangeReturn gst_base_rtp_payload_audio_change_state (GstElement
gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload * * element, GstStateChange transition);
basepayload, GstBuffer * buffer);
static GstFlowReturn static gboolean gst_base_rtp_payload_audio_handle_event (GstPad * pad,
gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload * GstEvent * event);
basepayload, GstBuffer * buffer);
static GstStateChangeReturn
gst_base_rtp_payload_audio_change_state (GstElement * element,
GstStateChange transition);
static gboolean
gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event);
GST_BOILERPLATE (GstBaseRTPAudioPayload, gst_base_rtp_audio_payload, GST_BOILERPLATE (GstBaseRTPAudioPayload, gst_base_rtp_audio_payload,
GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD); GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD);
@ -152,10 +161,6 @@ gst_base_rtp_audio_payload_init (GstBaseRTPAudioPayload * basertpaudiopayload,
basertpaudiopayload->priv = basertpaudiopayload->priv =
GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE (basertpaudiopayload); GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE (basertpaudiopayload);
basertpaudiopayload->base_ts = 0;
basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_NONE;
/* these need to be set by child object if frame based */ /* these need to be set by child object if frame based */
basertpaudiopayload->frame_size = 0; basertpaudiopayload->frame_size = 0;
basertpaudiopayload->frame_duration = 0; basertpaudiopayload->frame_duration = 0;
@ -184,17 +189,19 @@ gst_base_rtp_audio_payload_finalize (GObject * object)
* *
* Tells #GstBaseRTPAudioPayload that the child element is for a frame based * Tells #GstBaseRTPAudioPayload that the child element is for a frame based
* audio codec * audio codec
*
*/ */
void void
gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload * gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *
basertpaudiopayload) basertpaudiopayload)
{ {
g_return_if_fail (basertpaudiopayload != NULL); g_return_if_fail (basertpaudiopayload != NULL);
g_return_if_fail (basertpaudiopayload->priv->get_lengths == NULL);
g_return_if_fail (basertpaudiopayload->priv->get_duration == NULL);
g_return_if_fail (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_NONE); basertpaudiopayload->priv->get_lengths =
gst_base_rtp_audio_payload_get_frame_lengths;
basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_FRAME_BASED; basertpaudiopayload->priv->get_duration =
gst_base_rtp_audio_payload_get_frame_duration;
} }
/** /**
@ -203,17 +210,19 @@ gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *
* *
* Tells #GstBaseRTPAudioPayload that the child element is for a sample based * Tells #GstBaseRTPAudioPayload that the child element is for a sample based
* audio codec * audio codec
*
*/ */
void void
gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload * gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload *
basertpaudiopayload) basertpaudiopayload)
{ {
g_return_if_fail (basertpaudiopayload != NULL); g_return_if_fail (basertpaudiopayload != NULL);
g_return_if_fail (basertpaudiopayload->priv->get_lengths == NULL);
g_return_if_fail (basertpaudiopayload->priv->get_duration == NULL);
g_return_if_fail (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_NONE); basertpaudiopayload->priv->get_lengths =
gst_base_rtp_audio_payload_get_sample_lengths;
basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_SAMPLE_BASED; basertpaudiopayload->priv->get_duration =
gst_base_rtp_audio_payload_get_sample_duration;
} }
/** /**
@ -283,30 +292,6 @@ gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload
gst_adapter_clear (basertpaudiopayload->priv->adapter); gst_adapter_clear (basertpaudiopayload->priv->adapter);
} }
static GstFlowReturn
gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstFlowReturn ret;
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
ret = GST_FLOW_ERROR;
if (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_FRAME_BASED) {
ret = gst_base_rtp_audio_payload_handle_frame_based_buffer (basepayload,
buffer);
} else if (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_SAMPLE_BASED) {
ret = gst_base_rtp_audio_payload_handle_sample_based_buffer (basepayload,
buffer);
} else {
GST_DEBUG_OBJECT (basertpaudiopayload, "Audio codec type not set");
}
return ret;
}
/** /**
* gst_base_rtp_audio_payload_flush: * gst_base_rtp_audio_payload_flush:
* @baseaudiopayload: a #GstBaseRTPPayload * @baseaudiopayload: a #GstBaseRTPPayload
@ -352,6 +337,7 @@ gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload,
#define ALIGN_DOWN(val,len) ((val) - ((val) % (len))) #define ALIGN_DOWN(val,len) ((val) - ((val) % (len)))
/* this assumes all frames have a constant duration and a constant size */
static gboolean static gboolean
gst_base_rtp_audio_payload_get_frame_lengths (GstBaseRTPPayload * gst_base_rtp_audio_payload_get_frame_lengths (GstBaseRTPPayload *
basepayload, guint * min_payload_len, guint * max_payload_len, basepayload, guint * min_payload_len, guint * max_payload_len,
@ -412,91 +398,6 @@ gst_base_rtp_audio_payload_get_frame_duration (GstBaseRTPAudioPayload *
payload->frame_size); payload->frame_size);
} }
/* this assumes all frames have a constant duration and a constant size */
static GstFlowReturn
gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
basepayload, GstBuffer * buffer)
{
GstBaseRTPAudioPayload *payload;
guint align;
guint payload_len;
GstFlowReturn ret;
guint available;
guint min_payload_len;
guint max_payload_len;
guint size;
ret = GST_FLOW_OK;
payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
if (!gst_base_rtp_audio_payload_get_frame_lengths (basepayload,
&min_payload_len, &max_payload_len, &align))
goto config_error;
GST_DEBUG_OBJECT (payload,
"Calculated min_payload_len %u and max_payload_len %u",
min_payload_len, max_payload_len);
size = GST_BUFFER_SIZE (buffer);
/* shortcut, we don't need to use the adapter when the packet can be pushed
* through directly. */
available = gst_adapter_available (payload->priv->adapter);
if (available == 0 && (size >= min_payload_len && size <= max_payload_len)) {
/* If buffer fits on an RTP packet, let's just push it through
* this will check against max_ptime and max_mtu */
GST_DEBUG_OBJECT (payload, "Fast packet push");
ret = gst_base_rtp_audio_payload_push (payload,
GST_BUFFER_DATA (buffer), size, GST_BUFFER_TIMESTAMP (buffer));
gst_buffer_unref (buffer);
} else {
/* push the buffer in the adapter */
gst_adapter_push (payload->priv->adapter, buffer);
available += size;
/* as long as we have full frames */
while (available >= min_payload_len) {
guint64 distance;
GstClockTime timestamp;
/* We send as much as we can */
payload_len = ALIGN_DOWN (available, align);
payload_len = MIN (max_payload_len, payload_len);
/* calculate the timestamp */
timestamp =
gst_adapter_prev_timestamp (payload->priv->adapter, &distance);
GST_LOG_OBJECT (payload,
"last timestamp %" GST_TIME_FORMAT ", distance %" G_GUINT64_FORMAT,
GST_TIME_ARGS (timestamp), distance);
if (GST_CLOCK_TIME_IS_VALID (timestamp) && distance > 0) {
/* convert the number of bytes since the last timestamp to time and add to
* the last seen timestamp */
timestamp +=
gst_base_rtp_audio_payload_get_frame_duration (payload, distance);
}
/* and flush out the bytes from the adapter */
ret = gst_base_rtp_audio_payload_flush (payload, payload_len, timestamp);
available -= payload_len;
}
}
return ret;
/* ERRORS */
config_error:
{
GST_DEBUG_OBJECT (payload, "Required options not set");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
}
static gboolean static gboolean
gst_base_rtp_audio_payload_get_sample_lengths (GstBaseRTPPayload * gst_base_rtp_audio_payload_get_sample_lengths (GstBaseRTPPayload *
basepayload, guint * min_payload_len, guint * max_payload_len, basepayload, guint * min_payload_len, guint * max_payload_len,
@ -551,7 +452,7 @@ gst_base_rtp_audio_payload_get_sample_duration (GstBaseRTPAudioPayload *
} }
static GstFlowReturn static GstFlowReturn
gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload * gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
basepayload, GstBuffer * buffer) basepayload, GstBuffer * buffer)
{ {
GstBaseRTPAudioPayload *payload; GstBaseRTPAudioPayload *payload;
@ -567,8 +468,11 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload); payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
if (!gst_base_rtp_audio_payload_get_sample_lengths (basepayload, if (payload->priv->get_lengths == NULL || payload->priv->get_duration == NULL)
&min_payload_len, &max_payload_len, &align)) goto config_error;
if (!payload->priv->get_lengths (basepayload, &min_payload_len,
&max_payload_len, &align))
goto config_error; goto config_error;
GST_DEBUG_OBJECT (payload, GST_DEBUG_OBJECT (payload,
@ -617,8 +521,7 @@ gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
if (GST_CLOCK_TIME_IS_VALID (timestamp) && distance > 0) { if (GST_CLOCK_TIME_IS_VALID (timestamp) && distance > 0) {
/* convert the number of bytes since the last timestamp to time and add to /* convert the number of bytes since the last timestamp to time and add to
* the last seen timestamp */ * the last seen timestamp */
timestamp += timestamp += payload->priv->get_duration (payload, distance);
gst_base_rtp_audio_payload_get_sample_duration (payload, distance);
} }
/* and flush out the bytes from the adapter */ /* and flush out the bytes from the adapter */