From 26b4a028d98811c91740b48fbaf461aba77a083d Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Sun, 27 Jan 2008 07:37:40 +0000 Subject: ext/soundtouch/gstpitch.*: Implement LATENCY query and notify about latency changes. Original commit message from CVS: * ext/soundtouch/gstpitch.cc: * ext/soundtouch/gstpitch.hh: Implement LATENCY query and notify about latency changes. Unfortunately we don't have a fixed latency but it changes a bit with each buffer so we only send an LATENCY event with the maximum latency if it changes. Always calculate the timestamp, duration, etc from the sample rate instead of using a pre-calculated duration for one sample to prevent large rounding errors. --- ChangeLog | 13 ++++++ ext/soundtouch/gstpitch.cc | 105 +++++++++++++++++++++++++++++++++++++++------ ext/soundtouch/gstpitch.hh | 3 +- 3 files changed, 106 insertions(+), 15 deletions(-) diff --git a/ChangeLog b/ChangeLog index 88f2535f..97930237 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,16 @@ +2008-01-27 Sebastian Dröge + + * ext/soundtouch/gstpitch.cc: + * ext/soundtouch/gstpitch.hh: + Implement LATENCY query and notify about latency changes. + Unfortunately we don't have a fixed latency but it changes + a bit with each buffer so we only send an LATENCY event with + the maximum latency if it changes. + + Always calculate the timestamp, duration, etc from the sample + rate instead of using a pre-calculated duration for one sample + to prevent large rounding errors. + 2008-01-27 Sebastian Dröge Based on a patch by: diff --git a/ext/soundtouch/gstpitch.cc b/ext/soundtouch/gstpitch.cc index 7ee726f0..9e68fda4 100644 --- a/ext/soundtouch/gstpitch.cc +++ b/ext/soundtouch/gstpitch.cc @@ -194,6 +194,7 @@ gst_pitch_init (GstPitch * pitch, GstPitchClass * pitch_class) pitch->priv->st->setPitch (pitch->pitch); pitch->priv->stream_time_ratio = 1.0; + pitch->min_latency = pitch->max_latency = 0; } @@ -312,7 +313,6 @@ gst_pitch_sink_setcaps (GstPad * pad, GstCaps * caps) /* calculate sample size */ pitch->sample_size = (sizeof (gfloat) * channels); - pitch->sample_duration = gst_util_uint64_scale_int (GST_SECOND, 1, rate); GST_OBJECT_UNLOCK (pitch); @@ -371,7 +371,8 @@ gst_pitch_prepare_buffer (GstPitch * pitch) return NULL; } - GST_BUFFER_DURATION (buffer) = samples * pitch->sample_duration; + GST_BUFFER_DURATION (buffer) = + gst_util_uint64_scale (samples, GST_SECOND, pitch->samplerate); /* temporary store samples here, to avoid having to recalculate this */ GST_BUFFER_OFFSET (buffer) = (gint64) samples; @@ -466,18 +467,17 @@ gst_pitch_convert (GstPitch * pitch, GstFormat * dst_format, gint64 * dst_value) { gboolean res = TRUE; - GstClockTime sample_duration; guint sample_size; + gint samplerate; g_return_val_if_fail (dst_format && dst_value, FALSE); GST_OBJECT_LOCK (pitch); - sample_duration = pitch->sample_duration; sample_size = pitch->sample_size; + samplerate = pitch->samplerate; GST_OBJECT_UNLOCK (pitch); - if (sample_size == 0 || sample_duration == 0 || - sample_duration == GST_CLOCK_TIME_NONE) { + if (sample_size == 0 || samplerate == 0) { return FALSE; } @@ -490,11 +490,12 @@ gst_pitch_convert (GstPitch * pitch, case GST_FORMAT_BYTES: switch (*dst_format) { case GST_FORMAT_TIME: - *dst_value = src_value / sample_size; - *dst_value *= sample_duration; + *dst_value = + gst_util_uint64_scale_int (src_value, GST_SECOND, + sample_size * samplerate); break; case GST_FORMAT_DEFAULT: - *dst_value = src_value / sample_size; + *dst_value = gst_util_uint64_scale_int (src_value, 1, sample_size); break; default: res = FALSE; @@ -504,11 +505,13 @@ gst_pitch_convert (GstPitch * pitch, case GST_FORMAT_TIME: switch (*dst_format) { case GST_FORMAT_BYTES: - *dst_value = src_value / sample_duration; - *dst_value *= sample_size; + *dst_value = + gst_util_uint64_scale_int (src_value, samplerate * sample_size, + GST_SECOND); break; case GST_FORMAT_DEFAULT: - *dst_value = src_value / sample_duration; + *dst_value = + gst_util_uint64_scale_int (src_value, samplerate, GST_SECOND); break; default: res = FALSE; @@ -518,10 +521,11 @@ gst_pitch_convert (GstPitch * pitch, case GST_FORMAT_DEFAULT: switch (*dst_format) { case GST_FORMAT_BYTES: - *dst_value = src_value * sample_size; + *dst_value = gst_util_uint64_scale_int (src_value, sample_size, 1); break; case GST_FORMAT_TIME: - *dst_value = src_value * sample_duration; + *dst_value = + gst_util_uint64_scale_int (src_value, GST_SECOND, samplerate); break; default: res = FALSE; @@ -543,6 +547,7 @@ gst_pitch_get_query_types (GstPad * pad) GST_QUERY_POSITION, GST_QUERY_DURATION, GST_QUERY_CONVERT, + GST_QUERY_LATENCY, GST_QUERY_NONE }; @@ -630,6 +635,46 @@ gst_pitch_src_query (GstPad * pad, GstQuery * query) } break; } + case GST_QUERY_LATENCY: + { + GstClockTime min, max; + gboolean live; + GstPad *peer; + + if ((peer = gst_pad_get_peer (pitch->sinkpad))) { + if ((res = gst_pad_query (peer, query))) { + gst_query_parse_latency (query, &live, &min, &max); + + GST_DEBUG ("Peer latency: min %" + GST_TIME_FORMAT " max %" GST_TIME_FORMAT, + GST_TIME_ARGS (min), GST_TIME_ARGS (max)); + + /* add our own latency */ + + GST_DEBUG ("Our latency: min %" GST_TIME_FORMAT + ", max %" GST_TIME_FORMAT, + GST_TIME_ARGS (pitch->min_latency), + GST_TIME_ARGS (pitch->max_latency)); + + min += pitch->min_latency; + if (max != GST_CLOCK_TIME_NONE) + max += pitch->max_latency; + else + max = pitch->max_latency; + + GST_DEBUG ("Calculated total latency : min %" + GST_TIME_FORMAT " max %" GST_TIME_FORMAT, + GST_TIME_ARGS (min), GST_TIME_ARGS (max)); + g_print ("Calculated total latency : min %" + GST_TIME_FORMAT " max %" GST_TIME_FORMAT, + GST_TIME_ARGS (min), GST_TIME_ARGS (max)); + + gst_query_set_latency (query, live, min, max); + } + gst_object_unref (peer); + } + break; + } default: res = gst_pad_query_default (pad, query); break; @@ -728,10 +773,12 @@ gst_pitch_sink_event (GstPad * pad, GstEvent * event) case GST_EVENT_FLUSH_STOP: gst_pitch_flush_buffer (pitch, FALSE); pitch->priv->st->clear (); + pitch->min_latency = pitch->max_latency = 0; break; case GST_EVENT_EOS: gst_pitch_flush_buffer (pitch, TRUE); pitch->priv->st->clear (); + pitch->min_latency = pitch->max_latency = 0; break; case GST_EVENT_NEWSEGMENT: if (!gst_pitch_process_segment (pitch, &event)) { @@ -742,6 +789,7 @@ gst_pitch_sink_event (GstPad * pad, GstEvent * event) event = NULL; } pitch->priv->st->clear (); + pitch->min_latency = pitch->max_latency = 0; break; default: break; @@ -755,17 +803,41 @@ gst_pitch_sink_event (GstPad * pad, GstEvent * event) return res; } +static void +gst_pitch_update_latency (GstPitch * pitch, GstClockTime timestamp) +{ + GstClockTimeDiff current_latency, min_latency, max_latency; + + current_latency = + timestamp / pitch->priv->stream_time_ratio - pitch->next_buffer_time; + + min_latency = MIN (pitch->min_latency, current_latency); + max_latency = MAX (pitch->max_latency, current_latency); + + if (pitch->min_latency != min_latency || pitch->max_latency != max_latency) { + pitch->min_latency = min_latency; + pitch->max_latency = max_latency; + + gst_pad_push_event (pitch->sinkpad, gst_event_new_latency (max_latency)); + gst_element_post_message (GST_ELEMENT (pitch), + gst_message_new_latency (GST_OBJECT (pitch))); + } +} + static GstFlowReturn gst_pitch_chain (GstPad * pad, GstBuffer * buffer) { GstPitch *pitch; GstPitchPrivate *priv; + GstClockTime timestamp; pitch = GST_PITCH (GST_PAD_PARENT (pad)); priv = GST_PITCH_GET_PRIVATE (pitch); gst_object_sync_values (G_OBJECT (pitch), pitch->next_buffer_time); + timestamp = GST_BUFFER_TIMESTAMP (buffer); + /* push the received samples on the soundtouch buffer */ GST_LOG_OBJECT (pitch, "incoming buffer (%d samples)", (gint) (GST_BUFFER_SIZE (buffer) / pitch->sample_size)); @@ -793,6 +865,10 @@ gst_pitch_chain (GstPad * pad, GstBuffer * buffer) GST_BUFFER_SIZE (buffer) / pitch->sample_size); gst_buffer_unref (buffer); + /* Calculate latency */ + + gst_pitch_update_latency (pitch, timestamp); + /* and try to extract some samples from the soundtouch buffer */ if (!priv->st->isEmpty ()) { GstBuffer *out_buffer; @@ -818,6 +894,7 @@ gst_pitch_change_state (GstElement * element, GstStateChange transition) pitch->next_buffer_time = 0; pitch->next_buffer_offset = 0; pitch->priv->st->clear (); + pitch->min_latency = pitch->max_latency = 0; break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; diff --git a/ext/soundtouch/gstpitch.hh b/ext/soundtouch/gstpitch.hh index 24d42ba2..32cb04dd 100644 --- a/ext/soundtouch/gstpitch.hh +++ b/ext/soundtouch/gstpitch.hh @@ -67,12 +67,13 @@ struct _GstPitch gint samplerate; /* samplerate */ gint channels; /* number of audio channels */ gsize sample_size; /* number of bytes for a single sample */ - GstClockTime sample_duration; /* time for 1 sample */ /* stream tracking */ GstClockTime next_buffer_time; gint64 next_buffer_offset; + GstClockTimeDiff min_latency, max_latency; + GstPitchPrivate *priv; }; -- cgit v1.2.1