mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-23 10:11:08 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c
Carlos Rafael Giani b65eab915a audioringbuffer: Add support for DSD data 
Code is partially based on the DSD of Robert Tiemann <rtie@gmx.de>:
https://gitlab.freedesktop.org/rtiemann/gstreamer/-/tree/dsd

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3901>
2023-06-23 01:27:03 +00:00

2172 lines
58 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
*
* 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.
*/
/**
* SECTION:gstaudioringbuffer
* @title: GstAudioRingBuffer
* @short_description: Base class for audio ringbuffer implementations
* @see_also: #GstAudioBaseSink, #GstAudioSink
*
* This object is the base class for audio ringbuffers used by the base
* audio source and sink classes.
*
* The ringbuffer abstracts a circular buffer of data. One reader and
* one writer can operate on the data from different threads in a lockfree
* manner. The base class is sufficiently flexible to be used as an
* abstraction for DMA based ringbuffers as well as a pure software
* implementations.
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstdsd.h>
#include "gstaudioringbuffer.h"
GST_DEBUG_CATEGORY_STATIC (gst_audio_ring_buffer_debug);
#define GST_CAT_DEFAULT gst_audio_ring_buffer_debug
static void gst_audio_ring_buffer_dispose (GObject * object);
static void gst_audio_ring_buffer_finalize (GObject * object);
static gboolean gst_audio_ring_buffer_pause_unlocked (GstAudioRingBuffer * buf);
static void default_clear_all (GstAudioRingBuffer * buf);
static guint default_commit (GstAudioRingBuffer * buf, guint64 * sample,
guint8 * data, gint in_samples, gint out_samples, gint * accum);
/* ringbuffer abstract base class */
G_DEFINE_ABSTRACT_TYPE (GstAudioRingBuffer, gst_audio_ring_buffer,
GST_TYPE_OBJECT);
static void
gst_audio_ring_buffer_class_init (GstAudioRingBufferClass * klass)
{
GObjectClass *gobject_class;
GstAudioRingBufferClass *gstaudioringbuffer_class;
gobject_class = (GObjectClass *) klass;
gstaudioringbuffer_class = (GstAudioRingBufferClass *) klass;
GST_DEBUG_CATEGORY_INIT (gst_audio_ring_buffer_debug, "ringbuffer", 0,
"ringbuffer class");
gobject_class->dispose = gst_audio_ring_buffer_dispose;
gobject_class->finalize = gst_audio_ring_buffer_finalize;
gstaudioringbuffer_class->clear_all = GST_DEBUG_FUNCPTR (default_clear_all);
gstaudioringbuffer_class->commit = GST_DEBUG_FUNCPTR (default_commit);
}
static void
gst_audio_ring_buffer_init (GstAudioRingBuffer * ringbuffer)
{
ringbuffer->open = FALSE;
ringbuffer->acquired = FALSE;
ringbuffer->state = GST_AUDIO_RING_BUFFER_STATE_STOPPED;
g_cond_init (&ringbuffer->cond);
ringbuffer->waiting = 0;
ringbuffer->empty_seg = NULL;
ringbuffer->flushing = TRUE;
ringbuffer->segbase = 0;
ringbuffer->segdone = 0;
}
static void
gst_audio_ring_buffer_dispose (GObject * object)
{
GstAudioRingBuffer *ringbuffer = GST_AUDIO_RING_BUFFER (object);
gst_caps_replace (&ringbuffer->spec.caps, NULL);
G_OBJECT_CLASS (gst_audio_ring_buffer_parent_class)->dispose (G_OBJECT
(ringbuffer));
}
static void
gst_audio_ring_buffer_finalize (GObject * object)
{
GstAudioRingBuffer *ringbuffer = GST_AUDIO_RING_BUFFER (object);
g_cond_clear (&ringbuffer->cond);
g_free (ringbuffer->empty_seg);
if (ringbuffer->cb_data_notify != NULL)
ringbuffer->cb_data_notify (ringbuffer->cb_data);
G_OBJECT_CLASS (gst_audio_ring_buffer_parent_class)->finalize (G_OBJECT
(ringbuffer));
}
#ifndef GST_DISABLE_GST_DEBUG
static const gchar *format_type_names[] = {
"raw",
"mu law",
"a law",
"ima adpcm",
"mpeg",
"gsm",
"iec958",
"ac3",
"eac3",
"dts",
"aac mpeg2",
"aac mpeg4",
"aac mpeg2 raw",
"aac mpeg4 raw",
"flac"
};
#endif
/**
* gst_audio_ring_buffer_debug_spec_caps:
* @spec: the spec to debug
*
* Print debug info about the parsed caps in @spec to the debug log.
*/
void
gst_audio_ring_buffer_debug_spec_caps (GstAudioRingBufferSpec * spec)
{
#if 0
gint i, bytes;
#endif
GST_DEBUG ("spec caps: %p %" GST_PTR_FORMAT, spec->caps, spec->caps);
GST_DEBUG ("parsed caps: type: %d, '%s'", spec->type,
format_type_names[spec->type]);
#if 0
GST_DEBUG ("parsed caps: width: %d", spec->width);
GST_DEBUG ("parsed caps: sign: %d", spec->sign);
GST_DEBUG ("parsed caps: bigend: %d", spec->bigend);
GST_DEBUG ("parsed caps: rate: %d", spec->rate);
GST_DEBUG ("parsed caps: channels: %d", spec->channels);
GST_DEBUG ("parsed caps: sample bytes: %d", spec->bytes_per_sample);
bytes = (spec->width >> 3) * spec->channels;
for (i = 0; i < bytes; i++) {
GST_DEBUG ("silence byte %d: %02x", i, spec->silence_sample[i]);
}
#endif
}
/**
* gst_audio_ring_buffer_debug_spec_buff:
* @spec: the spec to debug
*
* Print debug info about the buffer sized in @spec to the debug log.
*/
void
gst_audio_ring_buffer_debug_spec_buff (GstAudioRingBufferSpec * spec)
{
gint bpf = GST_AUDIO_INFO_BPF (&spec->info);
GST_DEBUG ("acquire ringbuffer: buffer time: %" G_GINT64_FORMAT " usec",
spec->buffer_time);
GST_DEBUG ("acquire ringbuffer: latency time: %" G_GINT64_FORMAT " usec",
spec->latency_time);
GST_DEBUG ("acquire ringbuffer: total segments: %d", spec->segtotal);
GST_DEBUG ("acquire ringbuffer: latency segments: %d", spec->seglatency);
GST_DEBUG ("acquire ringbuffer: segment size: %d bytes = %d samples",
spec->segsize, (bpf != 0) ? (spec->segsize / bpf) : -1);
GST_DEBUG ("acquire ringbuffer: buffer size: %d bytes = %d samples",
spec->segsize * spec->segtotal,
(bpf != 0) ? (spec->segsize * spec->segtotal / bpf) : -1);
}
/**
* gst_audio_ring_buffer_parse_caps:
* @spec: a spec
* @caps: a #GstCaps
*
* Parse @caps into @spec.
*
* Returns: TRUE if the caps could be parsed.
*/
gboolean
gst_audio_ring_buffer_parse_caps (GstAudioRingBufferSpec * spec, GstCaps * caps)
{
const gchar *mimetype;
GstStructure *structure;
gint i;
GstAudioInfo info;
structure = gst_caps_get_structure (caps, 0);
gst_audio_info_init (&info);
/* we have to differentiate between int and float formats */
mimetype = gst_structure_get_name (structure);
if (g_str_equal (mimetype, "audio/x-raw")) {
if (!gst_audio_info_from_caps (&info, caps))
goto parse_error;
spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW;
} else if (g_str_equal (mimetype, "audio/x-alaw")) {
/* extract the needed information from the cap */
if (!(gst_structure_get_int (structure, "rate", &info.rate) &&
gst_structure_get_int (structure, "channels", &info.channels)))
goto parse_error;
if (!(gst_audio_channel_positions_from_mask (info.channels, 0,
info.position)))
goto parse_error;
spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW;
info.bpf = info.channels;
} else if (g_str_equal (mimetype, "audio/x-mulaw")) {
/* extract the needed information from the cap */
if (!(gst_structure_get_int (structure, "rate", &info.rate) &&
gst_structure_get_int (structure, "channels", &info.channels)))
goto parse_error;
if (!(gst_audio_channel_positions_from_mask (info.channels, 0,
info.position)))
goto parse_error;
spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW;
info.bpf = info.channels;
} else if (g_str_equal (mimetype, "audio/x-iec958")) {
/* extract the needed information from the cap */
if (!(gst_structure_get_int (structure, "rate", &info.rate)))
goto parse_error;
spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958;
info.bpf = 4;
} else if (g_str_equal (mimetype, "audio/x-ac3")) {
/* extract the needed information from the cap */
if (!(gst_structure_get_int (structure, "rate", &info.rate)))
goto parse_error;
gst_structure_get_int (structure, "channels", &info.channels);
spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3;
info.bpf = 4;
} else if (g_str_equal (mimetype, "audio/x-eac3")) {
/* extract the needed information from the cap */
if (!(gst_structure_get_int (structure, "rate", &info.rate)))
goto parse_error;
gst_structure_get_int (structure, "channels", &info.channels);
spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3;
info.bpf = 16;
} else if (g_str_equal (mimetype, "audio/x-dts")) {
/* extract the needed information from the cap */
if (!(gst_structure_get_int (structure, "rate", &info.rate)))
goto parse_error;
gst_structure_get_int (structure, "channels", &info.channels);
spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS;
info.bpf = 4;
} else if (g_str_equal (mimetype, "audio/mpeg") &&
gst_structure_get_int (structure, "mpegaudioversion", &i) &&
(i == 1 || i == 2 || i == 3)) {
/* Now we know this is MPEG-1, MPEG-2 or MPEG-2.5 (non AAC) */
/* extract the needed information from the cap */
if (!(gst_structure_get_int (structure, "rate", &info.rate)))
goto parse_error;
gst_structure_get_int (structure, "channels", &info.channels);
spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG;
info.bpf = 1;
} else if (g_str_equal (mimetype, "audio/mpeg") &&
gst_structure_get_int (structure, "mpegversion", &i) &&
(i == 2 || i == 4) &&
(!g_strcmp0 (gst_structure_get_string (structure, "stream-format"),
"adts")
|| !g_strcmp0 (gst_structure_get_string (structure, "stream-format"),
"raw"))) {
/* MPEG-2 AAC or MPEG-4 AAC */
if (!(gst_structure_get_int (structure, "rate", &info.rate)))
goto parse_error;
gst_structure_get_int (structure, "channels", &info.channels);
if (!g_strcmp0 (gst_structure_get_string (structure, "stream-format"),
"adts"))
spec->type = (i == 2) ? GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC :
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC;
else
spec->type = (i == 2) ?
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW :
GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW;
info.bpf = 1;
} else if (g_str_equal (mimetype, "audio/x-flac")) {
/* extract the needed information from the cap */
if (!(gst_structure_get_int (structure, "rate", &info.rate)))
goto parse_error;
gst_structure_get_int (structure, "channels", &info.channels);
spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC;
info.bpf = 1;
} else if (g_str_equal (mimetype, GST_DSD_MEDIA_TYPE)) {
/* Notes about what the "rate" means in DSD:
*
* In DSD, "sample formats" don't actually exist. There is only the DSD bit;
* this is what could be considered the closest equivalent to a "sample format".
* But since it is impractical to deal with individual bits in software, the
* bits are typically grouped into words (8/16/32 bit words). These are the
* DSDU8, DSDU16LE etc. "grouping formats".
*
* The "rate" in DSD information refers to the number of DSD _bytes_ per second
* (not bits per second, because, as said, per-bit handling in software does
* not usually make sense). The way the GstAudioRingBuffer works however requires
* the rate to be interpreted as the number of DSD _words_ per minute. This is
* in part because that's how ALSA uses the rate.
*
* If the word format is DSDU8, then there's no difference to just using the
* original byte rate. But if for example it is DSDU16LE, then the ringbuffer's
* rate needs to be half of the rate from GstDsdInfo. For this reason, it is
* essential to divide the rate from the DSD info by the word length (in bytes).
*
* Furthermore, the BPF is set to the stride (= format width * num channels).
* The GstAudioRingBuffer can only handle interleaved DSD. This means that
* there is a "stride", that is, the DSD word of channel #1 is stored first,
* followed by the DSD word of channel #2 etc. and then again we get a DSD
* word from channel #1, and so forth. This is similar to how interleaved
* PCM works. The stride is then the size (in bytes) of the DSD words for
* each channel that are played at the same time. Using this as the BPF is
* very important. Otherweise, timestamp and duration figures can be off,
* the segment sizes may not be an integer multiple of the DSD stride, etc.
*/
GstDsdInfo dsd_info;
guint format_width;
if (!gst_dsd_info_from_caps (&dsd_info, caps))
goto parse_error;
format_width = gst_dsd_format_get_width (dsd_info.format);
info.rate = dsd_info.rate / format_width;
info.channels = dsd_info.channels;
info.bpf = format_width * dsd_info.channels;
GST_INFO ("using DSD word rate %d instead of DSD byte rate %d "
"for ringbuffer", info.rate, dsd_info.rate);
memcpy (info.position, dsd_info.positions,
sizeof (GstAudioChannelPosition) * dsd_info.channels);
GST_AUDIO_RING_BUFFER_SPEC_DSD_FORMAT (spec) =
GST_DSD_INFO_FORMAT (&dsd_info);
spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DSD;
} else {
goto parse_error;
}
gst_caps_replace (&spec->caps, caps);
g_return_val_if_fail (spec->latency_time != 0, FALSE);
/* calculate suggested segsize and segtotal. segsize should be one unit
* of 'latency_time' samples, scaling for the fact that latency_time is
* currently stored in microseconds (FIXME: in 0.11) */
spec->segsize = gst_util_uint64_scale (info.rate * info.bpf,
spec->latency_time, GST_SECOND / GST_USECOND);
/* Round to an integer number of samples */
spec->segsize -= spec->segsize % info.bpf;
spec->segtotal = spec->buffer_time / spec->latency_time;
/* leave the latency undefined now, implementations can change it but if it's
* not changed, we assume the same value as segtotal */
spec->seglatency = -1;
spec->info = info;
gst_audio_ring_buffer_debug_spec_caps (spec);
gst_audio_ring_buffer_debug_spec_buff (spec);
return TRUE;
/* ERRORS */
parse_error:
{
GST_DEBUG ("could not parse caps");
return FALSE;
}
}
/**
* gst_audio_ring_buffer_convert:
* @buf: the #GstAudioRingBuffer
* @src_fmt: the source format
* @src_val: the source value
* @dest_fmt: the destination format
* @dest_val: (out): a location to store the converted value
*
* Convert @src_val in @src_fmt to the equivalent value in @dest_fmt. The result
* will be put in @dest_val.
*
* Returns: TRUE if the conversion succeeded.
*/
gboolean
gst_audio_ring_buffer_convert (GstAudioRingBuffer * buf,
GstFormat src_fmt, gint64 src_val, GstFormat dest_fmt, gint64 * dest_val)
{
gboolean res;
GST_OBJECT_LOCK (buf);
res =
gst_audio_info_convert (&buf->spec.info, src_fmt, src_val, dest_fmt,
dest_val);
GST_OBJECT_UNLOCK (buf);
return res;
}
/**
* gst_audio_ring_buffer_set_callback: (skip)
* @buf: the #GstAudioRingBuffer to set the callback on
* @cb: (allow-none): the callback to set
* @user_data: user data passed to the callback
*
* Sets the given callback function on the buffer. This function
* will be called every time a segment has been written to a device.
*
* MT safe.
*/
void
gst_audio_ring_buffer_set_callback (GstAudioRingBuffer * buf,
GstAudioRingBufferCallback cb, gpointer user_data)
{
gst_audio_ring_buffer_set_callback_full (buf, cb, user_data, NULL);
}
/**
* gst_audio_ring_buffer_set_callback_full: (rename-to gst_audio_ring_buffer_set_callback)
* @buf: the #GstAudioRingBuffer to set the callback on
* @cb: (allow-none): the callback to set
* @user_data: user data passed to the callback
* @notify: function to be called when @user_data is no longer needed
*
* Sets the given callback function on the buffer. This function
* will be called every time a segment has been written to a device.
*
* MT safe.
*
* Since: 1.12
*/
void
gst_audio_ring_buffer_set_callback_full (GstAudioRingBuffer * buf,
GstAudioRingBufferCallback cb, gpointer user_data, GDestroyNotify notify)
{
gpointer old_data = NULL;
GDestroyNotify old_notify;
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
GST_OBJECT_LOCK (buf);
old_notify = buf->cb_data_notify;
old_data = buf->cb_data;
buf->callback = cb;
buf->cb_data = user_data;
buf->cb_data_notify = notify;
GST_OBJECT_UNLOCK (buf);
if (old_notify) {
old_notify (old_data);
}
}
/**
* gst_audio_ring_buffer_open_device:
* @buf: the #GstAudioRingBuffer
*
* Open the audio device associated with the ring buffer. Does not perform any
* setup on the device. You must open the device before acquiring the ring
* buffer.
*
* Returns: TRUE if the device could be opened, FALSE on error.
*
* MT safe.
*/
gboolean
gst_audio_ring_buffer_open_device (GstAudioRingBuffer * buf)
{
gboolean res = TRUE;
GstAudioRingBufferClass *rclass;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
GST_DEBUG_OBJECT (buf, "opening device");
GST_OBJECT_LOCK (buf);
if (G_UNLIKELY (buf->open))
goto was_opened;
buf->open = TRUE;
/* if this fails, something is wrong in this file */
g_assert (!buf->acquired);
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
if (G_LIKELY (rclass->open_device))
res = rclass->open_device (buf);
if (G_UNLIKELY (!res))
goto open_failed;
GST_DEBUG_OBJECT (buf, "opened device");
done:
GST_OBJECT_UNLOCK (buf);
return res;
/* ERRORS */
was_opened:
{
GST_DEBUG_OBJECT (buf, "Device for ring buffer already open");
g_warning ("Device for ring buffer %p already open, fix your code", buf);
res = TRUE;
goto done;
}
open_failed:
{
buf->open = FALSE;
GST_DEBUG_OBJECT (buf, "failed opening device");
goto done;
}
}
/**
* gst_audio_ring_buffer_close_device:
* @buf: the #GstAudioRingBuffer
*
* Close the audio device associated with the ring buffer. The ring buffer
* should already have been released via gst_audio_ring_buffer_release().
*
* Returns: TRUE if the device could be closed, FALSE on error.
*
* MT safe.
*/
gboolean
gst_audio_ring_buffer_close_device (GstAudioRingBuffer * buf)
{
gboolean res = TRUE;
GstAudioRingBufferClass *rclass;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
GST_DEBUG_OBJECT (buf, "closing device");
GST_OBJECT_LOCK (buf);
if (G_UNLIKELY (!buf->open))
goto was_closed;
if (G_UNLIKELY (buf->acquired))
goto was_acquired;
buf->open = FALSE;
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
if (G_LIKELY (rclass->close_device))
res = rclass->close_device (buf);
if (G_UNLIKELY (!res))
goto close_error;
GST_DEBUG_OBJECT (buf, "closed device");
done:
GST_OBJECT_UNLOCK (buf);
return res;
/* ERRORS */
was_closed:
{
GST_DEBUG_OBJECT (buf, "Device for ring buffer already closed");
g_warning ("Device for ring buffer %p already closed, fix your code", buf);
res = TRUE;
goto done;
}
was_acquired:
{
GST_DEBUG_OBJECT (buf, "Resources for ring buffer still acquired");
g_critical ("Resources for ring buffer %p still acquired", buf);
res = FALSE;
goto done;
}
close_error:
{
buf->open = TRUE;
GST_DEBUG_OBJECT (buf, "error closing device");
goto done;
}
}
/**
* gst_audio_ring_buffer_device_is_open:
* @buf: the #GstAudioRingBuffer
*
* Checks the status of the device associated with the ring buffer.
*
* Returns: TRUE if the device was open, FALSE if it was closed.
*
* MT safe.
*/
gboolean
gst_audio_ring_buffer_device_is_open (GstAudioRingBuffer * buf)
{
gboolean res = TRUE;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
GST_OBJECT_LOCK (buf);
res = buf->open;
GST_OBJECT_UNLOCK (buf);
return res;
}
/**
* gst_audio_ring_buffer_acquire:
* @buf: the #GstAudioRingBuffer to acquire
* @spec: the specs of the buffer
*
* Allocate the resources for the ringbuffer. This function fills
* in the data pointer of the ring buffer with a valid #GstBuffer
* to which samples can be written.
*
* Returns: TRUE if the device could be acquired, FALSE on error.
*
* MT safe.
*/
gboolean
gst_audio_ring_buffer_acquire (GstAudioRingBuffer * buf,
GstAudioRingBufferSpec * spec)
{
gboolean res = FALSE;
GstAudioRingBufferClass *rclass;
gint segsize, bpf, i;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
GST_DEBUG_OBJECT (buf, "acquiring device %p", buf);
GST_OBJECT_LOCK (buf);
if (G_UNLIKELY (!buf->open))
goto not_opened;
if (G_UNLIKELY (buf->acquired))
goto was_acquired;
buf->acquired = TRUE;
buf->need_reorder = FALSE;
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
if (G_LIKELY (rclass->acquire))
res = rclass->acquire (buf, spec);
/* Only reorder for raw audio */
buf->need_reorder = (buf->need_reorder
&& buf->spec.type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW);
if (G_UNLIKELY (!res))
goto acquire_failed;
GST_INFO_OBJECT (buf, "Allocating an array for %d timestamps",
spec->segtotal);
buf->timestamps = g_new0 (GstClockTime, spec->segtotal);
/* initialize array with invalid timestamps */
for (i = 0; i < spec->segtotal; i++) {
buf->timestamps[i] = GST_CLOCK_TIME_NONE;
}
if (G_UNLIKELY ((bpf = buf->spec.info.bpf) == 0))
goto invalid_bpf;
/* if the seglatency was overwritten with something else than -1, use it, else
* assume segtotal as the latency */
if (buf->spec.seglatency == -1)
buf->spec.seglatency = buf->spec.segtotal;
segsize = buf->spec.segsize;
buf->samples_per_seg = segsize / bpf;
/* create an empty segment */
g_free (buf->empty_seg);
buf->empty_seg = g_malloc (segsize);
switch (buf->spec.type) {
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:
gst_audio_format_info_fill_silence (buf->spec.info.finfo, buf->empty_seg,
segsize);
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DSD:
memset (buf->empty_seg, GST_DSD_SILENCE_PATTERN_BYTE, segsize);
break;
default:
/* FIXME, non-raw formats get 0 as the empty sample */
memset (buf->empty_seg, 0, segsize);
}
GST_DEBUG_OBJECT (buf, "acquired device");
done:
GST_OBJECT_UNLOCK (buf);
return res;
/* ERRORS */
not_opened:
{
GST_DEBUG_OBJECT (buf, "device not opened");
g_critical ("Device for %p not opened", buf);
res = FALSE;
goto done;
}
was_acquired:
{
res = TRUE;
GST_DEBUG_OBJECT (buf, "device was acquired");
goto done;
}
acquire_failed:
{
buf->acquired = FALSE;
GST_DEBUG_OBJECT (buf, "failed to acquire device");
goto done;
}
invalid_bpf:
{
g_warning
("invalid bytes_per_frame from acquire ringbuffer %p, fix the element",
buf);
buf->acquired = FALSE;
res = FALSE;
goto done;
}
}
/**
* gst_audio_ring_buffer_release:
* @buf: the #GstAudioRingBuffer to release
*
* Free the resources of the ringbuffer.
*
* Returns: TRUE if the device could be released, FALSE on error.
*
* MT safe.
*/
gboolean
gst_audio_ring_buffer_release (GstAudioRingBuffer * buf)
{
gboolean res = FALSE;
GstAudioRingBufferClass *rclass;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
GST_DEBUG_OBJECT (buf, "releasing device");
gst_audio_ring_buffer_stop (buf);
GST_OBJECT_LOCK (buf);
if (G_LIKELY (buf->timestamps)) {
GST_INFO_OBJECT (buf, "Freeing timestamp buffer, %d entries",
buf->spec.segtotal);
g_free (buf->timestamps);
buf->timestamps = NULL;
}
if (G_UNLIKELY (!buf->acquired))
goto was_released;
buf->acquired = FALSE;
/* if this fails, something is wrong in this file */
g_assert (buf->open);
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
if (G_LIKELY (rclass->release))
res = rclass->release (buf);
/* signal any waiters */
GST_DEBUG_OBJECT (buf, "signal waiter");
GST_AUDIO_RING_BUFFER_SIGNAL (buf);
if (G_UNLIKELY (!res))
goto release_failed;
g_atomic_int_set (&buf->segdone, 0);
buf->segbase = 0;
g_free (buf->empty_seg);
buf->empty_seg = NULL;
gst_caps_replace (&buf->spec.caps, NULL);
gst_audio_info_init (&buf->spec.info);
GST_DEBUG_OBJECT (buf, "released device");
done:
GST_OBJECT_UNLOCK (buf);
return res;
/* ERRORS */
was_released:
{
res = TRUE;
GST_DEBUG_OBJECT (buf, "device was released");
goto done;
}
release_failed:
{
buf->acquired = TRUE;
GST_DEBUG_OBJECT (buf, "failed to release device");
goto done;
}
}
/**
* gst_audio_ring_buffer_is_acquired:
* @buf: the #GstAudioRingBuffer to check
*
* Check if the ringbuffer is acquired and ready to use.
*
* Returns: TRUE if the ringbuffer is acquired, FALSE on error.
*
* MT safe.
*/
gboolean
gst_audio_ring_buffer_is_acquired (GstAudioRingBuffer * buf)
{
gboolean res;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
GST_OBJECT_LOCK (buf);
res = buf->acquired;
GST_OBJECT_UNLOCK (buf);
return res;
}
/**
* gst_audio_ring_buffer_activate:
* @buf: the #GstAudioRingBuffer to activate
* @active: the new mode
*
* Activate @buf to start or stop pulling data.
*
* MT safe.
*
* Returns: TRUE if the device could be activated in the requested mode,
* FALSE on error.
*/
gboolean
gst_audio_ring_buffer_activate (GstAudioRingBuffer * buf, gboolean active)
{
gboolean res = FALSE;
GstAudioRingBufferClass *rclass;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
GST_DEBUG_OBJECT (buf, "activate device");
GST_OBJECT_LOCK (buf);
if (G_UNLIKELY (active && !buf->acquired))
goto not_acquired;
if (G_UNLIKELY (buf->active == active))
goto was_active;
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
/* if there is no activate function we assume it was started/released
* in the acquire method */
if (G_LIKELY (rclass->activate))
res = rclass->activate (buf, active);
else
res = TRUE;
if (G_UNLIKELY (!res))
goto activate_failed;
buf->active = active;
done:
GST_OBJECT_UNLOCK (buf);
return res;
/* ERRORS */
not_acquired:
{
GST_DEBUG_OBJECT (buf, "device not acquired");
g_critical ("Device for %p not acquired", buf);
res = FALSE;
goto done;
}
was_active:
{
res = TRUE;
GST_DEBUG_OBJECT (buf, "device was active in mode %d", active);
goto done;
}
activate_failed:
{
GST_DEBUG_OBJECT (buf, "failed to activate device");
goto done;
}
}
/**
* gst_audio_ring_buffer_is_active:
* @buf: the #GstAudioRingBuffer
*
* Check if @buf is activated.
*
* MT safe.
*
* Returns: TRUE if the device is active.
*/
gboolean
gst_audio_ring_buffer_is_active (GstAudioRingBuffer * buf)
{
gboolean res;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
GST_OBJECT_LOCK (buf);
res = buf->active;
GST_OBJECT_UNLOCK (buf);
return res;
}
/**
* gst_audio_ring_buffer_set_flushing:
* @buf: the #GstAudioRingBuffer to flush
* @flushing: the new mode
*
* Set the ringbuffer to flushing mode or normal mode.
*
* MT safe.
*/
void
gst_audio_ring_buffer_set_flushing (GstAudioRingBuffer * buf, gboolean flushing)
{
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
GST_OBJECT_LOCK (buf);
buf->flushing = flushing;
if (flushing) {
gst_audio_ring_buffer_pause_unlocked (buf);
} else {
gst_audio_ring_buffer_clear_all (buf);
}
GST_OBJECT_UNLOCK (buf);
}
/**
* gst_audio_ring_buffer_is_flushing:
* @buf: the #GstAudioRingBuffer
*
* Check if @buf is flushing.
*
* MT safe.
*
* Returns: TRUE if the device is flushing.
*/
gboolean
gst_audio_ring_buffer_is_flushing (GstAudioRingBuffer * buf)
{
gboolean res;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), TRUE);
GST_OBJECT_LOCK (buf);
res = buf->flushing;
GST_OBJECT_UNLOCK (buf);
return res;
}
/**
* gst_audio_ring_buffer_start:
* @buf: the #GstAudioRingBuffer to start
*
* Start processing samples from the ringbuffer.
*
* Returns: TRUE if the device could be started, FALSE on error.
*
* MT safe.
*/
gboolean
gst_audio_ring_buffer_start (GstAudioRingBuffer * buf)
{
gboolean res = FALSE;
GstAudioRingBufferClass *rclass;
gboolean resume = FALSE;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
GST_DEBUG_OBJECT (buf, "starting ringbuffer");
GST_OBJECT_LOCK (buf);
if (G_UNLIKELY (buf->flushing))
goto flushing;
if (G_UNLIKELY (!buf->acquired))
goto not_acquired;
if (G_UNLIKELY (!g_atomic_int_get (&buf->may_start)))
goto may_not_start;
/* if stopped, set to started */
res = g_atomic_int_compare_and_exchange (&buf->state,
GST_AUDIO_RING_BUFFER_STATE_STOPPED, GST_AUDIO_RING_BUFFER_STATE_STARTED);
if (!res) {
GST_DEBUG_OBJECT (buf, "was not stopped, try paused");
/* was not stopped, try from paused */
res = g_atomic_int_compare_and_exchange (&buf->state,
GST_AUDIO_RING_BUFFER_STATE_PAUSED,
GST_AUDIO_RING_BUFFER_STATE_STARTED);
if (!res) {
/* was not paused either, must be started then */
res = TRUE;
GST_DEBUG_OBJECT (buf, "was not paused, must have been started");
goto done;
}
resume = TRUE;
GST_DEBUG_OBJECT (buf, "resuming");
}
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
if (resume) {
if (G_LIKELY (rclass->resume))
res = rclass->resume (buf);
} else {
if (G_LIKELY (rclass->start))
res = rclass->start (buf);
}
if (G_UNLIKELY (!res)) {
buf->state = GST_AUDIO_RING_BUFFER_STATE_PAUSED;
GST_DEBUG_OBJECT (buf, "failed to start");
} else {
GST_DEBUG_OBJECT (buf, "started");
}
done:
GST_OBJECT_UNLOCK (buf);
return res;
flushing:
{
GST_DEBUG_OBJECT (buf, "we are flushing");
GST_OBJECT_UNLOCK (buf);
return FALSE;
}
not_acquired:
{
GST_DEBUG_OBJECT (buf, "we are not acquired");
GST_OBJECT_UNLOCK (buf);
return FALSE;
}
may_not_start:
{
GST_DEBUG_OBJECT (buf, "we may not start");
GST_OBJECT_UNLOCK (buf);
return FALSE;
}
}
static gboolean
gst_audio_ring_buffer_pause_unlocked (GstAudioRingBuffer * buf)
{
gboolean res = FALSE;
GstAudioRingBufferClass *rclass;
GST_DEBUG_OBJECT (buf, "pausing ringbuffer");
/* if started, set to paused */
res = g_atomic_int_compare_and_exchange (&buf->state,
GST_AUDIO_RING_BUFFER_STATE_STARTED, GST_AUDIO_RING_BUFFER_STATE_PAUSED);
if (!res)
goto not_started;
/* signal any waiters */
GST_DEBUG_OBJECT (buf, "signal waiter");
GST_AUDIO_RING_BUFFER_SIGNAL (buf);
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
if (G_LIKELY (rclass->pause))
res = rclass->pause (buf);
if (G_UNLIKELY (!res)) {
buf->state = GST_AUDIO_RING_BUFFER_STATE_STARTED;
GST_DEBUG_OBJECT (buf, "failed to pause");
} else {
GST_DEBUG_OBJECT (buf, "paused");
}
return res;
not_started:
{
/* was not started */
GST_DEBUG_OBJECT (buf, "was not started");
return TRUE;
}
}
/**
* gst_audio_ring_buffer_pause:
* @buf: the #GstAudioRingBuffer to pause
*
* Pause processing samples from the ringbuffer.
*
* Returns: TRUE if the device could be paused, FALSE on error.
*
* MT safe.
*/
gboolean
gst_audio_ring_buffer_pause (GstAudioRingBuffer * buf)
{
gboolean res = FALSE;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
GST_OBJECT_LOCK (buf);
if (G_UNLIKELY (buf->flushing))
goto flushing;
if (G_UNLIKELY (!buf->acquired))
goto not_acquired;
res = gst_audio_ring_buffer_pause_unlocked (buf);
GST_OBJECT_UNLOCK (buf);
return res;
/* ERRORS */
flushing:
{
GST_DEBUG_OBJECT (buf, "we are flushing");
GST_OBJECT_UNLOCK (buf);
return FALSE;
}
not_acquired:
{
GST_DEBUG_OBJECT (buf, "not acquired");
GST_OBJECT_UNLOCK (buf);
return FALSE;
}
}
/**
* gst_audio_ring_buffer_stop:
* @buf: the #GstAudioRingBuffer to stop
*
* Stop processing samples from the ringbuffer.
*
* Returns: TRUE if the device could be stopped, FALSE on error.
*
* MT safe.
*/
gboolean
gst_audio_ring_buffer_stop (GstAudioRingBuffer * buf)
{
gboolean res = FALSE;
GstAudioRingBufferClass *rclass;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
GST_DEBUG_OBJECT (buf, "stopping");
GST_OBJECT_LOCK (buf);
/* if started, set to stopped */
res = g_atomic_int_compare_and_exchange (&buf->state,
GST_AUDIO_RING_BUFFER_STATE_STARTED, GST_AUDIO_RING_BUFFER_STATE_STOPPED);
if (!res) {
GST_DEBUG_OBJECT (buf, "was not started, try paused");
/* was not started, try from paused */
res = g_atomic_int_compare_and_exchange (&buf->state,
GST_AUDIO_RING_BUFFER_STATE_PAUSED,
GST_AUDIO_RING_BUFFER_STATE_STOPPED);
if (!res) {
/* was not paused either, must have been stopped then */
res = TRUE;
GST_DEBUG_OBJECT (buf, "was not paused, must have been stopped");
goto done;
}
}
/* signal any waiters */
GST_DEBUG_OBJECT (buf, "signal waiter");
GST_AUDIO_RING_BUFFER_SIGNAL (buf);
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
if (G_LIKELY (rclass->stop))
res = rclass->stop (buf);
if (G_UNLIKELY (!res)) {
buf->state = GST_AUDIO_RING_BUFFER_STATE_STARTED;
GST_DEBUG_OBJECT (buf, "failed to stop");
} else {
GST_DEBUG_OBJECT (buf, "stopped");
}
done:
GST_OBJECT_UNLOCK (buf);
return res;
}
/**
* gst_audio_ring_buffer_delay:
* @buf: the #GstAudioRingBuffer to query
*
* Get the number of samples queued in the audio device. This is
* usually less than the segment size but can be bigger when the
* implementation uses another internal buffer between the audio
* device.
*
* For playback ringbuffers this is the amount of samples transferred from the
* ringbuffer to the device but still not played.
*
* For capture ringbuffers this is the amount of samples in the device that are
* not yet transferred to the ringbuffer.
*
* Returns: The number of samples queued in the audio device.
*
* MT safe.
*/
guint
gst_audio_ring_buffer_delay (GstAudioRingBuffer * buf)
{
GstAudioRingBufferClass *rclass;
guint res;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), 0);
/* buffer must be acquired */
if (G_UNLIKELY (!gst_audio_ring_buffer_is_acquired (buf)))
goto not_acquired;
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
if (G_LIKELY (rclass->delay))
res = rclass->delay (buf);
else
res = 0;
return res;
not_acquired:
{
GST_DEBUG_OBJECT (buf, "not acquired");
return 0;
}
}
/**
* gst_audio_ring_buffer_samples_done:
* @buf: the #GstAudioRingBuffer to query
*
* Get the number of samples that were processed by the ringbuffer
* since it was last started. This does not include the number of samples not
* yet processed (see gst_audio_ring_buffer_delay()).
*
* Returns: The number of samples processed by the ringbuffer.
*
* MT safe.
*/
guint64
gst_audio_ring_buffer_samples_done (GstAudioRingBuffer * buf)
{
gint segdone;
guint64 samples;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), 0);
/* get the amount of segments we processed */
segdone = g_atomic_int_get (&buf->segdone);
/* convert to samples */
samples = ((guint64) segdone) * buf->samples_per_seg;
return samples;
}
/**
* gst_audio_ring_buffer_set_sample:
* @buf: the #GstAudioRingBuffer to use
* @sample: the sample number to set
*
* Make sure that the next sample written to the device is
* accounted for as being the @sample sample written to the
* device. This value will be used in reporting the current
* sample position of the ringbuffer.
*
* This function will also clear the buffer with silence.
*
* MT safe.
*/
void
gst_audio_ring_buffer_set_sample (GstAudioRingBuffer * buf, guint64 sample)
{
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
if (sample == -1)
sample = 0;
if (G_UNLIKELY (buf->samples_per_seg == 0))
return;
/* FIXME, we assume the ringbuffer can restart at a random
* position, round down to the beginning and keep track of
* offset when calculating the processed samples. */
buf->segbase = buf->segdone - sample / buf->samples_per_seg;
gst_audio_ring_buffer_clear_all (buf);
GST_DEBUG_OBJECT (buf, "set sample to %" G_GUINT64_FORMAT ", segbase %d",
sample, buf->segbase);
}
/**
* default_clear_all:
* @buf: the #GstAudioRingBuffer to clear
*
* Fill the ringbuffer with silence.
*/
static void
default_clear_all (GstAudioRingBuffer * buf)
{
gint i;
/* not fatal, we just are not negotiated yet */
if (G_UNLIKELY (buf->spec.segtotal <= 0))
return;
GST_DEBUG_OBJECT (buf, "clear all segments");
for (i = 0; i < buf->spec.segtotal; i++) {
gst_audio_ring_buffer_clear (buf, i);
}
}
/**
* gst_audio_ring_buffer_clear_all:
* @buf: the #GstAudioRingBuffer to clear
*
* Clear all samples from the ringbuffer.
*
* MT safe.
*/
void
gst_audio_ring_buffer_clear_all (GstAudioRingBuffer * buf)
{
GstAudioRingBufferClass *rclass;
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
if (G_LIKELY (rclass->clear_all))
rclass->clear_all (buf);
}
static gboolean
wait_segment (GstAudioRingBuffer * buf)
{
gint segments;
gboolean wait = TRUE;
/* buffer must be started now or we deadlock since nobody is reading */
if (G_UNLIKELY (g_atomic_int_get (&buf->state) !=
GST_AUDIO_RING_BUFFER_STATE_STARTED)) {
/* see if we are allowed to start it */
if (G_UNLIKELY (!g_atomic_int_get (&buf->may_start)))
goto no_start;
GST_DEBUG_OBJECT (buf, "start!");
segments = g_atomic_int_get (&buf->segdone);
gst_audio_ring_buffer_start (buf);
/* After starting, the writer may have wrote segments already and then we
* don't need to wait anymore */
if (G_LIKELY (g_atomic_int_get (&buf->segdone) != segments))
wait = FALSE;
}
/* take lock first, then update our waiting flag */
GST_OBJECT_LOCK (buf);
if (G_UNLIKELY (buf->flushing))
goto flushing;
if (G_UNLIKELY (g_atomic_int_get (&buf->state) !=
GST_AUDIO_RING_BUFFER_STATE_STARTED))
goto not_started;
if (G_LIKELY (wait)) {
if (g_atomic_int_compare_and_exchange (&buf->waiting, 0, 1)) {
GST_DEBUG_OBJECT (buf, "waiting..");
GST_AUDIO_RING_BUFFER_WAIT (buf);
if (G_UNLIKELY (buf->flushing))
goto flushing;
if (G_UNLIKELY (g_atomic_int_get (&buf->state) !=
GST_AUDIO_RING_BUFFER_STATE_STARTED))
goto not_started;
}
}
GST_OBJECT_UNLOCK (buf);
return TRUE;
/* ERROR */
not_started:
{
g_atomic_int_compare_and_exchange (&buf->waiting, 1, 0);
GST_DEBUG_OBJECT (buf, "stopped processing");
GST_OBJECT_UNLOCK (buf);
return FALSE;
}
flushing:
{
g_atomic_int_compare_and_exchange (&buf->waiting, 1, 0);
GST_DEBUG_OBJECT (buf, "flushing");
GST_OBJECT_UNLOCK (buf);
return FALSE;
}
no_start:
{
GST_DEBUG_OBJECT (buf, "not allowed to start");
return FALSE;
}
}
#define REORDER_SAMPLE(d, s, l) \
G_STMT_START { \
gint i; \
for (i = 0; i < channels; i++) { \
memcpy (d + reorder_map[i] * bps, s + i * bps, bps); \
} \
} G_STMT_END
#define REORDER_SAMPLES(d, s, len) \
G_STMT_START { \
gint i, len_ = len / bpf; \
guint8 *d_ = d, *s_ = s; \
for (i = 0; i < len_; i++) { \
REORDER_SAMPLE(d_, s_, bpf); \
d_ += bpf; \
s_ += bpf; \
} \
} G_STMT_END
#define FWD_SAMPLES(s,se,d,de,F) \
G_STMT_START { \
/* no rate conversion */ \
guint towrite = MIN (se + bpf - s, de - d); \
/* simple copy */ \
if (!skip) \
F (d, s, towrite); \
in_samples -= towrite / bpf; \
out_samples -= towrite / bpf; \
s += towrite; \
GST_DEBUG ("copy %u bytes", towrite); \
} G_STMT_END
/* in_samples >= out_samples, rate > 1.0 */
#define FWD_UP_SAMPLES(s,se,d,de,F) \
G_STMT_START { \
guint8 *sb = s, *db = d; \
while (s <= se && d < de) { \
if (!skip) \
F (d, s, bpf); \
s += bpf; \
*accum += outr; \
if ((*accum << 1) >= inr) { \
*accum -= inr; \
d += bpf; \
} \
} \
in_samples -= (s - sb)/bpf; \
out_samples -= (d - db)/bpf; \
GST_DEBUG ("fwd_up end %d/%d",*accum,*toprocess); \
} G_STMT_END
/* out_samples > in_samples, for rates smaller than 1.0 */
#define FWD_DOWN_SAMPLES(s,se,d,de,F) \
G_STMT_START { \
guint8 *sb = s, *db = d; \
while (s <= se && d < de) { \
if (!skip) \
F (d, s, bpf); \
d += bpf; \
*accum += inr; \
if ((*accum << 1) >= outr) { \
*accum -= outr; \
s += bpf; \
} \
} \
in_samples -= (s - sb)/bpf; \
out_samples -= (d - db)/bpf; \
GST_DEBUG ("fwd_down end %d/%d",*accum,*toprocess); \
} G_STMT_END
#define REV_UP_SAMPLES(s,se,d,de,F) \
G_STMT_START { \
guint8 *sb = se, *db = d; \
while (s <= se && d < de) { \
if (!skip) \
F (d, se, bpf); \
se -= bpf; \
*accum += outr; \
while (d < de && (*accum << 1) >= inr) { \
*accum -= inr; \
d += bpf; \
} \
} \
in_samples -= (sb - se)/bpf; \
out_samples -= (d - db)/bpf; \
GST_DEBUG ("rev_up end %d/%d",*accum,*toprocess); \
} G_STMT_END
#define REV_DOWN_SAMPLES(s,se,d,de,F) \
G_STMT_START { \
guint8 *sb = se, *db = d; \
while (s <= se && d < de) { \
if (!skip) \
F (d, se, bpf); \
d += bpf; \
*accum += inr; \
while (s <= se && (*accum << 1) >= outr) { \
*accum -= outr; \
se -= bpf; \
} \
} \
in_samples -= (sb - se)/bpf; \
out_samples -= (d - db)/bpf; \
GST_DEBUG ("rev_down end %d/%d",*accum,*toprocess); \
} G_STMT_END
static guint
default_commit (GstAudioRingBuffer * buf, guint64 * sample,
guint8 * data, gint in_samples, gint out_samples, gint * accum)
{
gint segdone;
gint segsize, segtotal, channels, bps, bpf, sps;
guint8 *dest, *data_end;
gint writeseg, sampleoff;
gint *toprocess;
gint inr, outr;
gboolean reverse;
gboolean need_reorder;
g_return_val_if_fail (buf->memory != NULL, -1);
g_return_val_if_fail (data != NULL, -1);
need_reorder = buf->need_reorder;
channels = buf->spec.info.channels;
dest = buf->memory;
segsize = buf->spec.segsize;
segtotal = buf->spec.segtotal;
bpf = buf->spec.info.bpf;
bps = bpf / channels;
sps = buf->samples_per_seg;
reverse = out_samples < 0;
out_samples = ABS (out_samples);
if (in_samples >= out_samples)
toprocess = &in_samples;
else
toprocess = &out_samples;
inr = in_samples - 1;
outr = out_samples - 1;
/* data_end points to the last sample we have to write, not past it. This is
* needed to properly handle reverse playback: it points to the last sample. */
data_end = data + (bpf * inr);
/* figure out the segment and the offset inside the segment where
* the first sample should be written. */
writeseg = *sample / sps;
sampleoff = (*sample % sps) * bpf;
GST_DEBUG_OBJECT (buf, "write %d : %d", in_samples, out_samples);
/* write out all samples */
while (*toprocess > 0) {
gint avail;
guint8 *d, *d_end;
gint ws;
gboolean skip;
while (TRUE) {
gint diff;
/* get the currently processed segment */
segdone = g_atomic_int_get (&buf->segdone) - buf->segbase;
/* see how far away it is from the write segment */
diff = writeseg - segdone;
GST_DEBUG_OBJECT (buf,
"pointer at %d, write to %d-%d, diff %d, segtotal %d, segsize %d, base %d",
segdone, writeseg, sampleoff, diff, segtotal, segsize, buf->segbase);
/* segment too far ahead, writer too slow, we need to drop, hopefully UNLIKELY */
if (G_UNLIKELY (diff < 0)) {
/* we need to drop one segment at a time, pretend we wrote a segment. */
skip = TRUE;
break;
}
/* write segment is within writable range, we can break the loop and
* start writing the data. */
if (diff < segtotal) {
skip = FALSE;
break;
}
/* else we need to wait for the segment to become writable. */
if (!wait_segment (buf))
goto not_started;
}
/* we can write now */
ws = writeseg % segtotal;
avail = MIN (segsize - sampleoff, bpf * out_samples);
d = dest + (ws * segsize) + sampleoff;
d_end = d + avail;
*sample += avail / bpf;
GST_DEBUG_OBJECT (buf, "write @%p seg %d, sps %d, off %d, avail %d",
dest + ws * segsize, ws, sps, sampleoff, avail);
if (need_reorder) {
gint *reorder_map = buf->channel_reorder_map;
if (G_LIKELY (inr == outr && !reverse)) {
/* no rate conversion, simply copy samples */
FWD_SAMPLES (data, data_end, d, d_end, REORDER_SAMPLES);
} else if (!reverse) {
if (inr >= outr)
/* forward speed up */
FWD_UP_SAMPLES (data, data_end, d, d_end, REORDER_SAMPLE);
else
/* forward slow down */
FWD_DOWN_SAMPLES (data, data_end, d, d_end, REORDER_SAMPLE);
} else {
if (inr >= outr)
/* reverse speed up */
REV_UP_SAMPLES (data, data_end, d, d_end, REORDER_SAMPLE);
else
/* reverse slow down */
REV_DOWN_SAMPLES (data, data_end, d, d_end, REORDER_SAMPLE);
}
} else {
if (G_LIKELY (inr == outr && !reverse)) {
/* no rate conversion, simply copy samples */
FWD_SAMPLES (data, data_end, d, d_end, memcpy);
} else if (!reverse) {
if (inr >= outr)
/* forward speed up */
FWD_UP_SAMPLES (data, data_end, d, d_end, memcpy);
else
/* forward slow down */
FWD_DOWN_SAMPLES (data, data_end, d, d_end, memcpy);
} else {
if (inr >= outr)
/* reverse speed up */
REV_UP_SAMPLES (data, data_end, d, d_end, memcpy);
else
/* reverse slow down */
REV_DOWN_SAMPLES (data, data_end, d, d_end, memcpy);
}
}
/* for the next iteration we write to the next segment at the beginning. */
writeseg++;
sampleoff = 0;
}
/* we consumed all samples here */
data = data_end + bpf;
done:
return inr - ((data_end - data) / bpf);
/* ERRORS */
not_started:
{
GST_DEBUG_OBJECT (buf, "stopped processing");
goto done;
}
}
/**
* gst_audio_ring_buffer_commit:
* @buf: the #GstAudioRingBuffer to commit
* @sample: (inout): the sample position of the data
* @data: (array length=in_samples): the data to commit
* @in_samples: the number of samples in the data to commit
* @out_samples: the number of samples to write to the ringbuffer
* @accum: (inout): accumulator for rate conversion.
*
* Commit @in_samples samples pointed to by @data to the ringbuffer @buf.
*
* @in_samples and @out_samples define the rate conversion to perform on the
* samples in @data. For negative rates, @out_samples must be negative and
* @in_samples positive.
*
* When @out_samples is positive, the first sample will be written at position @sample
* in the ringbuffer. When @out_samples is negative, the last sample will be written to
* @sample in reverse order.
*
* @out_samples does not need to be a multiple of the segment size of the ringbuffer
* although it is recommended for optimal performance.
*
* @accum will hold a temporary accumulator used in rate conversion and should be
* set to 0 when this function is first called. In case the commit operation is
* interrupted, one can resume the processing by passing the previously returned
* @accum value back to this function.
*
* MT safe.
*
* Returns: The number of samples written to the ringbuffer or -1 on error. The
* number of samples written can be less than @out_samples when @buf was interrupted
* with a flush or stop.
*/
guint
gst_audio_ring_buffer_commit (GstAudioRingBuffer * buf, guint64 * sample,
guint8 * data, gint in_samples, gint out_samples, gint * accum)
{
GstAudioRingBufferClass *rclass;
guint res = -1;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), -1);
if (G_UNLIKELY (in_samples == 0 || out_samples == 0))
return in_samples;
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
if (G_LIKELY (rclass->commit))
res = rclass->commit (buf, sample, data, in_samples, out_samples, accum);
return res;
}
/**
* gst_audio_ring_buffer_read:
* @buf: the #GstAudioRingBuffer to read from
* @sample: the sample position of the data
* @data: (array length=len): where the data should be read
* @len: the number of samples in data to read
* @timestamp: (out): where the timestamp is returned
*
* Read @len samples from the ringbuffer into the memory pointed
* to by @data.
* The first sample should be read from position @sample in
* the ringbuffer.
*
* @len should not be a multiple of the segment size of the ringbuffer
* although it is recommended.
*
* @timestamp will return the timestamp associated with the data returned.
*
* Returns: The number of samples read from the ringbuffer or -1 on
* error.
*
* MT safe.
*/
guint
gst_audio_ring_buffer_read (GstAudioRingBuffer * buf, guint64 sample,
guint8 * data, guint len, GstClockTime * timestamp)
{
gint segdone;
gint segsize, segtotal, channels, bps, bpf, sps, readseg = 0;
guint8 *dest;
guint to_read;
gboolean need_reorder;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), -1);
g_return_val_if_fail (buf->memory != NULL, -1);
g_return_val_if_fail (data != NULL, -1);
need_reorder = buf->need_reorder;
dest = buf->memory;
segsize = buf->spec.segsize;
segtotal = buf->spec.segtotal;
channels = buf->spec.info.channels;
bpf = buf->spec.info.bpf;
bps = bpf / channels;
sps = buf->samples_per_seg;
to_read = len;
/* read enough samples */
while (to_read > 0) {
gint sampleslen;
gint sampleoff;
/* figure out the segment and the offset inside the segment where
* the sample should be read from. */
readseg = sample / sps;
sampleoff = (sample % sps);
while (TRUE) {
gint diff;
/* get the currently processed segment */
segdone = g_atomic_int_get (&buf->segdone) - buf->segbase;
/* see how far away it is from the read segment, normally segdone (where
* the hardware is writing) is bigger than readseg (where software is
* reading) */
diff = segdone - readseg;
GST_DEBUG_OBJECT
(buf, "pointer at %d, sample %" G_GUINT64_FORMAT
", read from %d-%d, to_read %d, diff %d, segtotal %d, segsize %d",
segdone, sample, readseg, sampleoff, to_read, diff, segtotal,
segsize);
/* segment too far ahead, reader too slow */
if (G_UNLIKELY (diff >= segtotal)) {
/* pretend we read an empty segment. */
sampleslen = MIN (sps, to_read);
memcpy (data, buf->empty_seg, sampleslen * bpf);
goto next;
}
/* read segment is within readable range, we can break the loop and
* start reading the data. */
if (diff > 0)
break;
/* else we need to wait for the segment to become readable. */
if (!wait_segment (buf))
goto not_started;
}
/* we can read now */
readseg = readseg % segtotal;
sampleslen = MIN (sps - sampleoff, to_read);
GST_DEBUG_OBJECT (buf, "read @%p seg %d, off %d, sampleslen %d",
dest + readseg * segsize, readseg, sampleoff, sampleslen);
if (need_reorder) {
guint8 *ptr = dest + (readseg * segsize) + (sampleoff * bpf);
gint i, j;
gint *reorder_map = buf->channel_reorder_map;
/* Reorder from device order to GStreamer order */
for (i = 0; i < sampleslen; i++) {
for (j = 0; j < channels; j++) {
memcpy (data + i * bpf + reorder_map[j] * bps, ptr + j * bps, bps);
}
ptr += bpf;
}
} else {
memcpy (data, dest + (readseg * segsize) + (sampleoff * bpf),
(sampleslen * bpf));
}
next:
to_read -= sampleslen;
sample += sampleslen;
data += sampleslen * bpf;
}
if (buf->timestamps && timestamp) {
*timestamp = buf->timestamps[readseg % segtotal];
GST_DEBUG_OBJECT (buf, "Retrieved timestamp %" GST_TIME_FORMAT
" @ %d", GST_TIME_ARGS (*timestamp), readseg % segtotal);
}
return len - to_read;
/* ERRORS */
not_started:
{
GST_DEBUG_OBJECT (buf, "stopped processing");
return len - to_read;
}
}
/**
* gst_audio_ring_buffer_prepare_read:
* @buf: the #GstAudioRingBuffer to read from
* @segment: (out): the segment to read
* @readptr: (out) (array length=len):
* the pointer to the memory where samples can be read
* @len: (out): the number of bytes to read
*
* Returns a pointer to memory where the data from segment @segment
* can be found. This function is mostly used by subclasses.
*
* Returns: FALSE if the buffer is not started.
*
* MT safe.
*/
gboolean
gst_audio_ring_buffer_prepare_read (GstAudioRingBuffer * buf, gint * segment,
guint8 ** readptr, gint * len)
{
guint8 *data;
gint segdone;
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
if (buf->callback == NULL) {
/* push mode, fail when nothing is started */
if (g_atomic_int_get (&buf->state) != GST_AUDIO_RING_BUFFER_STATE_STARTED)
return FALSE;
}
g_return_val_if_fail (buf->memory != NULL, FALSE);
g_return_val_if_fail (segment != NULL, FALSE);
g_return_val_if_fail (readptr != NULL, FALSE);
g_return_val_if_fail (len != NULL, FALSE);
data = buf->memory;
/* get the position of the pointer */
segdone = g_atomic_int_get (&buf->segdone);
*segment = segdone % buf->spec.segtotal;
*len = buf->spec.segsize;
*readptr = data + *segment * *len;
GST_LOG_OBJECT (buf, "prepare read from segment %d (real %d) @%p",
*segment, segdone, *readptr);
/* callback to fill the memory with data, for pull based
* scheduling. */
if (buf->callback)
buf->callback (buf, *readptr, *len, buf->cb_data);
return TRUE;
}
/**
* gst_audio_ring_buffer_advance:
* @buf: the #GstAudioRingBuffer to advance
* @advance: the number of segments written
*
* Subclasses should call this function to notify the fact that
* @advance segments are now processed by the device.
*
* MT safe.
*/
void
gst_audio_ring_buffer_advance (GstAudioRingBuffer * buf, guint advance)
{
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
/* update counter */
g_atomic_int_add (&buf->segdone, advance);
/* the lock is already taken when the waiting flag is set,
* we grab the lock as well to make sure the waiter is actually
* waiting for the signal */
if (g_atomic_int_compare_and_exchange (&buf->waiting, 1, 0)) {
GST_OBJECT_LOCK (buf);
GST_DEBUG_OBJECT (buf, "signal waiter");
GST_AUDIO_RING_BUFFER_SIGNAL (buf);
GST_OBJECT_UNLOCK (buf);
}
}
/**
* gst_audio_ring_buffer_clear:
* @buf: the #GstAudioRingBuffer to clear
* @segment: the segment to clear
*
* Clear the given segment of the buffer with silence samples.
* This function is used by subclasses.
*
* MT safe.
*/
void
gst_audio_ring_buffer_clear (GstAudioRingBuffer * buf, gint segment)
{
guint8 *data;
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
/* no data means it's already cleared */
if (G_UNLIKELY (buf->memory == NULL))
return;
/* no empty_seg means it's not opened */
if (G_UNLIKELY (buf->empty_seg == NULL))
return;
segment %= buf->spec.segtotal;
data = buf->memory;
data += segment * buf->spec.segsize;
GST_LOG_OBJECT (buf, "clear segment %d @%p", segment, data);
memcpy (data, buf->empty_seg, buf->spec.segsize);
}
/**
* gst_audio_ring_buffer_may_start:
* @buf: the #GstAudioRingBuffer
* @allowed: the new value
*
* Tell the ringbuffer that it is allowed to start playback when
* the ringbuffer is filled with samples.
*
* MT safe.
*/
void
gst_audio_ring_buffer_may_start (GstAudioRingBuffer * buf, gboolean allowed)
{
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
GST_LOG_OBJECT (buf, "may start: %d", allowed);
g_atomic_int_set (&buf->may_start, allowed);
}
/* GST_AUDIO_CHANNEL_POSITION_NONE is used for position-less
* mutually exclusive channels. In this case we should not attempt
* to do any reordering.
*/
static gboolean
position_less_channels (const GstAudioChannelPosition * pos, guint channels)
{
guint i;
for (i = 0; i < channels; i++) {
if (pos[i] != GST_AUDIO_CHANNEL_POSITION_NONE)
return FALSE;
}
return TRUE;
}
/**
* gst_audio_ring_buffer_set_channel_positions:
* @buf: the #GstAudioRingBuffer
* @position: (array): the device channel positions
*
* Tell the ringbuffer about the device's channel positions. This must
* be called in when the ringbuffer is acquired.
*/
void
gst_audio_ring_buffer_set_channel_positions (GstAudioRingBuffer * buf,
const GstAudioChannelPosition * position)
{
const GstAudioChannelPosition *to;
gint channels;
gint i;
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
g_return_if_fail (buf->acquired);
channels = buf->spec.info.channels;
to = buf->spec.info.position;
buf->need_reorder = FALSE;
if (memcmp (position, to, channels * sizeof (to[0])) == 0)
return;
if (position_less_channels (position, channels)) {
GST_LOG_OBJECT (buf, "position-less channels, no need to reorder");
return;
}
if (!gst_audio_get_channel_reorder_map (channels, position, to,
buf->channel_reorder_map))
g_return_if_reached ();
for (i = 0; i < channels; i++) {
if (buf->channel_reorder_map[i] != i) {
#ifndef GST_DISABLE_GST_DEBUG
{
gchar *tmp1, *tmp2;
tmp1 = gst_audio_channel_positions_to_string (position, channels);
tmp2 = gst_audio_channel_positions_to_string (to, channels);
GST_LOG_OBJECT (buf, "may have to reorder channels: %s -> %s", tmp1,
tmp2);
g_free (tmp1);
g_free (tmp2);
}
#endif /* GST_DISABLE_GST_DEBUG */
buf->need_reorder = TRUE;
break;
}
}
}
/**
* gst_ring_buffer_set_timestamp:
* @buf: the #GstRingBuffer
* @readseg: the current data segment
* @timestamp: The new timestamp of the buffer.
*
* Set a new timestamp on the buffer.
*
* MT safe.
*/
void
gst_audio_ring_buffer_set_timestamp (GstAudioRingBuffer * buf, gint readseg,
GstClockTime timestamp)
{
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
GST_DEBUG_OBJECT (buf, "Storing timestamp %" GST_TIME_FORMAT
" @ %d", GST_TIME_ARGS (timestamp), readseg);
GST_OBJECT_LOCK (buf);
if (G_UNLIKELY (!buf->acquired))
goto not_acquired;
buf->timestamps[readseg] = timestamp;
done:
GST_OBJECT_UNLOCK (buf);
return;
not_acquired:
{
GST_DEBUG_OBJECT (buf, "we are not acquired");
goto done;
}
}
Reference in a new issue View git blame Copy permalink