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path: root/omx/gstomxaudioenc.c
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Diffstat (limited to 'omx/gstomxaudioenc.c')
-rw-r--r--omx/gstomxaudioenc.c199
1 files changed, 103 insertions, 96 deletions
diff --git a/omx/gstomxaudioenc.c b/omx/gstomxaudioenc.c
index 496c539..1a2344b 100644
--- a/omx/gstomxaudioenc.c
+++ b/omx/gstomxaudioenc.c
@@ -282,7 +282,6 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
GstOMXBuffer *buf = NULL;
GstFlowReturn flow_ret = GST_FLOW_OK;
GstOMXAcquireBufferReturn acq_return;
- gboolean is_eos;
OMX_ERRORTYPE err;
klass = GST_OMX_AUDIO_ENC_GET_CLASS (self);
@@ -292,6 +291,8 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
goto component_error;
} else if (acq_return == GST_OMX_ACQUIRE_BUFFER_FLUSHING) {
goto flushing;
+ } else if (acq_return == GST_OMX_ACQUIRE_BUFFER_EOS) {
+ goto eos;
}
if (!gst_pad_has_current_caps (GST_AUDIO_ENCODER_SRC_PAD (self))
@@ -374,120 +375,103 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
}
g_assert (acq_return == GST_OMX_ACQUIRE_BUFFER_OK);
-
- if (buf) {
- GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %lu", buf->omx_buf->nFlags,
- buf->omx_buf->nTimeStamp);
-
- /* This prevents a deadlock between the srcpad stream
- * lock and the videocodec stream lock, if ::reset()
- * is called at the wrong time
- */
- if (gst_omx_port_is_flushing (self->enc_out_port)) {
- GST_DEBUG_OBJECT (self, "Flushing");
- gst_omx_port_release_buffer (self->enc_out_port, buf);
- goto flushing;
- }
-
+ if (!buf) {
+ g_assert ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER));
GST_AUDIO_ENCODER_STREAM_LOCK (self);
- is_eos = ! !(buf->omx_buf->nFlags & OMX_BUFFERFLAG_EOS);
+ goto eos;
+ }
- if ((buf->omx_buf->nFlags & OMX_BUFFERFLAG_CODECCONFIG)
- && buf->omx_buf->nFilledLen > 0) {
- GstCaps *caps;
- GstBuffer *codec_data;
- GstMapInfo map = GST_MAP_INFO_INIT;
+ GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %lu", buf->omx_buf->nFlags,
+ buf->omx_buf->nTimeStamp);
- GST_DEBUG_OBJECT (self, "Handling codec data");
- caps =
- gst_caps_copy (gst_pad_get_current_caps (GST_AUDIO_ENCODER_SRC_PAD
- (self)));
- codec_data = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
+ /* This prevents a deadlock between the srcpad stream
+ * lock and the videocodec stream lock, if ::reset()
+ * is called at the wrong time
+ */
+ if (gst_omx_port_is_flushing (self->enc_out_port)) {
+ GST_DEBUG_OBJECT (self, "Flushing");
+ gst_omx_port_release_buffer (self->enc_out_port, buf);
+ goto flushing;
+ }
- gst_buffer_map (codec_data, &map, GST_MAP_WRITE);
- memcpy (map.data,
- buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
- buf->omx_buf->nFilledLen);
- gst_buffer_unmap (codec_data, &map);
-
- gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data,
- NULL);
- if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
- gst_caps_unref (caps);
- if (buf)
- gst_omx_port_release_buffer (self->enc_out_port, buf);
- GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
- goto caps_failed;
- }
- gst_caps_unref (caps);
- flow_ret = GST_FLOW_OK;
- } else if (buf->omx_buf->nFilledLen > 0) {
- GstBuffer *outbuf;
- guint n_samples;
+ GST_AUDIO_ENCODER_STREAM_LOCK (self);
- GST_DEBUG_OBJECT (self, "Handling output data");
+ if ((buf->omx_buf->nFlags & OMX_BUFFERFLAG_CODECCONFIG)
+ && buf->omx_buf->nFilledLen > 0) {
+ GstCaps *caps;
+ GstBuffer *codec_data;
+ GstMapInfo map = GST_MAP_INFO_INIT;
- n_samples =
- klass->get_num_samples (self, self->enc_out_port,
- gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self)), buf);
+ GST_DEBUG_OBJECT (self, "Handling codec data");
+ caps =
+ gst_caps_copy (gst_pad_get_current_caps (GST_AUDIO_ENCODER_SRC_PAD
+ (self)));
+ codec_data = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
- if (buf->omx_buf->nFilledLen > 0) {
- GstMapInfo map = GST_MAP_INFO_INIT;
- outbuf = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
+ gst_buffer_map (codec_data, &map, GST_MAP_WRITE);
+ memcpy (map.data,
+ buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
+ buf->omx_buf->nFilledLen);
+ gst_buffer_unmap (codec_data, &map);
- gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
+ gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data, NULL);
+ if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
+ gst_caps_unref (caps);
+ if (buf)
+ gst_omx_port_release_buffer (self->enc_out_port, buf);
+ GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
+ goto caps_failed;
+ }
+ gst_caps_unref (caps);
+ flow_ret = GST_FLOW_OK;
+ } else if (buf->omx_buf->nFilledLen > 0) {
+ GstBuffer *outbuf;
+ guint n_samples;
- memcpy (map.data,
- buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
- buf->omx_buf->nFilledLen);
- gst_buffer_unmap (outbuf, &map);
+ GST_DEBUG_OBJECT (self, "Handling output data");
- } else {
- outbuf = gst_buffer_new ();
- }
+ n_samples =
+ klass->get_num_samples (self, self->enc_out_port,
+ gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self)), buf);
- GST_BUFFER_TIMESTAMP (outbuf) =
- gst_util_uint64_scale (buf->omx_buf->nTimeStamp, GST_SECOND,
- OMX_TICKS_PER_SECOND);
- if (buf->omx_buf->nTickCount != 0)
- GST_BUFFER_DURATION (outbuf) =
- gst_util_uint64_scale (buf->omx_buf->nTickCount, GST_SECOND,
- OMX_TICKS_PER_SECOND);
-
- flow_ret =
- gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (self),
- outbuf, n_samples);
- }
+ if (buf->omx_buf->nFilledLen > 0) {
+ GstMapInfo map = GST_MAP_INFO_INIT;
+ outbuf = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
- GST_DEBUG_OBJECT (self, "Handled output data");
+ gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
+
+ memcpy (map.data,
+ buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
+ buf->omx_buf->nFilledLen);
+ gst_buffer_unmap (outbuf, &map);
- if (is_eos || flow_ret == GST_FLOW_EOS) {
- g_mutex_lock (&self->drain_lock);
- if (self->draining) {
- GST_DEBUG_OBJECT (self, "Drained");
- self->draining = FALSE;
- g_cond_broadcast (&self->drain_cond);
- } else if (flow_ret == GST_FLOW_OK) {
- GST_DEBUG_OBJECT (self, "Component signalled EOS");
- flow_ret = GST_FLOW_EOS;
- }
- g_mutex_unlock (&self->drain_lock);
} else {
- GST_DEBUG_OBJECT (self, "Finished frame: %s",
- gst_flow_get_name (flow_ret));
+ outbuf = gst_buffer_new ();
}
- err = gst_omx_port_release_buffer (port, buf);
- if (err != OMX_ErrorNone)
- goto release_error;
+ GST_BUFFER_TIMESTAMP (outbuf) =
+ gst_util_uint64_scale (buf->omx_buf->nTimeStamp, GST_SECOND,
+ OMX_TICKS_PER_SECOND);
+ if (buf->omx_buf->nTickCount != 0)
+ GST_BUFFER_DURATION (outbuf) =
+ gst_util_uint64_scale (buf->omx_buf->nTickCount, GST_SECOND,
+ OMX_TICKS_PER_SECOND);
- self->downstream_flow_ret = flow_ret;
- } else {
- g_assert ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER));
- GST_AUDIO_ENCODER_STREAM_LOCK (self);
- flow_ret = GST_FLOW_EOS;
+ flow_ret =
+ gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (self),
+ outbuf, n_samples);
}
+ GST_DEBUG_OBJECT (self, "Handled output data");
+
+ GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
+
+ err = gst_omx_port_release_buffer (port, buf);
+ if (err != OMX_ErrorNone)
+ goto release_error;
+
+ self->downstream_flow_ret = flow_ret;
+
if (flow_ret != GST_FLOW_OK)
goto flow_error;
@@ -515,6 +499,29 @@ flushing:
self->started = FALSE;
return;
}
+eos:
+ {
+ g_mutex_lock (&self->drain_lock);
+ if (self->draining) {
+ GST_DEBUG_OBJECT (self, "Drained");
+ self->draining = FALSE;
+ g_cond_broadcast (&self->drain_cond);
+ flow_ret = GST_FLOW_OK;
+ } else {
+ GST_DEBUG_OBJECT (self, "Component signalled EOS");
+ flow_ret = GST_FLOW_EOS;
+ }
+ g_mutex_unlock (&self->drain_lock);
+
+ self->downstream_flow_ret = flow_ret;
+
+ if (flow_ret != GST_FLOW_OK)
+ goto flow_error;
+
+ GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
+
+ return;
+ }
flow_error:
{
if (flow_ret == GST_FLOW_EOS) {
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