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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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gst/audioresample/: Fix audioresample, seek torture, new segments, reverse negotiation etc.. work fine.
Original commit message from CVS: * gst/audioresample/buffer.c: (audioresample_buffer_queue_flush): * gst/audioresample/buffer.h: * gst/audioresample/gstaudioresample.c: * gst/audioresample/gstaudioresample.h: * gst/audioresample/resample.c: (resample_input_flush), (resample_input_pushthrough), (resample_input_eos), (resample_get_output_size_for_input), (resample_get_input_size_for_output), (resample_get_output_size), (resample_get_output_data): * gst/audioresample/resample.h: * gst/audioresample/resample_ref.c: (resample_scale_ref): Fix audioresample, seek torture, new segments, reverse negotiation etc.. work fine.
This commit is contained in:
Wim Taymans
parent
130b68aeff
commit
c962e657c3
7 changed files with 291 additions and 74 deletions
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@ -237,3 +237,17 @@ audioresample_buffer_queue_peek (AudioresampleBufferQueue * queue, int length)
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return newbuffer;
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}
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void
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audioresample_buffer_queue_flush (AudioresampleBufferQueue * queue)
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{
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GList *g;
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for (g = g_list_first (queue->buffers); g; g = g_list_next (g)) {
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audioresample_buffer_unref ((AudioresampleBuffer *) g->data);
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}
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g_list_free (queue->buffers);
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queue->buffers = NULL;
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queue->depth = 0;
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queue->offset = 0;
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}
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@ -28,21 +28,24 @@ struct _AudioresampleBufferQueue
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int offset;
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};
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AudioresampleBuffer *audioresample_buffer_new (void);
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AudioresampleBuffer *audioresample_buffer_new_and_alloc (int size);
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AudioresampleBuffer *audioresample_buffer_new_with_data (void *data, int size);
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AudioresampleBuffer *audioresample_buffer_new_subbuffer (AudioresampleBuffer * buffer, int offset,
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int length);
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void audioresample_buffer_ref (AudioresampleBuffer * buffer);
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void audioresample_buffer_unref (AudioresampleBuffer * buffer);
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AudioresampleBuffer * audioresample_buffer_new (void);
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AudioresampleBuffer * audioresample_buffer_new_and_alloc (int size);
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AudioresampleBuffer * audioresample_buffer_new_with_data (void *data, int size);
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AudioresampleBuffer * audioresample_buffer_new_subbuffer (AudioresampleBuffer * buffer,
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int offset,
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int length);
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void audioresample_buffer_ref (AudioresampleBuffer * buffer);
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void audioresample_buffer_unref (AudioresampleBuffer * buffer);
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AudioresampleBufferQueue *audioresample_buffer_queue_new (void);
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void audioresample_buffer_queue_free (AudioresampleBufferQueue * queue);
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int audioresample_buffer_queue_get_depth (AudioresampleBufferQueue * queue);
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int audioresample_buffer_queue_get_offset (AudioresampleBufferQueue * queue);
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void audioresample_buffer_queue_push (AudioresampleBufferQueue * queue,
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AudioresampleBuffer * buffer);
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AudioresampleBuffer *audioresample_buffer_queue_pull (AudioresampleBufferQueue * queue, int len);
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AudioresampleBuffer *audioresample_buffer_queue_peek (AudioresampleBufferQueue * queue, int len);
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AudioresampleBufferQueue *
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audioresample_buffer_queue_new (void);
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void audioresample_buffer_queue_free (AudioresampleBufferQueue * queue);
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int audioresample_buffer_queue_get_depth (AudioresampleBufferQueue * queue);
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int audioresample_buffer_queue_get_offset (AudioresampleBufferQueue * queue);
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void audioresample_buffer_queue_push (AudioresampleBufferQueue * queue,
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AudioresampleBuffer * buffer);
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AudioresampleBuffer * audioresample_buffer_queue_pull (AudioresampleBufferQueue * queue, int len);
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AudioresampleBuffer * audioresample_buffer_queue_peek (AudioresampleBufferQueue * queue, int len);
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void audioresample_buffer_queue_flush (AudioresampleBufferQueue * queue);
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#endif
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@ -48,6 +48,8 @@ enum
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LAST_SIGNAL
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};
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#define DEFAULT_FILTERLEN 16
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enum
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{
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ARG_0,
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@ -97,8 +99,12 @@ GST_STATIC_CAPS ( \
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GstCaps * outcaps, guint * outsize);
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gboolean audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
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GstCaps * outcaps);
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static GstFlowReturn audioresample_pushthrough (GstAudioresample *
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audioresample);
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static GstFlowReturn audioresample_transform (GstBaseTransform * base,
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GstBuffer * inbuf, GstBuffer * outbuf);
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static gboolean audioresample_event (GstBaseTransform * base,
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GstEvent * event);
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/*static guint gst_audioresample_signals[LAST_SIGNAL] = { 0 }; */
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@ -133,7 +139,8 @@ static void gst_audioresample_class_init (GstAudioresampleClass * klass)
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FILTERLEN,
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g_param_spec_int ("filter_length", "filter_length", "filter_length",
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0, G_MAXINT, 16, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
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0, G_MAXINT, DEFAULT_FILTERLEN,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
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GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
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GST_DEBUG_FUNCPTR (audioresample_transform_size);
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@ -145,19 +152,32 @@ static void gst_audioresample_class_init (GstAudioresampleClass * klass)
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GST_DEBUG_FUNCPTR (audioresample_set_caps);
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GST_BASE_TRANSFORM_CLASS (klass)->transform =
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GST_DEBUG_FUNCPTR (audioresample_transform);
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GST_BASE_TRANSFORM_CLASS (klass)->event =
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GST_DEBUG_FUNCPTR (audioresample_event);
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GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE;
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}
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static void gst_audioresample_init (GstAudioresample * audioresample,
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static void
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gst_audioresample_init (GstAudioresample * audioresample,
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GstAudioresampleClass * klass)
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{
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ResampleState *r;
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GstBaseTransform *trans;
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trans = GST_BASE_TRANSFORM (audioresample);
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/* buffer alloc passthrough is too impossible. FIXME, it
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* is trivial in the passtrough case. */
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gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
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r = resample_new ();
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audioresample->resample = r;
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audioresample->ts_offset = -1;
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audioresample->offset = -1;
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audioresample->next_ts = -1;
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resample_set_filter_length (r, 64);
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resample_set_filter_length (r, DEFAULT_FILTERLEN);
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resample_set_format (r, RESAMPLE_FORMAT_S16);
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}
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@ -197,16 +217,14 @@ gboolean
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GstCaps *audioresample_transform_caps (GstBaseTransform * base,
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GstPadDirection direction, GstCaps * caps)
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{
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GstCaps *temp, *res;
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const GstCaps *templcaps;
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GstCaps *res;
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GstStructure *structure;
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temp = gst_caps_copy (caps);
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structure = gst_caps_get_structure (temp, 0);
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gst_structure_remove_field (structure, "rate");
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templcaps = gst_pad_get_pad_template_caps (base->srcpad);
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res = gst_caps_intersect (templcaps, temp);
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gst_caps_unref (temp);
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/* transform caps gives one single caps so we can just replace
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* the rate property with our range. */
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res = gst_caps_copy (caps);
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structure = gst_caps_get_structure (res, 0);
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gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
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return res;
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}
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@ -286,6 +304,7 @@ gboolean
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GST_DEBUG_OBJECT (audioresample,
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"caps are not the set caps, creating state");
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state = resample_new ();
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resample_set_filter_length (state, audioresample->filter_length);
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resample_set_state_from_caps (state, sinkcaps, srccaps, NULL, NULL, NULL);
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}
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@ -293,12 +312,9 @@ gboolean
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/* asked to convert size of an incoming buffer */
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*othersize = resample_get_output_size_for_input (state, size);
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} else {
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/* take a best guess, this is called cheating */
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*othersize = floor (size * state->i_rate / state->o_rate);
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*othersize -= *othersize % state->sample_size;
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/* asked to convert size of an outgoing buffer */
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*othersize = resample_get_input_size_for_output (state, size);
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}
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*othersize += state->sample_size;
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g_assert (*othersize % state->sample_size == 0);
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/* we make room for one extra sample, given that the resampling filter
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@ -346,35 +362,50 @@ gboolean
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return TRUE;
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}
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static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event)
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{
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GstAudioresample *audioresample;
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audioresample = GST_AUDIORESAMPLE (base);
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_FLUSH_START:
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break;
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case GST_EVENT_FLUSH_STOP:
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resample_input_flush (audioresample->resample);
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audioresample->ts_offset = -1;
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audioresample->next_ts = -1;
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audioresample->offset = -1;
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break;
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case GST_EVENT_NEWSEGMENT:
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resample_input_pushthrough (audioresample->resample);
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audioresample_pushthrough (audioresample);
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audioresample->ts_offset = -1;
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audioresample->next_ts = -1;
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audioresample->offset = -1;
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break;
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case GST_EVENT_EOS:
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resample_input_eos (audioresample->resample);
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audioresample_pushthrough (audioresample);
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break;
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default:
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break;
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}
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parent_class->event (base, event);
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return TRUE;
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}
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static GstFlowReturn
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audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
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audioresample_do_output (GstAudioresample * audioresample,
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GstBuffer * outbuf)
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{
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/* FIXME: this-> */
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GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
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ResampleState *r;
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guchar *data;
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gulong size;
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int outsize;
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int outsamples;
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/* FIXME: move to _inplace */
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#if 0
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if (audioresample->passthru) {
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gst_pad_push (audioresample->srcpad, GST_DATA (buf));
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return;
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}
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#endif
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ResampleState *r;
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r = audioresample->resample;
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data = GST_BUFFER_DATA (inbuf);
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size = GST_BUFFER_SIZE (inbuf);
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GST_DEBUG_OBJECT (audioresample, "got buffer of %ld bytes", size);
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resample_add_input_data (r, data, size, NULL, NULL);
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outsize = resample_get_output_size (r);
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GST_DEBUG_OBJECT (audioresample, "audioresample can give me %d bytes",
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outsize);
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@ -399,18 +430,27 @@ static GstFlowReturn
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outsize, outsamples);
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GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
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GST_BUFFER_TIMESTAMP (outbuf) = base->segment.start +
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audioresample->offset * GST_SECOND / audioresample->o_rate;
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GST_BUFFER_TIMESTAMP (outbuf) = audioresample->next_ts;
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audioresample->offset += outsamples;
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GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset;
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if (audioresample->ts_offset != -1) {
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audioresample->offset += outsamples;
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audioresample->ts_offset += outsamples;
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audioresample->next_ts =
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gst_util_uint64_scale_int (audioresample->ts_offset, GST_SECOND,
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audioresample->o_rate);
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GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset;
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/* we calculate DURATION as the difference between "next" timestamp
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* and current timestamp so we ensure a contiguous stream, instead of
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* having rounding errors. */
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GST_BUFFER_DURATION (outbuf) = base->segment.start +
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audioresample->offset * GST_SECOND / audioresample->o_rate -
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GST_BUFFER_TIMESTAMP (outbuf);
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/* we calculate DURATION as the difference between "next" timestamp
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* and current timestamp so we ensure a contiguous stream, instead of
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* having rounding errors. */
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GST_BUFFER_DURATION (outbuf) = audioresample->next_ts -
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GST_BUFFER_TIMESTAMP (outbuf);
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} else {
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/* no valid offset know, we can still sortof calculate the duration though */
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GST_BUFFER_DURATION (outbuf) =
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gst_util_uint64_scale_int (outsamples, GST_SECOND,
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audioresample->o_rate);
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}
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/* check for possible mem corruption */
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if (outsize > GST_BUFFER_SIZE (outbuf)) {
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@ -429,10 +469,87 @@ static GstFlowReturn
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"audioresample's written outsize %d too far from outbuffer's size %d",
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outsize, GST_BUFFER_SIZE (outbuf));
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}
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GST_BUFFER_SIZE (outbuf) = outsize;
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return GST_FLOW_OK;
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}
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static GstFlowReturn
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audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
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GstBuffer * outbuf)
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{
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GstAudioresample *audioresample;
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ResampleState *r;
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guchar *data;
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gulong size;
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GstClockTime timestamp;
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audioresample = GST_AUDIORESAMPLE (base);
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r = audioresample->resample;
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data = GST_BUFFER_DATA (inbuf);
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size = GST_BUFFER_SIZE (inbuf);
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timestamp = GST_BUFFER_TIMESTAMP (inbuf);
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GST_DEBUG_OBJECT (audioresample, "got buffer of %ld bytes", size);
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if (audioresample->ts_offset == -1) {
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/* if we don't know the initial offset yet, calculate it based on the
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* input timestamp. */
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if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
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GstClockTime stime;
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/* offset used to calculate the timestamps. We use the sample offset for this
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* to make it more accurate. We want the first buffer to have the same timestamp
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* as the incomming timestamp. */
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audioresample->next_ts = timestamp;
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audioresample->ts_offset =
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gst_util_uint64_scale_int (timestamp, r->o_rate, GST_SECOND);
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/* offset used to set as the buffer offset, this offset is always relative
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* to the stream time, note that timestamp is not... */
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stime = (timestamp - base->segment.start) + base->segment.time;
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audioresample->offset =
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gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND);
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}
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}
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/* need to memdup, resample takes ownership. */
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resample_add_input_data (r, g_memdup (data, size), size, NULL, NULL);
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return audioresample_do_output (audioresample, outbuf);
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}
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/* push remaining data in the buffers out */
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static GstFlowReturn
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audioresample_pushthrough (GstAudioresample * audioresample)
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{
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int outsize;
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ResampleState *r;
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GstBuffer *outbuf;
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GstFlowReturn res = GST_FLOW_OK;
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GstBaseTransform *trans;
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r = audioresample->resample;
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outsize = resample_get_output_size (r);
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if (outsize == 0)
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goto done;
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outbuf = gst_buffer_new_and_alloc (outsize);
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res = audioresample_do_output (audioresample, outbuf);
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if (res != GST_FLOW_OK)
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goto done;
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trans = GST_BASE_TRANSFORM (audioresample);
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res = gst_pad_push (trans->srcpad, outbuf);
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done:
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return res;
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}
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static void
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gst_audioresample_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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@ -53,6 +53,8 @@ struct _GstAudioresample {
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gboolean passthru;
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guint64 offset;
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guint64 ts_offset;
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GstClockTime next_ts;
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int channels;
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int i_rate;
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@ -121,19 +121,50 @@ resample_add_input_data (ResampleState * r, void *data, int size,
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}
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void
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resample_input_eos (ResampleState * r)
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resample_input_flush (ResampleState * r)
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{
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RESAMPLE_DEBUG ("flush");
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audioresample_buffer_queue_flush (r->queue);
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r->buffer_filled = 0;
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r->need_reinit = 1;
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}
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void
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resample_input_pushthrough (ResampleState * r)
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{
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AudioresampleBuffer *buffer;
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int sample_size;
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int filter_bytes;
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int buffer_filled;
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sample_size = r->n_channels * resample_format_size (r->format);
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if (r->sample_size == 0)
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return;
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buffer = audioresample_buffer_new_and_alloc (sample_size *
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(r->filter_length / 2));
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filter_bytes = r->filter_length * r->sample_size;
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buffer_filled = r->buffer_filled;
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RESAMPLE_DEBUG ("pushthrough filter_bytes %d, filled %d",
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filter_bytes, buffer_filled);
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/* if we have no pending samples, we don't need to do anything. */
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if (buffer_filled <= 0)
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return;
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/* send filter_length/2 number of samples so we can get to the
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* last queued samples */
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buffer = audioresample_buffer_new_and_alloc (filter_bytes / 2);
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memset (buffer->data, 0, buffer->length);
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|
||||
audioresample_buffer_queue_push (r->queue, buffer);
|
||||
RESAMPLE_DEBUG ("pushthrough", buffer->length);
|
||||
|
||||
audioresample_buffer_queue_push (r->queue, buffer);
|
||||
}
|
||||
|
||||
void
|
||||
resample_input_eos (ResampleState * r)
|
||||
{
|
||||
RESAMPLE_DEBUG ("EOS");
|
||||
resample_input_pushthrough (r);
|
||||
r->eos = 1;
|
||||
}
|
||||
|
||||
|
|
@ -142,22 +173,61 @@ resample_get_output_size_for_input (ResampleState * r, int size)
|
|||
{
|
||||
int outsize;
|
||||
double outd;
|
||||
int avail;
|
||||
int filter_bytes;
|
||||
int buffer_filled;
|
||||
|
||||
g_return_val_if_fail (r->sample_size != 0, 0);
|
||||
if (r->sample_size == 0)
|
||||
return 0;
|
||||
|
||||
filter_bytes = r->filter_length * r->sample_size;
|
||||
buffer_filled = filter_bytes / 2 - r->buffer_filled / 2;
|
||||
|
||||
avail =
|
||||
audioresample_buffer_queue_get_depth (r->queue) + size - buffer_filled;
|
||||
|
||||
RESAMPLE_DEBUG ("avail %d, o_rate %f, i_rate %f, filter_bytes %d, filled %d",
|
||||
avail, r->o_rate, r->i_rate, filter_bytes, buffer_filled);
|
||||
if (avail <= 0)
|
||||
return 0;
|
||||
|
||||
outd = (double) avail *r->o_rate / r->i_rate;
|
||||
|
||||
RESAMPLE_DEBUG ("size %d, o_rate %f, i_rate %f", size, r->o_rate, r->i_rate);
|
||||
outd = (double) size / r->i_rate * r->o_rate;
|
||||
outsize = (int) floor (outd);
|
||||
|
||||
/* round off for sample size */
|
||||
return outsize - (outsize % r->sample_size);
|
||||
outsize -= outsize % r->sample_size;
|
||||
|
||||
return outsize;
|
||||
}
|
||||
|
||||
int
|
||||
resample_get_input_size_for_output (ResampleState * r, int size)
|
||||
{
|
||||
int outsize;
|
||||
double outd;
|
||||
int avail;
|
||||
|
||||
if (r->sample_size == 0)
|
||||
return 0;
|
||||
|
||||
avail = size;
|
||||
|
||||
RESAMPLE_DEBUG ("size %d, o_rate %f, i_rate %f", avail, r->o_rate, r->i_rate);
|
||||
outd = (double) avail *r->i_rate / r->o_rate;
|
||||
|
||||
outsize = (int) ceil (outd);
|
||||
|
||||
/* round off for sample size */
|
||||
outsize -= outsize % r->sample_size;
|
||||
|
||||
return outsize;
|
||||
}
|
||||
|
||||
int
|
||||
resample_get_output_size (ResampleState * r)
|
||||
{
|
||||
return resample_get_output_size_for_input (r,
|
||||
audioresample_buffer_queue_get_depth (r->queue));
|
||||
return resample_get_output_size_for_input (r, 0);
|
||||
}
|
||||
|
||||
int
|
||||
|
|
@ -166,6 +236,9 @@ resample_get_output_data (ResampleState * r, void *data, int size)
|
|||
r->o_buf = data;
|
||||
r->o_size = size;
|
||||
|
||||
if (size == 0)
|
||||
return 0;
|
||||
|
||||
switch (r->method) {
|
||||
case 0:
|
||||
resample_scale_ref (r);
|
||||
|
|
|
|||
|
|
@ -67,6 +67,7 @@ struct _ResampleState {
|
|||
|
||||
void *buffer;
|
||||
int buffer_len;
|
||||
int buffer_filled;
|
||||
|
||||
double i_start;
|
||||
double o_start;
|
||||
|
|
@ -98,8 +99,12 @@ void resample_free (ResampleState *state);
|
|||
void resample_add_input_data (ResampleState * r, void *data, int size,
|
||||
ResampleCallback free_func, void *closure);
|
||||
void resample_input_eos (ResampleState *r);
|
||||
void resample_input_flush (ResampleState *r);
|
||||
void resample_input_pushthrough (ResampleState *r);
|
||||
|
||||
int resample_get_output_size_for_input (ResampleState * r, int size);
|
||||
int resample_get_input_size_for_output (ResampleState * r, int size);
|
||||
|
||||
int resample_get_output_size (ResampleState *r);
|
||||
int resample_get_output_data (ResampleState *r, void *data, int size);
|
||||
|
||||
|
|
|
|||
|
|
@ -63,6 +63,7 @@ resample_scale_ref (ResampleState * r)
|
|||
r->buffer_len = r->sample_size * r->filter_length;
|
||||
r->buffer = malloc (r->buffer_len);
|
||||
memset (r->buffer, 0, r->buffer_len);
|
||||
r->buffer_filled = 0;
|
||||
|
||||
r->i_inc = r->o_rate / r->i_rate;
|
||||
r->o_inc = r->i_rate / r->o_rate;
|
||||
|
|
@ -127,6 +128,8 @@ resample_scale_ref (ResampleState * r)
|
|||
|
||||
memcpy (r->buffer + r->buffer_len - r->sample_size, buffer->data,
|
||||
r->sample_size);
|
||||
r->buffer_filled = MIN (r->buffer_filled + r->sample_size, r->buffer_len);
|
||||
|
||||
audioresample_buffer_unref (buffer);
|
||||
}
|
||||
|
||||
|
|
|
|||
Loading…
Reference in a new issue