gstreamer/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c

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/* GStreamer
* Copyright (C) 2017 Sebastian Dröge <sebastian@centricular.com>
*
* gstaudiostreamalign.h:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstaudiostreamalign.h"
/**
* SECTION:gstaudiostreamalign
* @title: GstAudioStreamAlign
* @short_description: Helper object for tracking audio stream alignment and discontinuities
*
* #GstAudioStreamAlign provides a helper object that helps tracking audio
* stream alignment and discontinuities, and detects discontinuities if
* possible.
*
* See gst_audio_stream_align_new() for a description of its parameters and
* gst_audio_stream_align_process() for the details of the processing.
*/
G_DEFINE_BOXED_TYPE (GstAudioStreamAlign, gst_audio_stream_align,
(GBoxedCopyFunc) gst_audio_stream_align_copy,
(GBoxedFreeFunc) gst_audio_stream_align_free);
struct _GstAudioStreamAlign
{
gint rate;
GstClockTime alignment_threshold;
GstClockTime discont_wait;
/* counter to keep track of timestamps */
guint64 next_offset;
GstClockTime timestamp_at_discont;
guint64 samples_since_discont;
/* Last time we noticed a discont */
GstClockTime discont_time;
};
/**
* gst_audio_stream_align_new:
* @rate: a sample rate
* @alignment_threshold: a alignment threshold in nanoseconds
* @discont_wait: discont wait in nanoseconds
*
* Allocate a new #GstAudioStreamAlign with the given configuration. All
* processing happens according to sample rate @rate, until
* gst_audio_stream_align_set_rate() is called with a new @rate.
* A negative rate can be used for reverse playback.
*
* @alignment_threshold gives the tolerance in nanoseconds after which a
* timestamp difference is considered a discontinuity. Once detected,
* @discont_wait nanoseconds have to pass without going below the threshold
* again until the output buffer is marked as a discontinuity. These can later
* be re-configured with gst_audio_stream_align_set_alignment_threshold() and
* gst_audio_stream_align_set_discont_wait().
*
* Returns: a new #GstAudioStreamAlign. free with gst_audio_stream_align_free().
*
* Since: 1.14
*/
GstAudioStreamAlign *
gst_audio_stream_align_new (gint rate, GstClockTime alignment_threshold,
GstClockTime discont_wait)
{
GstAudioStreamAlign *align;
g_return_val_if_fail (rate != 0, NULL);
g_return_val_if_fail (GST_CLOCK_TIME_IS_VALID (alignment_threshold), NULL);
g_return_val_if_fail (GST_CLOCK_TIME_IS_VALID (discont_wait), NULL);
align = g_new0 (GstAudioStreamAlign, 1);
align->rate = rate;
align->alignment_threshold = alignment_threshold;
align->discont_wait = discont_wait;
align->timestamp_at_discont = GST_CLOCK_TIME_NONE;
align->samples_since_discont = 0;
gst_audio_stream_align_mark_discont (align);
return align;
}
/**
* gst_audio_stream_align_copy:
* @align: a #GstAudioStreamAlign
*
* Copy a GstAudioStreamAlign structure.
*
* Returns: a new #GstAudioStreamAlign. free with gst_audio_stream_align_free.
*
* Since: 1.14
*/
GstAudioStreamAlign *
gst_audio_stream_align_copy (const GstAudioStreamAlign * align)
{
GstAudioStreamAlign *copy;
g_return_val_if_fail (align != NULL, NULL);
copy = g_new0 (GstAudioStreamAlign, 1);
*copy = *align;
return copy;
}
/**
* gst_audio_stream_align_free:
* @align: a #GstAudioStreamAlign
*
* Free a GstAudioStreamAlign structure previously allocated with gst_audio_stream_align_new()
* or gst_audio_stream_align_copy().
*
* Since: 1.14
*/
void
gst_audio_stream_align_free (GstAudioStreamAlign * align)
{
g_return_if_fail (align != NULL);
g_free (align);
}
/**
* gst_audio_stream_align_set_rate:
* @align: a #GstAudioStreamAlign
* @rate: a new sample rate
*
* Sets @rate as new sample rate for the following processing. If the sample
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* rate differs this implicitly marks the next data as discontinuous.
*
* Since: 1.14
*/
void
gst_audio_stream_align_set_rate (GstAudioStreamAlign * align, gint rate)
{
g_return_if_fail (align != NULL);
g_return_if_fail (rate != 0);
if (align->rate == rate)
return;
align->rate = rate;
gst_audio_stream_align_mark_discont (align);
}
/**
* gst_audio_stream_align_get_rate:
* @align: a #GstAudioStreamAlign
*
* Gets the currently configured sample rate.
*
* Returns: The currently configured sample rate
*
* Since: 1.14
*/
gint
gst_audio_stream_align_get_rate (const GstAudioStreamAlign * align)
{
g_return_val_if_fail (align != NULL, 0);
return align->rate;
}
/**
* gst_audio_stream_align_set_alignment_threshold:
* @align: a #GstAudioStreamAlign
* @alignment_threshold: a new alignment threshold
*
* Sets @alignment_treshold as new alignment threshold for the following processing.
*
* Since: 1.14
*/
void
gst_audio_stream_align_set_alignment_threshold (GstAudioStreamAlign *
align, GstClockTime alignment_threshold)
{
g_return_if_fail (align != NULL);
g_return_if_fail (GST_CLOCK_TIME_IS_VALID (alignment_threshold));
align->alignment_threshold = alignment_threshold;
}
/**
* gst_audio_stream_align_get_alignment_threshold:
* @align: a #GstAudioStreamAlign
*
* Gets the currently configured alignment threshold.
*
* Returns: The currently configured alignment threshold
*
* Since: 1.14
*/
GstClockTime
gst_audio_stream_align_get_alignment_threshold (const GstAudioStreamAlign *
align)
{
g_return_val_if_fail (align != NULL, 0);
return align->alignment_threshold;
}
/**
* gst_audio_stream_align_set_discont_wait:
* @align: a #GstAudioStreamAlign
* @discont_wait: a new discont wait
*
* Sets @alignment_treshold as new discont wait for the following processing.
*
* Since: 1.14
*/
void
gst_audio_stream_align_set_discont_wait (GstAudioStreamAlign * align,
GstClockTime discont_wait)
{
g_return_if_fail (align != NULL);
g_return_if_fail (GST_CLOCK_TIME_IS_VALID (discont_wait));
align->discont_wait = discont_wait;
}
/**
* gst_audio_stream_align_get_discont_wait:
* @align: a #GstAudioStreamAlign
*
* Gets the currently configured discont wait.
*
* Returns: The currently configured discont wait
*
* Since: 1.14
*/
GstClockTime
gst_audio_stream_align_get_discont_wait (const GstAudioStreamAlign * align)
{
g_return_val_if_fail (align != NULL, 0);
return align->discont_wait;
}
/**
* gst_audio_stream_align_mark_discont:
* @align: a #GstAudioStreamAlign
*
* Marks the next buffer as discontinuous and resets timestamp tracking.
*
* Since: 1.14
*/
void
gst_audio_stream_align_mark_discont (GstAudioStreamAlign * align)
{
g_return_if_fail (align != NULL);
align->next_offset = -1;
align->discont_time = GST_CLOCK_TIME_NONE;
}
/**
* gst_audio_stream_align_get_timestamp_at_discont:
* @align: a #GstAudioStreamAlign
*
* Timestamp that was passed when a discontinuity was detected, i.e. the first
* timestamp after the discontinuity.
*
* Returns: The last timestamp at when a discontinuity was detected
*
* Since: 1.14
*/
GstClockTime
gst_audio_stream_align_get_timestamp_at_discont (const GstAudioStreamAlign *
align)
{
g_return_val_if_fail (align != NULL, GST_CLOCK_TIME_NONE);
return align->timestamp_at_discont;
}
/**
* gst_audio_stream_align_get_samples_since_discont:
* @align: a #GstAudioStreamAlign
*
* Returns the number of samples that were processed since the last
* discontinuity was detected.
*
* Returns: The number of samples processed since the last discontinuity.
*
* Since: 1.14
*/
guint64
gst_audio_stream_align_get_samples_since_discont (const GstAudioStreamAlign *
align)
{
g_return_val_if_fail (align != NULL, 0);
return align->samples_since_discont;
}
/**
* gst_audio_stream_align_process:
* @align: a #GstAudioStreamAlign
* @discont: if this data is considered to be discontinuous
* @timestamp: a #GstClockTime of the start of the data
* @n_samples: number of samples to process
* @out_timestamp: (out): output timestamp of the data
* @out_duration: (out): output duration of the data
* @out_sample_position: (out): output sample position of the start of the data
*
* Processes data with @timestamp and @n_samples, and returns the output
* timestamp, duration and sample position together with a boolean to signal
* whether a discontinuity was detected or not. All non-discontinuous data
* will have perfect timestamps and durations.
*
* A discontinuity is detected once the difference between the actual
* timestamp and the timestamp calculated from the sample count since the last
* discontinuity differs by more than the alignment threshold for a duration
* longer than discont wait.
*
* Note: In reverse playback, every buffer is considered discontinuous in the
* context of buffer flags because the last sample of the previous buffer is
* discontinuous with the first sample of the current one. However for this
* function they are only considered discontinuous in reverse playback if the
* first sample of the previous buffer is discontinuous with the last sample
* of the current one.
*
* Returns: %TRUE if a discontinuity was detected, %FALSE otherwise.
*
* Since: 1.14
*/
#define ABSDIFF(a, b) ((a) > (b) ? (a) - (b) : (b) - (a))
gboolean
gst_audio_stream_align_process (GstAudioStreamAlign * align,
gboolean discont, GstClockTime timestamp, guint n_samples,
GstClockTime * out_timestamp, GstClockTime * out_duration,
guint64 * out_sample_position)
{
GstClockTime start_time, end_time, duration;
guint64 start_offset, end_offset;
g_return_val_if_fail (align != NULL, FALSE);
start_time = timestamp;
start_offset =
gst_util_uint64_scale (start_time, ABS (align->rate), GST_SECOND);
end_offset = start_offset + n_samples;
end_time =
gst_util_uint64_scale_int (end_offset, GST_SECOND, ABS (align->rate));
duration = end_time - start_time;
if (align->next_offset == (guint64) - 1 || discont) {
discont = TRUE;
} else {
guint64 diff, max_sample_diff;
/* Check discont */
if (align->rate > 0) {
diff = ABSDIFF (start_offset, align->next_offset);
} else {
diff = ABSDIFF (end_offset, align->next_offset);
}
max_sample_diff =
gst_util_uint64_scale_int (align->alignment_threshold,
ABS (align->rate), GST_SECOND);
/* Discont! */
if (G_UNLIKELY (diff >= max_sample_diff)) {
if (align->discont_wait > 0) {
if (align->discont_time == GST_CLOCK_TIME_NONE) {
align->discont_time = align->rate > 0 ? start_time : end_time;
} else if ((align->rate > 0
&& ABSDIFF (start_time,
align->discont_time) >= align->discont_wait)
|| (align->rate < 0
&& ABSDIFF (end_time,
align->discont_time) >= align->discont_wait)) {
discont = TRUE;
align->discont_time = GST_CLOCK_TIME_NONE;
}
} else {
discont = TRUE;
}
} else if (G_UNLIKELY (align->discont_time != GST_CLOCK_TIME_NONE)) {
/* we have had a discont, but are now back on track! */
align->discont_time = GST_CLOCK_TIME_NONE;
}
}
if (discont) {
/* Have discont, need resync and use the capture timestamps */
if (align->next_offset != (guint64) - 1)
GST_INFO ("Have discont. Expected %"
G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
align->next_offset, start_offset);
align->next_offset = align->rate > 0 ? end_offset : start_offset;
align->timestamp_at_discont = start_time;
align->samples_since_discont = 0;
/* Got a discont and adjusted, reset the discont_time marker */
align->discont_time = GST_CLOCK_TIME_NONE;
} else {
/* No discont, just keep counting */
if (align->rate > 0) {
timestamp =
gst_util_uint64_scale (align->next_offset, GST_SECOND,
ABS (align->rate));
start_offset = align->next_offset;
align->next_offset += n_samples;
duration =
gst_util_uint64_scale (align->next_offset, GST_SECOND,
ABS (align->rate)) - timestamp;
} else {
guint64 old_offset = align->next_offset;
if (align->next_offset > n_samples)
align->next_offset -= n_samples;
else
align->next_offset = 0;
start_offset = align->next_offset;
timestamp =
gst_util_uint64_scale (align->next_offset, GST_SECOND,
ABS (align->rate));
duration =
gst_util_uint64_scale (old_offset, GST_SECOND,
ABS (align->rate)) - timestamp;
}
}
align->samples_since_discont += n_samples;
if (out_timestamp)
*out_timestamp = timestamp;
if (out_duration)
*out_duration = duration;
if (out_sample_position)
*out_sample_position = start_offset;
return discont;
}
#undef ABSDIFF