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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-23 18:21:04 +00:00
baseaudioencoder: minor fixes and code simplifications
Also modify and elaborate a bit on pre_push (though currently unused to no harm).
This commit is contained in:
Mark Nauwelaerts
Tim-Philipp Müller
parent
d0e9fbf3db
commit
9ce2edc918
2 changed files with 29 additions and 43 deletions
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@ -196,10 +196,6 @@ struct _GstBaseAudioEncoderPrivate
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GstAdapter *adapter;
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/* offset in adapter up to which already supplied to encoder */
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gint offset;
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/* collected encoded data */
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GstAdapter *adapter_out;
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/* (estimated) samples (w.r.t. input rate) represented in adapter_out */
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gint samples_out;
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/* mark outgoing discont */
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gboolean discont;
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/* to guess duration of drained data */
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@ -375,7 +371,6 @@ gst_base_audio_encoder_init (GstBaseAudioEncoder * enc,
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GST_DEBUG_OBJECT (enc, "src created");
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enc->priv->adapter = gst_adapter_new ();
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enc->priv->adapter_out = gst_adapter_new ();
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enc->ctx = &enc->priv->ctx;
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/* property default */
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@ -409,14 +404,12 @@ gst_base_audio_encoder_reset (GstBaseAudioEncoder * enc, gboolean full)
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gst_segment_init (&enc->segment, GST_FORMAT_TIME);
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gst_adapter_clear (enc->priv->adapter);
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gst_adapter_clear (enc->priv->adapter_out);
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enc->priv->got_data = FALSE;
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enc->priv->drained = TRUE;
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enc->priv->offset = 0;
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enc->priv->base_ts = GST_CLOCK_TIME_NONE;
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enc->priv->base_gp = -1;
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enc->priv->samples = 0;
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enc->priv->samples_out = 0;
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enc->priv->discont = FALSE;
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GST_OBJECT_UNLOCK (enc);
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@ -428,7 +421,6 @@ gst_base_audio_encoder_finalize (GObject * object)
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GstBaseAudioEncoder *enc = GST_BASE_AUDIO_ENCODER (object);
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g_object_unref (enc->priv->adapter);
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g_object_unref (enc->priv->adapter_out);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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@ -464,7 +456,6 @@ gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc, GstBuffer * buf,
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GstBaseAudioEncoderPrivate *priv;
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GstBaseAudioEncoderContext *ctx;
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GstFlowReturn ret = GST_FLOW_OK;
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gint av;
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klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
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priv = enc->priv;
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@ -487,8 +478,8 @@ gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc, GstBuffer * buf,
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samples = (enc->priv->offset / ctx->state.bpf);
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if (G_LIKELY (samples)) {
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/* track upstream ts at output collection start if so configured */
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if (!enc->perfect_ts && !priv->samples_out) {
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/* track upstream ts if so configured */
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if (!enc->perfect_ts) {
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guint64 ts, distance;
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ts = gst_adapter_prev_timestamp (priv->adapter, &distance);
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@ -544,21 +535,12 @@ gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc, GstBuffer * buf,
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if (G_UNLIKELY (gst_adapter_available (priv->adapter) == 0))
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gst_adapter_clear (priv->adapter);
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}
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if (G_LIKELY (buf)) {
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priv->samples_out += samples;
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samples = 0;
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}
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/* otherwise retain count of to-be-discarded samples */
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/* sample count advanced below after buffer handling */
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}
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/* collect output */
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if (G_LIKELY (buf))
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gst_adapter_push (enc->priv->adapter_out, buf);
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av = gst_adapter_available (priv->adapter_out);
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if (av) {
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GST_LOG_OBJECT (enc, "collecting all %d bytes for output", av);
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buf = gst_adapter_take_buffer (priv->adapter_out, av);
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if (G_LIKELY (buf)) {
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GST_LOG_OBJECT (enc, "taking %d bytes for output", GST_BUFFER_SIZE (buf));
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buf = gst_buffer_make_metadata_writable (buf);
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/* decorate */
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@ -570,10 +552,9 @@ gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc, GstBuffer * buf,
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GST_BUFFER_TIMESTAMP (buf) = priv->base_ts +
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gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
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ctx->state.rate);
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GST_DEBUG_OBJECT (enc, "out samples %d", (gint) priv->samples_out);
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if (G_LIKELY (priv->samples_out > 0)) {
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priv->samples += priv->samples_out;
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priv->samples_out = 0;
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GST_DEBUG_OBJECT (enc, "out samples %d", samples);
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if (G_LIKELY (samples > 0)) {
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priv->samples += samples;
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GST_BUFFER_DURATION (buf) = priv->base_ts +
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gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
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ctx->state.rate) - GST_BUFFER_TIMESTAMP (buf);
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@ -599,29 +580,39 @@ gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc, GstBuffer * buf,
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}
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}
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priv->bytes_out += GST_BUFFER_SIZE (buf);
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if (G_UNLIKELY (priv->discont)) {
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GST_LOG_OBJECT (enc, "marking discont");
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GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
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priv->discont = FALSE;
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}
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// TODO return value ?
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if (klass->pre_push)
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klass->pre_push (enc, buf);
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if (klass->pre_push) {
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/* last chance for subclass to do some dirty stuff */
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ret = klass->pre_push (enc, &buf);
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if (ret != GST_FLOW_OK || !buf) {
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GST_DEBUG_OBJECT (enc, "subclass returned %s, buf %p",
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gst_flow_get_name (ret), buf);
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if (buf)
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gst_buffer_unref (buf);
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goto exit;
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}
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}
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GST_LOG_OBJECT (enc, "pushing buffer of size %d with ts %" GST_TIME_FORMAT
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", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
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GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
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priv->bytes_out += GST_BUFFER_SIZE (buf);
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ret = gst_pad_push (enc->srcpad, buf);
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GST_LOG_OBJECT (enc, "buffer pushed: %s", gst_flow_get_name (ret));
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} else {
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/* merely advance samples, most work for that already done above */
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priv->samples += samples;
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}
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/* account for discarded */
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priv->samples += samples;
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exit:
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return ret;
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/* ERRORS */
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@ -657,10 +648,6 @@ gst_base_audio_encoder_push_buffers (GstBaseAudioEncoder * enc, gboolean force)
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priv = enc->priv;
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ctx = enc->ctx;
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/* ensure clear start */
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gst_adapter_clear (priv->adapter_out);
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priv->samples_out = 0;
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while (ret == GST_FLOW_OK) {
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buf = NULL;
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@ -723,9 +710,6 @@ gst_base_audio_encoder_push_buffers (GstBaseAudioEncoder * enc, gboolean force)
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}
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}
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if (gst_adapter_available (priv->adapter_out))
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gst_base_audio_encoder_finish_frame (enc, NULL, 0);
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return ret;
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}
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@ -188,6 +188,8 @@ struct _GstBaseAudioEncoder {
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* (i.e. not unref'ed).
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* @pre_push: Optional.
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* Called just prior to pushing (encoded data) buffer downstream.
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* Subclass has full discretionary access to buffer,
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* and a not OK flow return will abort downstream pushing.
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* @getcaps: Optional.
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* Allows for a custom sink getcaps implementation (e.g.
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* for multichannel input specification). If not implemented,
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@ -216,7 +218,7 @@ struct _GstBaseAudioEncoderClass {
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void (*flush) (GstBaseAudioEncoder *enc);
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GstFlowReturn (*pre_push) (GstBaseAudioEncoder *enc,
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GstBuffer *buffer);
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GstBuffer **buffer);
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gboolean (*event) (GstBaseAudioEncoder *enc,
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GstEvent *event);
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