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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-29 05:01:23 +00:00
audiodecoder: improve reverse playback
... by doing some more (reverse) timestamp interpolating and refactoring downstream pushing. Fixes #661983.
This commit is contained in:
Mark Nauwelaerts
parent
cd21e69913
commit
38615abdd8
1 changed files with 94 additions and 54 deletions
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@ -558,6 +558,78 @@ gst_audio_decoder_setup (GstAudioDecoder * dec)
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dec->priv->agg = !!res;
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}
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static GstFlowReturn
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gst_audio_decoder_push_forward (GstAudioDecoder * dec, GstBuffer * buf)
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{
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GstAudioDecoderClass *klass;
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GstAudioDecoderPrivate *priv;
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GstAudioDecoderContext *ctx;
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GstFlowReturn ret = GST_FLOW_OK;
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klass = GST_AUDIO_DECODER_GET_CLASS (dec);
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priv = dec->priv;
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ctx = &dec->priv->ctx;
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g_return_val_if_fail (ctx->info.bpf != 0, GST_FLOW_ERROR);
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if (G_UNLIKELY (!buf)) {
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g_assert_not_reached ();
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return GST_FLOW_OK;
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}
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GST_LOG_OBJECT (dec, "clipping 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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/* clip buffer */
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buf = gst_audio_buffer_clip (buf, &dec->segment, ctx->info.rate,
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ctx->info.bpf);
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if (G_UNLIKELY (!buf)) {
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GST_DEBUG_OBJECT (dec, "no data after clipping to segment");
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goto exit;
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}
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/* decorate */
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gst_buffer_set_caps (buf, GST_PAD_CAPS (dec->srcpad));
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if (G_UNLIKELY (priv->discont)) {
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GST_LOG_OBJECT (dec, "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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/* track where we are */
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if (G_LIKELY (GST_BUFFER_TIMESTAMP_IS_VALID (buf))) {
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/* duration should always be valid for raw audio */
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g_assert (GST_BUFFER_DURATION_IS_VALID (buf));
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dec->segment.last_stop =
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GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
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}
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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 (dec, &buf);
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if (ret != GST_FLOW_OK || !buf) {
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GST_DEBUG_OBJECT (dec, "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 (dec, "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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ret = gst_pad_push (dec->srcpad, buf);
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exit:
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return ret;
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}
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/* mini aggregator combining output buffers into fewer larger ones,
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* if so allowed/configured */
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static GstFlowReturn
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@ -577,27 +649,10 @@ gst_audio_decoder_output (GstAudioDecoder * dec, GstBuffer * buf)
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gst_audio_decoder_setup (dec);
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if (G_LIKELY (buf)) {
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g_return_val_if_fail (ctx->info.bpf != 0, GST_FLOW_ERROR);
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GST_LOG_OBJECT (dec, "output 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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/* clip buffer */
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buf = gst_audio_buffer_clip (buf, &dec->segment, ctx->info.rate,
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ctx->info.bpf);
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if (G_UNLIKELY (!buf)) {
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GST_DEBUG_OBJECT (dec, "no data after clipping to segment");
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} else {
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GST_LOG_OBJECT (dec,
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"buffer after segment clipping has 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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}
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} else {
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GST_DEBUG_OBJECT (dec, "no output buffer");
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}
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again:
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@ -647,42 +702,8 @@ again:
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}
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if (G_LIKELY (buf)) {
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/* decorate */
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gst_buffer_set_caps (buf, GST_PAD_CAPS (dec->srcpad));
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if (G_UNLIKELY (priv->discont)) {
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GST_LOG_OBJECT (dec, "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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if (G_LIKELY (GST_BUFFER_TIMESTAMP_IS_VALID (buf))) {
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/* duration should always be valid for raw audio */
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g_assert (GST_BUFFER_DURATION_IS_VALID (buf));
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dec->segment.last_stop =
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GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
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}
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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 (dec, &buf);
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if (ret != GST_FLOW_OK || !buf) {
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GST_DEBUG_OBJECT (dec, "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 (dec, "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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if (dec->segment.rate > 0.0) {
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ret = gst_pad_push (dec->srcpad, buf);
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ret = gst_audio_decoder_push_forward (dec, buf);
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GST_LOG_OBJECT (dec, "buffer pushed: %s", gst_flow_get_name (ret));
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} else {
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ret = GST_FLOW_OK;
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@ -690,7 +711,6 @@ again:
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GST_LOG_OBJECT (dec, "buffer queued");
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}
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exit:
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if (inbuf) {
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buf = inbuf;
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goto again;
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@ -1176,6 +1196,7 @@ gst_audio_decoder_flush_decode (GstAudioDecoder * dec)
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{
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GstAudioDecoderPrivate *priv = dec->priv;
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GstFlowReturn res = GST_FLOW_OK;
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GstClockTime timestamp;
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GList *walk;
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walk = priv->decode;
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@ -1213,9 +1234,28 @@ gst_audio_decoder_flush_decode (GstAudioDecoder * dec)
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gst_audio_decoder_drain (dec);
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/* now send queued data downstream */
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timestamp = GST_CLOCK_TIME_NONE;
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while (priv->queued) {
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GstBuffer *buf = GST_BUFFER_CAST (priv->queued->data);
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/* duration should always be valid for raw audio */
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g_assert (GST_BUFFER_DURATION_IS_VALID (buf));
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/* interpolate (backward) if needed */
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if (G_LIKELY (timestamp != -1))
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timestamp -= GST_BUFFER_DURATION (buf);
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if (!GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
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GST_LOG_OBJECT (dec, "applying reverse interpolated ts %"
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GST_TIME_FORMAT, GST_TIME_ARGS (timestamp));
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GST_BUFFER_TIMESTAMP (buf) = timestamp;
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} else {
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/* track otherwise */
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timestamp = GST_BUFFER_TIMESTAMP (buf);
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GST_LOG_OBJECT (dec, "tracking ts %" GST_TIME_FORMAT,
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GST_TIME_ARGS (timestamp));
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}
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if (G_LIKELY (res == GST_FLOW_OK)) {
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GST_DEBUG_OBJECT (dec, "pushing buffer %p of size %u, "
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"time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, buf,
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@ -1226,7 +1266,7 @@ gst_audio_decoder_flush_decode (GstAudioDecoder * dec)
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/* avoid stray DISCONT from forward processing,
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* which have no meaning in reverse pushing */
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GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
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res = gst_pad_push (dec->srcpad, buf);
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res = gst_audio_decoder_push_forward (dec, buf);
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} else {
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gst_buffer_unref (buf);
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}
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