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gstreamer/gst-libs/gst/audio/gstnonstreamaudiodecoder.c
Thibault Saunier e8ffa0d39c openmpt: Namespace enum GType names
2020-06-23 13:02:57 -04:00

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/* GStreamer
* Copyright (C) <2017> Carlos Rafael Giani <dv at pseudoterminal dot org>
*
* 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:gstnonstreamaudiodecoder
* @short_description: Base class for decoding of non-streaming audio
* @see_also: #GstAudioDecoder
*
* This base class is for decoders which do not operate on a streaming model.
* That is: they load the encoded media at once, as part of an initialization,
* and afterwards can decode samples (sometimes referred to as "rendering the
* samples").
*
* This sets it apart from GstAudioDecoder, which is a base class for
* streaming audio decoders.
*
* The base class is conceptually a mix between decoder and parser. This is
* unavoidable, since virtually no format that isn't streaming based has a
* clear distinction between parsing and decoding. As a result, this class
* also handles seeking.
*
* Non-streaming audio formats tend to have some characteristics unknown to
* more "regular" bitstreams. These include subsongs and looping.
*
* Subsongs are a set of songs-within-a-song. An analogy would be a multitrack
* recording, where each track is its own song. The first subsong is typically
* the "main" one. Subsongs were popular for video games to enable context-
* aware music; for example, subsong `#0` would be the "main" song, `#1` would be
* an alternate song playing when a fight started, `#2` would be heard during
* conversations etc. The base class is designed to always have at least one
* subsong. If the subclass doesn't provide any, the base class creates a
* "pseudo" subsong, which is actually the whole song.
* Downstream is informed about the subsong using a table of contents (TOC),
* but only if there are at least 2 subsongs.
*
* Looping refers to jumps within the song, typically backwards to the loop
* start (although bi-directional looping is possible). The loop is defined
* by a chronological start and end; once the playback position reaches the
* loop end, it jumps back to the loop start.
* Depending on the subclass, looping may not be possible at all, or it
* may only be possible to enable/disable it (that is, either no looping, or
* an infinite amount of loops), or it may allow for defining a finite number
* of times the loop is repeated.
* Looping can affect output in two ways. Either, the playback position is
* reset to the start of the loop, similar to what happens after a seek event.
* Or, it is not reset, so the pipeline sees playback steadily moving forwards,
* the playback position monotonically increasing. However, seeking must
* always happen within the confines of the defined subsong duration; for
* example, if a subsong is 2 minutes long, steady playback is at 5 minutes
* (because infinite looping is enabled), then seeking will still place the
* position within the 2 minute period.
* Loop count 0 means no looping. Loop count -1 means infinite looping.
* Nonzero positive values indicate how often a loop shall occur.
*
* If the initial subsong and loop count are set to values the subclass does
* not support, the subclass has a chance to correct these values.
* @get_property then reports the corrected versions.
*
* The base class operates as follows:
* * Unloaded mode
* - Initial values are set. If a current subsong has already been
* defined (for example over the command line with gst-launch), then
* the subsong index is copied over to current_subsong .
* Same goes for the num-loops and output-mode properties.
* Media is NOT loaded yet.
* - Once the sinkpad is activated, the process continues. The sinkpad is
* activated in push mode, and the class accumulates the incoming media
* data in an adapter inside the sinkpad's chain function until either an
* EOS event is received from upstream, or the number of bytes reported
* by upstream is reached. Then it loads the media, and starts the decoder
* output task.
* - If upstream cannot respond to the size query (in bytes) of @load_from_buffer
* fails, an error is reported, and the pipeline stops.
* - If there are no errors, @load_from_buffer is called to load the media. The
* subclass must at least call gst_nonstream_audio_decoder_set_output_format()
* there, and is free to make use of the initial subsong, output mode, and
* position. If the actual output mode or position differs from the initial
* value,it must set the initial value to the actual one (for example, if
* the actual starting position is always 0, set *initial_position to 0).
* If loading is unsuccessful, an error is reported, and the pipeline
* stops. Otherwise, the base class calls @get_current_subsong to retrieve
* the actual current subsong, @get_subsong_duration to report the current
* subsong's duration in a duration event and message, and @get_subsong_tags
* to send tags downstream in an event (these functions are optional; if
* set to NULL, the associated operation is skipped). Afterwards, the base
* class switches to loaded mode, and starts the decoder output task.
*
* * Loaded mode</title>
* - Inside the decoder output task, the base class repeatedly calls @decode,
* which returns a buffer with decoded, ready-to-play samples. If the
* subclass reached the end of playback, @decode returns FALSE, otherwise
* TRUE.
* - Upon reaching a loop end, subclass either ignores that, or loops back
* to the beginning of the loop. In the latter case, if the output mode is set
* to LOOPING, the subclass must call gst_nonstream_audio_decoder_handle_loop()
* *after* the playback position moved to the start of the loop. In
* STEADY mode, the subclass must *not* call this function.
* Since many decoders only provide a callback for when the looping occurs,
* and that looping occurs inside the decoding operation itself, the following
* mechanism for subclass is suggested: set a flag inside such a callback.
* Then, in the next @decode call, before doing the decoding, check this flag.
* If it is set, gst_nonstream_audio_decoder_handle_loop() is called, and the
* flag is cleared.
* (This function call is necessary in LOOPING mode because it updates the
* current segment and makes sure the next buffer that is sent downstream
* has its DISCONT flag set.)
* - When the current subsong is switched, @set_current_subsong is called.
* If it fails, a warning is reported, and nothing else is done. Otherwise,
* it calls @get_subsong_duration to get the new current subsongs's
* duration, @get_subsong_tags to get its tags, reports a new duration
* (i.e. it sends a duration event downstream and generates a duration
* message), updates the current segment, and sends the subsong's tags in
* an event downstream. (If @set_current_subsong has been set to NULL by
* the subclass, attempts to set a current subsong are ignored; likewise,
* if @get_subsong_duration is NULL, no duration is reported, and if
* @get_subsong_tags is NULL, no tags are sent downstream.)
* - When an attempt is made to switch the output mode, it is checked against
* the bitmask returned by @get_supported_output_modes. If the proposed
* new output mode is supported, the current segment is updated
* (it is open-ended in STEADY mode, and covers the (sub)song length in
* LOOPING mode), and the subclass' @set_output_mode function is called
* unless it is set to NULL. Subclasses should reset internal loop counters
* in this function.
*
* The relationship between (sub)song duration, output mode, and number of loops
* is defined this way (this is all done by the base class automatically):
*
* * Segments have their duration and stop values set to GST_CLOCK_TIME_NONE in
* STEADY mode, and to the duration of the (sub)song in LOOPING mode.
*
* * The duration that is returned to a DURATION query is always the duration
* of the (sub)song, regardless of number of loops or output mode. The same
* goes for DURATION messages and tags.
*
* * If the number of loops is >0 or -1, durations of TOC entries are set to
* the duration of the respective subsong in LOOPING mode and to G_MAXINT64 in
* STEADY mode. If the number of loops is 0, entry durations are set to the
* subsong duration regardless of the output mode.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdio.h>
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include "gstnonstreamaudiodecoder.h"
GST_DEBUG_CATEGORY (nonstream_audiodecoder_debug);
#define GST_CAT_DEFAULT nonstream_audiodecoder_debug
enum
{
PROP_0,
PROP_CURRENT_SUBSONG,
PROP_SUBSONG_MODE,
PROP_NUM_LOOPS,
PROP_OUTPUT_MODE
};
#define DEFAULT_CURRENT_SUBSONG 0
#define DEFAULT_SUBSONG_MODE GST_NONSTREAM_AUDIO_SUBSONG_MODE_DECODER_DEFAULT
#define DEFAULT_NUM_SUBSONGS 0
#define DEFAULT_NUM_LOOPS 0
#define DEFAULT_OUTPUT_MODE GST_NONSTREAM_AUDIO_OUTPUT_MODE_STEADY
static GstElementClass *gst_nonstream_audio_decoder_parent_class = NULL;
static void
gst_nonstream_audio_decoder_class_init (GstNonstreamAudioDecoderClass * klass);
static void gst_nonstream_audio_decoder_init (GstNonstreamAudioDecoder * dec,
GstNonstreamAudioDecoderClass * klass);
static void gst_nonstream_audio_decoder_finalize (GObject * object);
static void gst_nonstream_audio_decoder_set_property (GObject * object,
guint prop_id, GValue const *value, GParamSpec * pspec);
static void gst_nonstream_audio_decoder_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_nonstream_audio_decoder_change_state (GstElement
* element, GstStateChange transition);
static gboolean gst_nonstream_audio_decoder_sink_event (GstPad * pad,
GstObject * parent, GstEvent * event);
static gboolean gst_nonstream_audio_decoder_sink_query (GstPad * pad,
GstObject * parent, GstQuery * query);
static GstFlowReturn gst_nonstream_audio_decoder_chain (GstPad * pad,
GstObject * parent, GstBuffer * buffer);
static gboolean gst_nonstream_audio_decoder_src_event (GstPad * pad,
GstObject * parent, GstEvent * event);
static gboolean gst_nonstream_audio_decoder_src_query (GstPad * pad,
GstObject * parent, GstQuery * query);
static void
gst_nonstream_audio_decoder_set_initial_state (GstNonstreamAudioDecoder * dec);
static void gst_nonstream_audio_decoder_cleanup_state (GstNonstreamAudioDecoder
* dec);
static gboolean gst_nonstream_audio_decoder_negotiate (GstNonstreamAudioDecoder
* dec);
static gboolean
gst_nonstream_audio_decoder_negotiate_default (GstNonstreamAudioDecoder * dec);
static gboolean
gst_nonstream_audio_decoder_decide_allocation_default (GstNonstreamAudioDecoder
* dec, GstQuery * query);
static gboolean
gst_nonstream_audio_decoder_propose_allocation_default (GstNonstreamAudioDecoder
* dec, GstQuery * query);
static gboolean
gst_nonstream_audio_decoder_get_upstream_size (GstNonstreamAudioDecoder * dec,
gint64 * length);
static gboolean
gst_nonstream_audio_decoder_load_from_buffer (GstNonstreamAudioDecoder * dec,
GstBuffer * buffer);
static gboolean
gst_nonstream_audio_decoder_load_from_custom (GstNonstreamAudioDecoder * dec);
static gboolean
gst_nonstream_audio_decoder_finish_load (GstNonstreamAudioDecoder * dec,
gboolean load_ok, GstClockTime initial_position,
gboolean send_stream_start);
static gboolean gst_nonstream_audio_decoder_start_task (GstNonstreamAudioDecoder
* dec);
static gboolean gst_nonstream_audio_decoder_stop_task (GstNonstreamAudioDecoder
* dec);
static gboolean
gst_nonstream_audio_decoder_switch_to_subsong (GstNonstreamAudioDecoder * dec,
guint new_subsong, guint32 const *seqnum);
static void gst_nonstream_audio_decoder_update_toc (GstNonstreamAudioDecoder *
dec, GstNonstreamAudioDecoderClass * klass);
static void
gst_nonstream_audio_decoder_update_subsong_duration (GstNonstreamAudioDecoder *
dec, GstClockTime duration);
static void
gst_nonstream_audio_decoder_output_new_segment (GstNonstreamAudioDecoder * dec,
GstClockTime start_position);
static gboolean gst_nonstream_audio_decoder_do_seek (GstNonstreamAudioDecoder *
dec, GstEvent * event);
static GstTagList
* gst_nonstream_audio_decoder_add_main_tags (GstNonstreamAudioDecoder * dec,
GstTagList * tags);
static void gst_nonstream_audio_decoder_output_task (GstNonstreamAudioDecoder *
dec);
static char const *get_seek_type_name (GstSeekType seek_type);
static GType gst_nonstream_audio_decoder_output_mode_get_type (void);
#define GST_TYPE_NONSTREAM_AUDIO_DECODER_OUTPUT_MODE (gst_nonstream_audio_decoder_output_mode_get_type())
static GType gst_nonstream_audio_decoder_subsong_mode_get_type (void);
#define GST_TYPE_NONSTREAM_AUDIO_DECODER_SUBSONG_MODE (gst_nonstream_audio_decoder_subsong_mode_get_type())
static GType
gst_nonstream_audio_decoder_output_mode_get_type (void)
{
static GType gst_nonstream_audio_decoder_output_mode_type = 0;
if (!gst_nonstream_audio_decoder_output_mode_type) {
static GEnumValue output_mode_values[] = {
{GST_NONSTREAM_AUDIO_OUTPUT_MODE_LOOPING, "Looping output", "looping"},
{GST_NONSTREAM_AUDIO_OUTPUT_MODE_STEADY, "Steady output", "steady"},
{0, NULL, NULL},
};
gst_nonstream_audio_decoder_output_mode_type =
g_enum_register_static ("GstNonstreamAudioOutputMode",
output_mode_values);
}
return gst_nonstream_audio_decoder_output_mode_type;
}
static GType
gst_nonstream_audio_decoder_subsong_mode_get_type (void)
{
static GType gst_nonstream_audio_decoder_subsong_mode_type = 0;
if (!gst_nonstream_audio_decoder_subsong_mode_type) {
static GEnumValue subsong_mode_values[] = {
{GST_NONSTREAM_AUDIO_SUBSONG_MODE_SINGLE, "Play single subsong",
"single"},
{GST_NONSTREAM_AUDIO_SUBSONG_MODE_ALL, "Play all subsongs", "all"},
{GST_NONSTREAM_AUDIO_SUBSONG_MODE_DECODER_DEFAULT,
"Decoder specific default behavior", "default"},
{0, NULL, NULL},
};
gst_nonstream_audio_decoder_subsong_mode_type =
g_enum_register_static ("GstNonstreamAudioSubsongMode",
subsong_mode_values);
}
return gst_nonstream_audio_decoder_subsong_mode_type;
}
/* Manually defining the GType instead of using G_DEFINE_TYPE_WITH_CODE()
* because the _init() function needs to be able to access the derived
* class' sink- and srcpads */
GType
gst_nonstream_audio_decoder_get_type (void)
{
static volatile gsize nonstream_audio_decoder_type = 0;
if (g_once_init_enter (&nonstream_audio_decoder_type)) {
GType type_;
static const GTypeInfo nonstream_audio_decoder_info = {
sizeof (GstNonstreamAudioDecoderClass),
NULL,
NULL,
(GClassInitFunc) gst_nonstream_audio_decoder_class_init,
NULL,
NULL,
sizeof (GstNonstreamAudioDecoder),
0,
(GInstanceInitFunc) gst_nonstream_audio_decoder_init,
NULL
};
type_ = g_type_register_static (GST_TYPE_ELEMENT,
"GstNonstreamAudioDecoder",
&nonstream_audio_decoder_info, G_TYPE_FLAG_ABSTRACT);
g_once_init_leave (&nonstream_audio_decoder_type, type_);
}
return nonstream_audio_decoder_type;
}
static void
gst_nonstream_audio_decoder_class_init (GstNonstreamAudioDecoderClass * klass)
{
GObjectClass *object_class;
GstElementClass *element_class;
object_class = G_OBJECT_CLASS (klass);
element_class = GST_ELEMENT_CLASS (klass);
gst_nonstream_audio_decoder_parent_class = g_type_class_peek_parent (klass);
GST_DEBUG_CATEGORY_INIT (nonstream_audiodecoder_debug,
"nonstreamaudiodecoder", 0, "nonstream audio decoder base class");
object_class->finalize =
GST_DEBUG_FUNCPTR (gst_nonstream_audio_decoder_finalize);
object_class->set_property =
GST_DEBUG_FUNCPTR (gst_nonstream_audio_decoder_set_property);
object_class->get_property =
GST_DEBUG_FUNCPTR (gst_nonstream_audio_decoder_get_property);
element_class->change_state =
GST_DEBUG_FUNCPTR (gst_nonstream_audio_decoder_change_state);
klass->seek = NULL;
klass->tell = NULL;
klass->load_from_buffer = NULL;
klass->load_from_custom = NULL;
klass->get_main_tags = NULL;
klass->get_current_subsong = NULL;
klass->set_current_subsong = NULL;
klass->get_num_subsongs = NULL;
klass->get_subsong_duration = NULL;
klass->get_subsong_tags = NULL;
klass->set_subsong_mode = NULL;
klass->set_num_loops = NULL;
klass->get_num_loops = NULL;
klass->decode = NULL;
klass->negotiate =
GST_DEBUG_FUNCPTR (gst_nonstream_audio_decoder_negotiate_default);
klass->decide_allocation =
GST_DEBUG_FUNCPTR (gst_nonstream_audio_decoder_decide_allocation_default);
klass->propose_allocation =
GST_DEBUG_FUNCPTR
(gst_nonstream_audio_decoder_propose_allocation_default);
klass->loads_from_sinkpad = TRUE;
g_object_class_install_property (object_class,
PROP_CURRENT_SUBSONG,
g_param_spec_uint ("current-subsong",
"Currently active subsong",
"Subsong that is currently selected for playback",
0, G_MAXUINT,
DEFAULT_CURRENT_SUBSONG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
g_object_class_install_property (object_class,
PROP_SUBSONG_MODE,
g_param_spec_enum ("subsong-mode",
"Subsong mode",
"Mode which defines how to treat subsongs",
GST_TYPE_NONSTREAM_AUDIO_DECODER_SUBSONG_MODE,
DEFAULT_SUBSONG_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
g_object_class_install_property (object_class,
PROP_NUM_LOOPS,
g_param_spec_int ("num-loops",
"Number of playback loops",
"Number of times a playback loop shall be executed (special values: 0 = no looping; -1 = infinite loop)",
-1, G_MAXINT,
DEFAULT_NUM_LOOPS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
g_object_class_install_property (object_class,
PROP_OUTPUT_MODE,
g_param_spec_enum ("output-mode",
"Output mode",
"Which mode playback shall use when a loop is encountered; looping = reset position to start of loop, steady = do not reset position",
GST_TYPE_NONSTREAM_AUDIO_DECODER_OUTPUT_MODE,
DEFAULT_OUTPUT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
);
}
static void
gst_nonstream_audio_decoder_init (GstNonstreamAudioDecoder * dec,
GstNonstreamAudioDecoderClass * klass)
{
GstPadTemplate *pad_template;
/* These are set here, not in gst_nonstream_audio_decoder_set_initial_state(),
* because these are values for the properties; they are not supposed to be
* reset in the READY->NULL state change */
dec->current_subsong = DEFAULT_CURRENT_SUBSONG;
dec->subsong_mode = DEFAULT_SUBSONG_MODE;
dec->output_mode = DEFAULT_OUTPUT_MODE;
dec->num_loops = DEFAULT_NUM_LOOPS;
/* Calling this here, not in the NULL->READY state change,
* to make sure get_property calls return valid values */
gst_nonstream_audio_decoder_set_initial_state (dec);
dec->input_data_adapter = gst_adapter_new ();
g_mutex_init (&(dec->mutex));
{
/* set up src pad */
pad_template =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "src");
g_return_if_fail (pad_template != NULL); /* derived class is supposed to define a src pad template */
dec->srcpad = gst_pad_new_from_template (pad_template, "src");
gst_pad_set_event_function (dec->srcpad,
GST_DEBUG_FUNCPTR (gst_nonstream_audio_decoder_src_event));
gst_pad_set_query_function (dec->srcpad,
GST_DEBUG_FUNCPTR (gst_nonstream_audio_decoder_src_query));
gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
}
if (klass->loads_from_sinkpad) {
/* set up sink pad if this class loads from a sinkpad */
pad_template =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink");
g_return_if_fail (pad_template != NULL); /* derived class is supposed to define a sink pad template */
dec->sinkpad = gst_pad_new_from_template (pad_template, "sink");
gst_pad_set_event_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_nonstream_audio_decoder_sink_event));
gst_pad_set_query_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_nonstream_audio_decoder_sink_query));
gst_pad_set_chain_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_nonstream_audio_decoder_chain));
gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
}
}
static void
gst_nonstream_audio_decoder_finalize (GObject * object)
{
GstNonstreamAudioDecoder *dec = GST_NONSTREAM_AUDIO_DECODER (object);
g_mutex_clear (&(dec->mutex));
g_object_unref (G_OBJECT (dec->input_data_adapter));
G_OBJECT_CLASS (gst_nonstream_audio_decoder_parent_class)->finalize (object);
}
static void
gst_nonstream_audio_decoder_set_property (GObject * object, guint prop_id,
GValue const *value, GParamSpec * pspec)
{
GstNonstreamAudioDecoder *dec = GST_NONSTREAM_AUDIO_DECODER (object);
GstNonstreamAudioDecoderClass *klass =
GST_NONSTREAM_AUDIO_DECODER_GET_CLASS (dec);
switch (prop_id) {
case PROP_OUTPUT_MODE:
{
GstNonstreamAudioOutputMode new_output_mode;
new_output_mode = g_value_get_enum (value);
g_assert (klass->get_supported_output_modes);
if ((klass->get_supported_output_modes (dec) & (1u << new_output_mode)) ==
0) {
GST_WARNING_OBJECT (dec,
"could not set output mode to %s (not supported by subclass)",
(new_output_mode ==
GST_NONSTREAM_AUDIO_OUTPUT_MODE_STEADY) ? "steady" : "looping");
break;
}
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
if (new_output_mode != dec->output_mode) {
gboolean proceed = TRUE;
if (dec->loaded_mode) {
GstClockTime cur_position;
if (klass->set_output_mode != NULL) {
if (klass->set_output_mode (dec, new_output_mode, &cur_position))
proceed = TRUE;
else {
proceed = FALSE;
GST_WARNING_OBJECT (dec, "switching to new output mode failed");
}
} else {
GST_DEBUG_OBJECT (dec,
"cannot call set_output_mode, since it is NULL");
proceed = FALSE;
}
if (proceed) {
gst_nonstream_audio_decoder_output_new_segment (dec, cur_position);
dec->output_mode = new_output_mode;
}
}
if (proceed) {
/* store output mode in case the property is set before the media got loaded */
dec->output_mode = new_output_mode;
}
}
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
break;
}
case PROP_CURRENT_SUBSONG:
{
guint new_subsong = g_value_get_uint (value);
gst_nonstream_audio_decoder_switch_to_subsong (dec, new_subsong, NULL);
break;
}
case PROP_SUBSONG_MODE:
{
GstNonstreamAudioSubsongMode new_subsong_mode = g_value_get_enum (value);
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
if (new_subsong_mode != dec->subsong_mode) {
gboolean proceed = TRUE;
if (dec->loaded_mode) {
GstClockTime cur_position;
if (klass->set_subsong_mode != NULL) {
if (klass->set_subsong_mode (dec, new_subsong_mode, &cur_position))
proceed = TRUE;
else {
proceed = FALSE;
GST_WARNING_OBJECT (dec, "switching to new subsong mode failed");
}
} else {
GST_DEBUG_OBJECT (dec,
"cannot call set_subsong_mode, since it is NULL");
proceed = FALSE;
}
if (proceed) {
if (GST_CLOCK_TIME_IS_VALID (cur_position))
gst_nonstream_audio_decoder_output_new_segment (dec,
cur_position);
dec->subsong_mode = new_subsong_mode;
}
}
if (proceed) {
/* store subsong mode in case the property is set before the media got loaded */
dec->subsong_mode = new_subsong_mode;
}
}
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
break;
}
case PROP_NUM_LOOPS:
{
gint new_num_loops = g_value_get_int (value);
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
if (new_num_loops != dec->num_loops) {
if (dec->loaded_mode) {
if (klass->set_num_loops != NULL) {
if (!(klass->set_num_loops (dec, new_num_loops)))
GST_WARNING_OBJECT (dec, "setting number of loops to %u failed",
new_num_loops);
} else
GST_DEBUG_OBJECT (dec,
"cannot call set_num_loops, since it is NULL");
}
/* store number of loops in case the property is set before the media got loaded */
dec->num_loops = new_num_loops;
}
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_nonstream_audio_decoder_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstNonstreamAudioDecoder *dec = GST_NONSTREAM_AUDIO_DECODER (object);
switch (prop_id) {
case PROP_OUTPUT_MODE:
{
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
g_value_set_enum (value, dec->output_mode);
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
break;
}
case PROP_CURRENT_SUBSONG:
{
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
g_value_set_uint (value, dec->current_subsong);
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
break;
}
case PROP_SUBSONG_MODE:
{
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
g_value_set_enum (value, dec->subsong_mode);
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
break;
}
case PROP_NUM_LOOPS:
{
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
g_value_set_int (value, dec->num_loops);
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_nonstream_audio_decoder_change_state (GstElement * element,
GstStateChange transition)
{
GstStateChangeReturn ret;
ret =
GST_ELEMENT_CLASS (gst_nonstream_audio_decoder_parent_class)->change_state
(element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
{
GstNonstreamAudioDecoder *dec = GST_NONSTREAM_AUDIO_DECODER (element);
GstNonstreamAudioDecoderClass *klass =
GST_NONSTREAM_AUDIO_DECODER_GET_CLASS (dec);
/* For decoders that load with some custom method,
* this is now the time to load
*
* It is done *after* calling the parent class' change_state vfunc,
* since the pad states need to be set up in order for the loading
* to succeed, since it will try to push a new_caps event
* downstream etc. (upwards state changes typically are handled
* *before* calling the parent class' change_state vfunc ; this is
* a special case) */
if (!(klass->loads_from_sinkpad) && !(dec->loaded_mode)) {
gboolean ret;
/* load_from_custom is required if loads_from_sinkpad is FALSE */
g_assert (klass->load_from_custom != NULL);
ret = gst_nonstream_audio_decoder_load_from_custom (dec);
if (!ret) {
GST_ERROR_OBJECT (dec, "loading from custom source failed");
return GST_STATE_CHANGE_FAILURE;
}
if (!gst_nonstream_audio_decoder_start_task (dec))
return GST_STATE_CHANGE_FAILURE;
}
break;
}
case GST_STATE_CHANGE_PAUSED_TO_READY:
{
GstNonstreamAudioDecoder *dec = GST_NONSTREAM_AUDIO_DECODER (element);
if (!gst_nonstream_audio_decoder_stop_task (dec))
return GST_STATE_CHANGE_FAILURE;
break;
}
case GST_STATE_CHANGE_READY_TO_NULL:
{
GstNonstreamAudioDecoder *dec = GST_NONSTREAM_AUDIO_DECODER (element);
/* In the READY->NULL state change, reset the decoder to an
* initial state ensure it can be used for a fresh new session */
gst_nonstream_audio_decoder_cleanup_state (dec);
break;
}
default:
break;
}
return ret;
}
static gboolean
gst_nonstream_audio_decoder_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
gboolean res = FALSE;
GstNonstreamAudioDecoder *dec = GST_NONSTREAM_AUDIO_DECODER (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:
{
/* Upstream sends in a byte segment, which is uninteresting here,
* since a custom segment event is generated anyway */
gst_event_unref (event);
res = TRUE;
break;
}
case GST_EVENT_EOS:
{
gsize avail_size;
GstBuffer *adapter_buffer;
if (dec->loaded_mode) {
/* If media has already been loaded, then the decoder
* task has been started; the EOS event can be ignored */
GST_DEBUG_OBJECT (dec,
"EOS received after media was loaded -> ignoring");
res = TRUE;
} else {
/* take all data in the input data adapter,
* and try to load the media from it */
avail_size = gst_adapter_available (dec->input_data_adapter);
if (avail_size == 0) {
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("EOS event raised, but no data was received - cannot load anything"));
return FALSE;
}
adapter_buffer =
gst_adapter_take_buffer (dec->input_data_adapter, avail_size);
if (!gst_nonstream_audio_decoder_load_from_buffer (dec, adapter_buffer)) {
return FALSE;
}
res = gst_nonstream_audio_decoder_start_task (dec);
}
break;
}
default:
res = gst_pad_event_default (pad, parent, event);
}
return res;
}
static gboolean
gst_nonstream_audio_decoder_sink_query (GstPad * pad, GstObject * parent,
GstQuery * query)
{
gboolean res = FALSE;
GstNonstreamAudioDecoder *dec;
GstNonstreamAudioDecoderClass *klass;
dec = GST_NONSTREAM_AUDIO_DECODER (parent);
klass = GST_NONSTREAM_AUDIO_DECODER_GET_CLASS (dec);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_ALLOCATION:
{
if (klass->propose_allocation != NULL)
res = klass->propose_allocation (dec, query);
break;
}
default:
res = gst_pad_query_default (pad, parent, query);
}
return res;
}
static GstFlowReturn
gst_nonstream_audio_decoder_chain (G_GNUC_UNUSED GstPad * pad,
GstObject * parent, GstBuffer * buffer)
{
GstFlowReturn flow_ret = GST_FLOW_OK;
GstNonstreamAudioDecoder *dec = GST_NONSTREAM_AUDIO_DECODER (parent);
/* query upstream size in bytes to know how many bytes to expect
* this is a safety measure to prevent the case when upstream never
* reaches EOS (or only after a long time) and we keep loading and
* loading and eventually run out of memory */
if (dec->upstream_size < 0) {
if (!gst_nonstream_audio_decoder_get_upstream_size (dec,
&(dec->upstream_size))) {
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("Cannot load - upstream size (in bytes) could not be determined"));
return GST_FLOW_ERROR;
}
}
if (dec->loaded_mode) {
/* media is already loaded - discard any incoming
* buffers, since they are not needed */
GST_DEBUG_OBJECT (dec, "received data after media was loaded - ignoring");
gst_buffer_unref (buffer);
} else {
/* accumulate data until end-of-stream or the upstream
* size is reached, then load media and commence playback */
gint64 avail_size;
gst_adapter_push (dec->input_data_adapter, buffer);
avail_size = gst_adapter_available (dec->input_data_adapter);
if (avail_size >= dec->upstream_size) {
GstBuffer *adapter_buffer =
gst_adapter_take_buffer (dec->input_data_adapter, avail_size);
if (gst_nonstream_audio_decoder_load_from_buffer (dec, adapter_buffer))
flow_ret =
gst_nonstream_audio_decoder_start_task (dec) ? GST_FLOW_OK :
GST_FLOW_ERROR;
else
flow_ret = GST_FLOW_ERROR;
}
}
return flow_ret;
}
static gboolean
gst_nonstream_audio_decoder_src_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
gboolean res = FALSE;
GstNonstreamAudioDecoder *dec = GST_NONSTREAM_AUDIO_DECODER (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
{
res = gst_nonstream_audio_decoder_do_seek (dec, event);
break;
}
case GST_EVENT_TOC_SELECT:
{
/* NOTE: This event may be received multiple times if it
* was originally sent to a bin containing multiple sink
* elements (for example, playbin). This is OK and does
* not break anything. */
gchar *uid = NULL;
guint subsong_idx = 0;
guint32 seqnum;
gst_event_parse_toc_select (event, &uid);
if ((uid != NULL)
&& (sscanf (uid, "nonstream-subsong-%05u", &subsong_idx) == 1)) {
seqnum = gst_event_get_seqnum (event);
GST_DEBUG_OBJECT (dec,
"received TOC select event (sequence number %" G_GUINT32_FORMAT
"), switching to subsong %u", seqnum, subsong_idx);
gst_nonstream_audio_decoder_switch_to_subsong (dec, subsong_idx,
&seqnum);
}
g_free (uid);
res = TRUE;
break;
}
default:
res = gst_pad_event_default (pad, parent, event);
}
return res;
}
static gboolean
gst_nonstream_audio_decoder_src_query (GstPad * pad, GstObject * parent,
GstQuery * query)
{
gboolean res = FALSE;
GstNonstreamAudioDecoder *dec;
GstNonstreamAudioDecoderClass *klass;
dec = GST_NONSTREAM_AUDIO_DECODER (parent);
klass = GST_NONSTREAM_AUDIO_DECODER_GET_CLASS (dec);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_DURATION:
{
GstFormat format;
GST_TRACE_OBJECT (parent, "duration query");
if (!(dec->loaded_mode)) {
GST_DEBUG_OBJECT (parent,
"cannot respond to duration query: nothing is loaded yet");
break;
}
GST_TRACE_OBJECT (parent, "parsing duration query");
gst_query_parse_duration (query, &format, NULL);
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
if ((format == GST_FORMAT_TIME)
&& (dec->subsong_duration != GST_CLOCK_TIME_NONE)) {
GST_DEBUG_OBJECT (parent,
"responding to query with duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (dec->subsong_duration));
gst_query_set_duration (query, format, dec->subsong_duration);
res = TRUE;
} else if (format != GST_FORMAT_TIME)
GST_DEBUG_OBJECT (parent,
"cannot respond to duration query: format is %s, expected time format",
gst_format_get_name (format));
else if (dec->subsong_duration == GST_CLOCK_TIME_NONE)
GST_DEBUG_OBJECT (parent,
"cannot respond to duration query: no valid subsong duration available");
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
break;
}
case GST_QUERY_POSITION:
{
GstFormat format;
if (!(dec->loaded_mode)) {
GST_DEBUG_OBJECT (parent,
"cannot respond to position query: nothing is loaded yet");
break;
}
if (klass->tell == NULL) {
GST_DEBUG_OBJECT (parent,
"cannot respond to position query: subclass does not have tell() function defined");
break;
}
gst_query_parse_position (query, &format, NULL);
if (format == GST_FORMAT_TIME) {
GstClockTime pos;
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
pos = klass->tell (dec);
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
GST_DEBUG_OBJECT (parent,
"position query received with format TIME -> reporting position %"
GST_TIME_FORMAT, GST_TIME_ARGS (pos));
gst_query_set_position (query, format, pos);
res = TRUE;
} else {
GST_DEBUG_OBJECT (parent,
"position query received with unsupported format %s -> not reporting anything",
gst_format_get_name (format));
}
break;
}
case GST_QUERY_SEEKING:
{
GstFormat fmt;
GstClockTime duration;
if (!dec->loaded_mode) {
GST_DEBUG_OBJECT (parent,
"cannot respond to seeking query: nothing is loaded yet");
break;
}
if (klass->seek == NULL) {
GST_DEBUG_OBJECT (parent,
"cannot respond to seeking query: subclass does not have seek() function defined");
break;
}
gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL);
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
duration = dec->subsong_duration;
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
if (fmt == GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (parent,
"seeking query received with format TIME -> can seek: yes");
gst_query_set_seeking (query, fmt, TRUE, 0, duration);
res = TRUE;
} else {
GST_DEBUG_OBJECT (parent,
"seeking query received with unsupported format %s -> can seek: no",
gst_format_get_name (fmt));
gst_query_set_seeking (query, fmt, FALSE, 0, -1);
res = TRUE;
}
break;
}
default:
res = gst_pad_query_default (pad, parent, query);
}
return res;
}
static void
gst_nonstream_audio_decoder_set_initial_state (GstNonstreamAudioDecoder * dec)
{
dec->upstream_size = -1;
dec->loaded_mode = FALSE;
dec->subsong_duration = GST_CLOCK_TIME_NONE;
dec->output_format_changed = FALSE;
gst_audio_info_init (&(dec->output_audio_info));
dec->num_decoded_samples = 0;
dec->cur_pos_in_samples = 0;
gst_segment_init (&(dec->cur_segment), GST_FORMAT_TIME);
dec->discont = FALSE;
dec->toc = NULL;
dec->allocator = NULL;
}
static void
gst_nonstream_audio_decoder_cleanup_state (GstNonstreamAudioDecoder * dec)
{
gst_adapter_clear (dec->input_data_adapter);
if (dec->allocator != NULL) {
gst_object_unref (dec->allocator);
dec->allocator = NULL;
}
if (dec->toc != NULL) {
gst_toc_unref (dec->toc);
dec->toc = NULL;
}
gst_nonstream_audio_decoder_set_initial_state (dec);
}
static gboolean
gst_nonstream_audio_decoder_negotiate (GstNonstreamAudioDecoder * dec)
{
/* must be called with lock */
GstNonstreamAudioDecoderClass *klass;
gboolean res = TRUE;
klass = GST_NONSTREAM_AUDIO_DECODER_GET_CLASS (dec);
/* protected by a mutex, since the allocator might currently be in use */
if (klass->negotiate != NULL)
res = klass->negotiate (dec);
return res;
}
static gboolean
gst_nonstream_audio_decoder_negotiate_default (GstNonstreamAudioDecoder * dec)
{
/* mutex is locked when this is called */
GstCaps *caps;
GstNonstreamAudioDecoderClass *klass;
gboolean res = TRUE;
GstQuery *query = NULL;
GstAllocator *allocator;
GstAllocationParams allocation_params;
g_return_val_if_fail (GST_IS_NONSTREAM_AUDIO_DECODER (dec), FALSE);
g_return_val_if_fail (GST_AUDIO_INFO_IS_VALID (&(dec->output_audio_info)),
FALSE);
klass = GST_NONSTREAM_AUDIO_DECODER_CLASS (G_OBJECT_GET_CLASS (dec));
caps = gst_audio_info_to_caps (&(dec->output_audio_info));
GST_DEBUG_OBJECT (dec, "setting src caps %" GST_PTR_FORMAT, (gpointer) caps);
res = gst_pad_push_event (dec->srcpad, gst_event_new_caps (caps));
/* clear any pending reconfigure flag */
gst_pad_check_reconfigure (dec->srcpad);
if (!res) {
GST_WARNING_OBJECT (dec, "could not push new caps event downstream");
goto done;
}
GST_TRACE_OBJECT (dec, "src caps set");
dec->output_format_changed = FALSE;
query = gst_query_new_allocation (caps, TRUE);
if (!gst_pad_peer_query (dec->srcpad, query)) {
GST_DEBUG_OBJECT (dec, "didn't get downstream ALLOCATION hints");
}
g_assert (klass->decide_allocation != NULL);
res = klass->decide_allocation (dec, query);
GST_DEBUG_OBJECT (dec, "ALLOCATION (%d) params: %" GST_PTR_FORMAT, res,
(gpointer) query);
if (!res)
goto no_decide_allocation;
/* we got configuration from our peer or the decide_allocation method,
* parse them */
if (gst_query_get_n_allocation_params (query) > 0) {
gst_query_parse_nth_allocation_param (query, 0, &allocator,
&allocation_params);
} else {
allocator = NULL;
gst_allocation_params_init (&allocation_params);
}
if (dec->allocator != NULL)
gst_object_unref (dec->allocator);
dec->allocator = allocator;
dec->allocation_params = allocation_params;
done:
if (query != NULL)
gst_query_unref (query);
gst_caps_unref (caps);
return res;
no_decide_allocation:
{
GST_WARNING_OBJECT (dec, "subclass failed to decide allocation");
goto done;
}
}
static gboolean
gst_nonstream_audio_decoder_decide_allocation_default (G_GNUC_UNUSED
GstNonstreamAudioDecoder * dec, GstQuery * query)
{
GstAllocator *allocator = NULL;
GstAllocationParams params;
gboolean update_allocator;
/* we got configuration from our peer or the decide_allocation method,
* parse them */
if (gst_query_get_n_allocation_params (query) > 0) {
/* try the allocator */
gst_query_parse_nth_allocation_param (query, 0, &allocator, &params);
update_allocator = TRUE;
} else {
allocator = NULL;
gst_allocation_params_init (&params);
update_allocator = FALSE;
}
if (update_allocator)
gst_query_set_nth_allocation_param (query, 0, allocator, &params);
else
gst_query_add_allocation_param (query, allocator, &params);
if (allocator)
gst_object_unref (allocator);
return TRUE;
}
static gboolean
gst_nonstream_audio_decoder_propose_allocation_default (G_GNUC_UNUSED
GstNonstreamAudioDecoder * dec, G_GNUC_UNUSED GstQuery * query)
{
return TRUE;
}
static gboolean
gst_nonstream_audio_decoder_get_upstream_size (GstNonstreamAudioDecoder * dec,
gint64 * length)
{
return gst_pad_peer_query_duration (dec->sinkpad, GST_FORMAT_BYTES, length)
&& (*length >= 0);
}
static gboolean
gst_nonstream_audio_decoder_load_from_buffer (GstNonstreamAudioDecoder * dec,
GstBuffer * buffer)
{
gboolean load_ok;
GstClockTime initial_position;
GstNonstreamAudioDecoderClass *klass;
gboolean ret;
klass = GST_NONSTREAM_AUDIO_DECODER_CLASS (G_OBJECT_GET_CLASS (dec));
g_assert (klass->load_from_buffer != NULL);
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
GST_LOG_OBJECT (dec, "read %" G_GSIZE_FORMAT " bytes from upstream",
gst_buffer_get_size (buffer));
initial_position = 0;
load_ok =
klass->load_from_buffer (dec, buffer, dec->current_subsong,
dec->subsong_mode, &initial_position, &(dec->output_mode),
&(dec->num_loops));
gst_buffer_unref (buffer);
ret =
gst_nonstream_audio_decoder_finish_load (dec, load_ok, initial_position,
FALSE);
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
return ret;
}
static gboolean
gst_nonstream_audio_decoder_load_from_custom (GstNonstreamAudioDecoder * dec)
{
gboolean load_ok;
GstClockTime initial_position;
GstNonstreamAudioDecoderClass *klass;
gboolean ret;
klass = GST_NONSTREAM_AUDIO_DECODER_CLASS (G_OBJECT_GET_CLASS (dec));
g_assert (klass->load_from_custom != NULL);
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
GST_LOG_OBJECT (dec,
"reading song from custom source defined by derived class");
initial_position = 0;
load_ok =
klass->load_from_custom (dec, dec->current_subsong, dec->subsong_mode,
&initial_position, &(dec->output_mode), &(dec->num_loops));
ret =
gst_nonstream_audio_decoder_finish_load (dec, load_ok, initial_position,
TRUE);
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
return ret;
}
static gboolean
gst_nonstream_audio_decoder_finish_load (GstNonstreamAudioDecoder * dec,
gboolean load_ok, GstClockTime initial_position, gboolean send_stream_start)
{
/* must be called with lock */
GstNonstreamAudioDecoderClass *klass =
GST_NONSTREAM_AUDIO_DECODER_CLASS (G_OBJECT_GET_CLASS (dec));
GST_TRACE_OBJECT (dec, "enter finish_load");
/* Prerequisites */
if (!load_ok) {
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Loading failed"));
return FALSE;
}
if (!GST_AUDIO_INFO_IS_VALID (&(dec->output_audio_info))) {
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("Audio info is invalid after loading"));
return FALSE;
}
/* Log the number of available subsongs */
if (klass->get_num_subsongs != NULL)
GST_DEBUG_OBJECT (dec, "%u subsong(s) available",
klass->get_num_subsongs (dec));
/* Set the current subsong (or use the default value) */
if (klass->get_current_subsong != NULL) {
GST_TRACE_OBJECT (dec, "requesting current subsong");
dec->current_subsong = klass->get_current_subsong (dec);
}
/* Handle the subsong duration */
if (klass->get_subsong_duration != NULL) {
GstClockTime duration;
GST_TRACE_OBJECT (dec, "requesting subsong duration");
duration = klass->get_subsong_duration (dec, dec->current_subsong);
gst_nonstream_audio_decoder_update_subsong_duration (dec, duration);
}
/* Send tags downstream (if some exist) */
if (klass->get_subsong_tags != NULL) {
/* Subsong tags available */
GstTagList *tags;
GST_TRACE_OBJECT (dec, "requesting subsong tags");
tags = klass->get_subsong_tags (dec, dec->current_subsong);
if (tags != NULL)
tags = gst_nonstream_audio_decoder_add_main_tags (dec, tags);
if (tags != NULL)
gst_pad_push_event (dec->srcpad, gst_event_new_tag (tags));
} else {
/* No subsong tags - just send main tags out */
GstTagList *tags = gst_tag_list_new_empty ();
tags = gst_nonstream_audio_decoder_add_main_tags (dec, tags);
gst_pad_push_event (dec->srcpad, gst_event_new_tag (tags));
}
/* Send stream start downstream if requested */
if (send_stream_start) {
gchar *stream_id;
GstEvent *event;
stream_id =
gst_pad_create_stream_id (dec->srcpad, GST_ELEMENT_CAST (dec), NULL);
GST_DEBUG_OBJECT (dec, "pushing STREAM_START with stream id \"%s\"",
stream_id);
event = gst_event_new_stream_start (stream_id);
gst_event_set_group_id (event, gst_util_group_id_next ());
gst_pad_push_event (dec->srcpad, event);
g_free (stream_id);
}
/* Update the table of contents */
gst_nonstream_audio_decoder_update_toc (dec, klass);
/* Negotiate output caps and an allocator */
GST_TRACE_OBJECT (dec, "negotiating caps and allocator");
if (!gst_nonstream_audio_decoder_negotiate (dec)) {
GST_ERROR_OBJECT (dec, "negotiation failed - aborting load");
return FALSE;
}
/* Send new segment downstream */
gst_nonstream_audio_decoder_output_new_segment (dec, initial_position);
dec->loaded_mode = TRUE;
GST_TRACE_OBJECT (dec, "exit finish_load");
return TRUE;
}
static gboolean
gst_nonstream_audio_decoder_start_task (GstNonstreamAudioDecoder * dec)
{
if (!gst_pad_start_task (dec->srcpad,
(GstTaskFunction) gst_nonstream_audio_decoder_output_task, dec,
NULL)) {
GST_ERROR_OBJECT (dec, "could not start decoder output task");
return FALSE;
} else
return TRUE;
}
static gboolean
gst_nonstream_audio_decoder_stop_task (GstNonstreamAudioDecoder * dec)
{
if (!gst_pad_stop_task (dec->srcpad)) {
GST_ERROR_OBJECT (dec, "could not stop decoder output task");
return FALSE;
} else
return TRUE;
}
static gboolean
gst_nonstream_audio_decoder_switch_to_subsong (GstNonstreamAudioDecoder * dec,
guint new_subsong, guint32 const *seqnum)
{
gboolean ret = TRUE;
GstNonstreamAudioDecoderClass *klass =
GST_NONSTREAM_AUDIO_DECODER_GET_CLASS (dec);
if (klass->set_current_subsong == NULL) {
/* If set_current_subsong wasn't set by the subclass, then
* subsongs are not supported. It is not an error if this
* function is called in that case, since it might happen
* because the current-subsong property was set (and since
* this is a base class property, it is always available). */
GST_DEBUG_OBJECT (dec, "cannot call set_current_subsong, since it is NULL");
goto finish;
}
if (dec->loaded_mode) {
GstEvent *fevent;
GstClockTime new_position;
GstClockTime new_subsong_duration = GST_CLOCK_TIME_NONE;
/* Check if (a) new_subsong is already the current subsong
* and (b) if new_subsong exceeds the number of available
* subsongs. Do this here, when the song is loaded,
* because prior to loading, the number of subsong is usually
* not known (and the loading process might choose a specific
* subsong to be the current one at the start of playback). */
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
if (new_subsong == dec->current_subsong) {
GST_DEBUG_OBJECT (dec,
"subsong %u is already the current subsong - ignoring call",
new_subsong);
goto finish_unlock;
}
if (klass->get_num_subsongs) {
guint num_subsongs = klass->get_num_subsongs (dec);
if (new_subsong >= num_subsongs) {
GST_WARNING_OBJECT (dec,
"subsong %u is out of bounds (there are %u subsongs) - not switching",
new_subsong, num_subsongs);
goto finish_unlock;
}
}
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
/* Switching subsongs during playback is very similar to a
* flushing seek. Therefore, the stream lock must be taken,
* flush-start/flush-stop events have to be sent, and
* the pad task has to be restarted. */
fevent = gst_event_new_flush_start ();
if (seqnum != NULL) {
gst_event_set_seqnum (fevent, *seqnum);
GST_DEBUG_OBJECT (dec,
"sending flush start event with sequence number %" G_GUINT32_FORMAT,
*seqnum);
} else
GST_DEBUG_OBJECT (dec, "sending flush start event (no sequence number)");
gst_pad_push_event (dec->srcpad, gst_event_ref (fevent));
/* unlock upstream pull_range */
if (klass->loads_from_sinkpad)
gst_pad_push_event (dec->sinkpad, fevent);
else
gst_event_unref (fevent);
GST_PAD_STREAM_LOCK (dec->srcpad);
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
if (!(klass->set_current_subsong (dec, new_subsong, &new_position))) {
/* Switch failed. Do _not_ exit early from here - playback must
* continue from the current subsong, and it cannot do that if
* we exit here. Try getting the current position and proceed as
* if the switch succeeded (but set the return value to FALSE.) */
ret = FALSE;
if (klass->tell)
new_position = klass->tell (dec);
else
new_position = 0;
GST_WARNING_OBJECT (dec, "switching to new subsong %u failed",
new_subsong);
}
/* Flushing seek resets the base time, which means num_decoded_samples
* needs to be set to 0, since it defines the segment.base value */
dec->num_decoded_samples = 0;
fevent = gst_event_new_flush_stop (TRUE);
if (seqnum != NULL) {
gst_event_set_seqnum (fevent, *seqnum);
GST_DEBUG_OBJECT (dec,
"sending flush stop event with sequence number %" G_GUINT32_FORMAT,
*seqnum);
} else
GST_DEBUG_OBJECT (dec, "sending flush stop event (no sequence number)");
gst_pad_push_event (dec->srcpad, gst_event_ref (fevent));
/* unlock upstream pull_range */
if (klass->loads_from_sinkpad)
gst_pad_push_event (dec->sinkpad, fevent);
else
gst_event_unref (fevent);
/* use the new subsong's duration (if one exists) */
if (klass->get_subsong_duration != NULL)
new_subsong_duration = klass->get_subsong_duration (dec, new_subsong);
gst_nonstream_audio_decoder_update_subsong_duration (dec,
new_subsong_duration);
/* create a new segment for the new subsong */
gst_nonstream_audio_decoder_output_new_segment (dec, new_position);
/* use the new subsong's tags (if any exist) */
if (klass->get_subsong_tags != NULL) {
GstTagList *subsong_tags = klass->get_subsong_tags (dec, new_subsong);
if (subsong_tags != NULL)
subsong_tags =
gst_nonstream_audio_decoder_add_main_tags (dec, subsong_tags);
if (subsong_tags != NULL)
gst_pad_push_event (dec->srcpad, gst_event_new_tag (subsong_tags));
}
GST_DEBUG_OBJECT (dec, "successfully switched to new subsong %u",
new_subsong);
dec->current_subsong = new_subsong;
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
/* Subsong has been switched, and all necessary events have been
* pushed downstream. Restart srcpad task. */
gst_nonstream_audio_decoder_start_task (dec);
/* Unlock stream, we are done */
GST_PAD_STREAM_UNLOCK (dec->srcpad);
} else {
/* If song hasn't been loaded yet, then playback cannot currently
* been happening. In this case, a "switch" is simple - just store
* the current subsong index. When the song is loaded, it will
* start playing this subsong. */
GST_DEBUG_OBJECT (dec,
"playback hasn't started yet - storing subsong index %u as the current subsong",
new_subsong);
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
dec->current_subsong = new_subsong;
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
}
finish:
return ret;
finish_unlock:
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
goto finish;
}
static void
gst_nonstream_audio_decoder_update_toc (GstNonstreamAudioDecoder * dec,
GstNonstreamAudioDecoderClass * klass)
{
/* must be called with lock */
guint num_subsongs, i;
if (dec->toc != NULL) {
gst_toc_unref (dec->toc);
dec->toc = NULL;
}
if (klass->get_num_subsongs == NULL)
return;
num_subsongs = klass->get_num_subsongs (dec);
if (num_subsongs <= 1) {
GST_DEBUG_OBJECT (dec, "no need for a TOC since there is only one subsong");
return;
}
dec->toc = gst_toc_new (GST_TOC_SCOPE_GLOBAL);
if (klass->get_main_tags) {
GstTagList *main_tags = klass->get_main_tags (dec);
if (main_tags)
gst_toc_set_tags (dec->toc, main_tags);
}
for (i = 0; i < num_subsongs; ++i) {
gchar *uid;
GstTocEntry *entry;
GstClockTime duration;
GstTagList *tags;
duration =
(klass->get_subsong_duration !=
NULL) ? klass->get_subsong_duration (dec, i) : GST_CLOCK_TIME_NONE;
tags =
(klass->get_subsong_tags != NULL) ? klass->get_subsong_tags (dec,
i) : NULL;
if (!tags)
tags = gst_tag_list_new_empty ();
uid = g_strdup_printf ("nonstream-subsong-%05u", i);
entry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_TRACK, uid);
/* Set the UID as title tag for TOC entry if no title already present */
gst_tag_list_add (tags, GST_TAG_MERGE_KEEP, GST_TAG_TITLE, uid, NULL);
/* Set the subsong duration as duration tag for TOC entry if no duration already present */
if (duration != GST_CLOCK_TIME_NONE)
gst_tag_list_add (tags, GST_TAG_MERGE_KEEP, GST_TAG_DURATION, duration,
NULL);
/* FIXME: TOC does not allow GST_CLOCK_TIME_NONE as a stop value */
if (duration == GST_CLOCK_TIME_NONE)
duration = G_MAXINT64;
/* Subsongs always start at 00:00 */
gst_toc_entry_set_start_stop_times (entry, 0, duration);
gst_toc_entry_set_tags (entry, tags);
/* NOTE: *not* adding loop count via gst_toc_entry_set_loop(), since
* in GstNonstreamAudioDecoder, looping is a playback property, not
* a property of the subsongs themselves */
GST_DEBUG_OBJECT (dec,
"new toc entry: uid: \"%s\" duration: %" GST_TIME_FORMAT " tags: %"
GST_PTR_FORMAT, uid, GST_TIME_ARGS (duration), (gpointer) tags);
gst_toc_append_entry (dec->toc, entry);
g_free (uid);
}
gst_pad_push_event (dec->srcpad, gst_event_new_toc (dec->toc, FALSE));
}
static void
gst_nonstream_audio_decoder_update_subsong_duration (GstNonstreamAudioDecoder *
dec, GstClockTime duration)
{
/* must be called with lock */
dec->subsong_duration = duration;
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
gst_element_post_message (GST_ELEMENT (dec),
gst_message_new_duration_changed (GST_OBJECT (dec)));
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
}
static void
gst_nonstream_audio_decoder_output_new_segment (GstNonstreamAudioDecoder * dec,
GstClockTime start_position)
{
/* must be called with lock */
GstSegment segment;
gst_segment_init (&segment, GST_FORMAT_TIME);
segment.base =
gst_util_uint64_scale_int (dec->num_decoded_samples, GST_SECOND,
dec->output_audio_info.rate);
segment.start = 0;
segment.time = start_position;
segment.offset = 0;
segment.position = 0;
/* note that num_decoded_samples isn't being reset; it is the
* analogue to the segment base value, and thus is supposed to
* monotonically increase, except for when a flushing seek happens
* (since a flushing seek is supposed to be a fresh restart for
* the whole pipeline) */
dec->cur_pos_in_samples = 0;
/* stop/duration members are not set, on purpose - in case of loops,
* new segments will be generated, which automatically put an implicit
* end on the current segment (the segment implicitly "ends" when the
* new one starts), and having a stop value might cause very slight
* gaps occasionally due to slight jitter in the calculation of
* base times etc. */
GST_DEBUG_OBJECT (dec,
"output new segment with base %" GST_TIME_FORMAT " time %"
GST_TIME_FORMAT, GST_TIME_ARGS (segment.base),
GST_TIME_ARGS (segment.time));
dec->cur_segment = segment;
dec->discont = TRUE;
gst_pad_push_event (dec->srcpad, gst_event_new_segment (&segment));
}
static gboolean
gst_nonstream_audio_decoder_do_seek (GstNonstreamAudioDecoder * dec,
GstEvent * event)
{
gboolean res;
gdouble rate;
GstFormat format;
GstSeekFlags flags;
GstSeekType start_type, stop_type;
GstClockTime new_position;
gint64 start, stop;
GstSegment segment;
guint32 seqnum;
gboolean flush;
GstNonstreamAudioDecoderClass *klass =
GST_NONSTREAM_AUDIO_DECODER_GET_CLASS (dec);
if (klass->seek == NULL) {
GST_DEBUG_OBJECT (dec,
"cannot seek: subclass does not have seek() function defined");
return FALSE;
}
if (!dec->loaded_mode) {
GST_DEBUG_OBJECT (dec, "nothing loaded yet - cannot seek");
return FALSE;
}
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
if (!GST_AUDIO_INFO_IS_VALID (&(dec->output_audio_info))) {
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
GST_DEBUG_OBJECT (dec, "no valid output audioinfo present - cannot seek");
return FALSE;
}
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
GST_DEBUG_OBJECT (dec, "starting seek");
gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start,
&stop_type, &stop);
seqnum = gst_event_get_seqnum (event);
GST_DEBUG_OBJECT (dec,
"seek event data: "
"rate %f format %s "
"start type %s start %" GST_TIME_FORMAT " "
"stop type %s stop %" GST_TIME_FORMAT,
rate, gst_format_get_name (format),
get_seek_type_name (start_type), GST_TIME_ARGS (start),
get_seek_type_name (stop_type), GST_TIME_ARGS (stop)
);
if (format != GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (dec, "seeking is only supported in TIME format");
return FALSE;
}
if (rate < 0) {
GST_DEBUG_OBJECT (dec, "only positive seek rates are supported");
return FALSE;
}
flush = ((flags & GST_SEEK_FLAG_FLUSH) == GST_SEEK_FLAG_FLUSH);
if (flush) {
GstEvent *fevent = gst_event_new_flush_start ();
gst_event_set_seqnum (fevent, seqnum);
GST_DEBUG_OBJECT (dec,
"sending flush start event with sequence number %" G_GUINT32_FORMAT,
seqnum);
gst_pad_push_event (dec->srcpad, gst_event_ref (fevent));
/* unlock upstream pull_range */
if (klass->loads_from_sinkpad)
gst_pad_push_event (dec->sinkpad, fevent);
else
gst_event_unref (fevent);
} else
gst_pad_pause_task (dec->srcpad);
GST_PAD_STREAM_LOCK (dec->srcpad);
segment = dec->cur_segment;
if (!gst_segment_do_seek (&segment,
rate, format, flags, start_type, start, stop_type, stop, NULL)) {
GST_DEBUG_OBJECT (dec, "could not seek in segment");
GST_PAD_STREAM_UNLOCK (dec->srcpad);
return FALSE;
}
GST_DEBUG_OBJECT (dec,
"segment data: "
"seek event data: "
"rate %f applied rate %f "
"format %s "
"base %" GST_TIME_FORMAT " "
"offset %" GST_TIME_FORMAT " "
"start %" GST_TIME_FORMAT " "
"stop %" GST_TIME_FORMAT " "
"time %" GST_TIME_FORMAT " "
"position %" GST_TIME_FORMAT " "
"duration %" GST_TIME_FORMAT,
segment.rate, segment.applied_rate,
gst_format_get_name (segment.format),
GST_TIME_ARGS (segment.base),
GST_TIME_ARGS (segment.offset),
GST_TIME_ARGS (segment.start),
GST_TIME_ARGS (segment.stop),
GST_TIME_ARGS (segment.time),
GST_TIME_ARGS (segment.position), GST_TIME_ARGS (segment.duration)
);
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
new_position = segment.position;
res = klass->seek (dec, &new_position);
segment.position = new_position;
dec->cur_segment = segment;
dec->cur_pos_in_samples =
gst_util_uint64_scale_int (dec->cur_segment.position,
dec->output_audio_info.rate, GST_SECOND);
dec->num_decoded_samples = 0;
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
if (flush) {
GstEvent *fevent = gst_event_new_flush_stop (TRUE);
gst_event_set_seqnum (fevent, seqnum);
GST_DEBUG_OBJECT (dec,
"sending flush stop event with sequence number %" G_GUINT32_FORMAT,
seqnum);
gst_pad_push_event (dec->srcpad, gst_event_ref (fevent));
if (klass->loads_from_sinkpad)
gst_pad_push_event (dec->sinkpad, fevent);
else
gst_event_unref (fevent);
}
if (res) {
if (flags & GST_SEEK_FLAG_SEGMENT) {
GST_DEBUG_OBJECT (dec, "posting SEGMENT_START message");
gst_element_post_message (GST_ELEMENT (dec),
gst_message_new_segment_start (GST_OBJECT (dec),
GST_FORMAT_TIME, segment.start)
);
}
gst_pad_push_event (dec->srcpad, gst_event_new_segment (&segment));
GST_INFO_OBJECT (dec, "seek succeeded");
gst_nonstream_audio_decoder_start_task (dec);
} else {
GST_WARNING_OBJECT (dec, "seek failed");
}
GST_PAD_STREAM_UNLOCK (dec->srcpad);
gst_event_unref (event);
return res;
}
static GstTagList *
gst_nonstream_audio_decoder_add_main_tags (GstNonstreamAudioDecoder * dec,
GstTagList * tags)
{
GstNonstreamAudioDecoderClass *klass =
GST_NONSTREAM_AUDIO_DECODER_GET_CLASS (dec);
if (!klass->get_main_tags)
return tags;
tags = gst_tag_list_make_writable (tags);
if (tags) {
GstClockTime duration;
GstTagList *main_tags;
/* Get main tags. If some exist, merge them with the given tags,
* and return the merged result. Otherwise, just return the given tags. */
main_tags = klass->get_main_tags (dec);
if (main_tags) {
tags = gst_tag_list_merge (main_tags, tags, GST_TAG_MERGE_REPLACE);
gst_tag_list_unref (main_tags);
}
/* Add subsong duration if available */
duration = dec->subsong_duration;
if (GST_CLOCK_TIME_IS_VALID (duration))
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE, GST_TAG_DURATION, duration,
NULL);
return tags;
} else {
GST_ERROR_OBJECT (dec, "could not make subsong tags writable");
return NULL;
}
}
static void
gst_nonstream_audio_decoder_output_task (GstNonstreamAudioDecoder * dec)
{
GstFlowReturn flow;
GstBuffer *outbuf;
guint num_samples;
GstNonstreamAudioDecoderClass *klass;
klass = GST_NONSTREAM_AUDIO_DECODER_CLASS (G_OBJECT_GET_CLASS (dec));
g_assert (klass->decode != NULL);
GST_NONSTREAM_AUDIO_DECODER_LOCK_MUTEX (dec);
/* perform the actual decoding */
if (!(klass->decode (dec, &outbuf, &num_samples))) {
/* EOS case */
GST_INFO_OBJECT (dec, "decode() reports end -> sending EOS event");
gst_pad_push_event (dec->srcpad, gst_event_new_eos ());
goto pause_unlock;
}
if (outbuf == NULL) {
GST_ERROR_OBJECT (outbuf, "decode() produced NULL buffer");
goto pause_unlock;
}
/* set the buffer's metadata */
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (num_samples, GST_SECOND,
dec->output_audio_info.rate);
GST_BUFFER_OFFSET (outbuf) = dec->cur_pos_in_samples;
GST_BUFFER_OFFSET_END (outbuf) = dec->cur_pos_in_samples + num_samples;
GST_BUFFER_PTS (outbuf) =
gst_util_uint64_scale_int (dec->cur_pos_in_samples, GST_SECOND,
dec->output_audio_info.rate);
GST_BUFFER_DTS (outbuf) = GST_BUFFER_PTS (outbuf);
if (G_UNLIKELY (dec->discont)) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
dec->discont = FALSE;
}
GST_LOG_OBJECT (dec,
"output buffer stats: num_samples = %u duration = %" GST_TIME_FORMAT
" cur_pos_in_samples = %" G_GUINT64_FORMAT " timestamp = %"
GST_TIME_FORMAT, num_samples,
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), dec->cur_pos_in_samples,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf))
);
/* increment sample counters */
dec->cur_pos_in_samples += num_samples;
dec->num_decoded_samples += num_samples;
/* the decode() call might have set a new output format -> renegotiate
* before sending the new buffer downstream */
if (G_UNLIKELY (dec->output_format_changed ||
(GST_AUDIO_INFO_IS_VALID (&(dec->output_audio_info))
&& gst_pad_check_reconfigure (dec->srcpad))
)) {
if (!gst_nonstream_audio_decoder_negotiate (dec)) {
gst_buffer_unref (outbuf);
GST_LOG_OBJECT (dec, "could not push output buffer: negotiation failed");
goto pause_unlock;
}
}
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
/* push new samples downstream
* no need to unref buffer - gst_pad_push() does it in
* all cases (success and failure) */
flow = gst_pad_push (dec->srcpad, outbuf);
switch (flow) {
case GST_FLOW_OK:
break;
case GST_FLOW_FLUSHING:
GST_LOG_OBJECT (dec, "pipeline is being flushed - pausing task");
goto pause;
case GST_FLOW_NOT_NEGOTIATED:
if (gst_pad_needs_reconfigure (dec->srcpad)) {
GST_DEBUG_OBJECT (dec, "trying to renegotiate");
break;
}
/* fallthrough to default */
default:
GST_ELEMENT_ERROR (dec, STREAM, FAILED, ("Internal data flow error."),
("streaming task paused, reason %s (%d)", gst_flow_get_name (flow),
flow));
}
return;
pause:
GST_INFO_OBJECT (dec, "pausing task");
/* NOT using stop_task here, since that would cause a deadlock.
* See the gst_pad_stop_task() documentation for details. */
gst_pad_pause_task (dec->srcpad);
return;
pause_unlock:
GST_NONSTREAM_AUDIO_DECODER_UNLOCK_MUTEX (dec);
goto pause;
}
static char const *
get_seek_type_name (GstSeekType seek_type)
{
switch (seek_type) {
case GST_SEEK_TYPE_NONE:
return "none";
case GST_SEEK_TYPE_SET:
return "set";
case GST_SEEK_TYPE_END:
return "end";
default:
return "<unknown>";
}
}
/**
* gst_nonstream_audio_decoder_handle_loop:
* @dec: a #GstNonstreamAudioDecoder
* @new_position New position the next loop starts with
*
* Reports that a loop has been completed and creates a new appropriate
* segment for the next loop.
*
* @new_position exists because a loop may not start at the beginning.
*
* This function is only useful for subclasses which can be in the
* GST_NONSTREAM_AUDIO_OUTPUT_MODE_LOOPING output mode, since in the
* GST_NONSTREAM_AUDIO_OUTPUT_MODE_STEADY output mode, this function
* does nothing. See #GstNonstreamAudioOutputMode for more details.
*
* The subclass calls this during playback when it loops. It produces
* a new segment with updated base time and internal time values, to allow
* for seamless looping. It does *not* check the number of elapsed loops;
* this is up the subclass.
*
* Note that if this function is called, then it must be done after the
* last samples of the loop have been decoded and pushed downstream.
*
* This function must be called with the decoder mutex lock held, since it
* is typically called from within @decode (which in turn are called with
* the lock already held).
*/
void
gst_nonstream_audio_decoder_handle_loop (GstNonstreamAudioDecoder * dec,
GstClockTime new_position)
{
if (dec->output_mode == GST_NONSTREAM_AUDIO_OUTPUT_MODE_STEADY) {
/* handle_loop makes no sense with open-ended decoders */
GST_WARNING_OBJECT (dec,
"ignoring handle_loop() call, since the decoder output mode is \"steady\"");
return;
}
GST_DEBUG_OBJECT (dec,
"handle_loop() invoked with new_position = %" GST_TIME_FORMAT,
GST_TIME_ARGS (new_position));
dec->discont = TRUE;
gst_nonstream_audio_decoder_output_new_segment (dec, new_position);
}
/**
* gst_nonstream_audio_decoder_set_output_format:
* @dec: a #GstNonstreamAudioDecoder
* @audio_info: Valid audio info structure containing the output format
*
* Sets the output caps by means of a GstAudioInfo structure.
*
* This must be called latest in the first @decode call, to ensure src caps are
* set before decoded samples are sent downstream. Typically, this is called
* from inside @load_from_buffer or @load_from_custom.
*
* This function must be called with the decoder mutex lock held, since it
* is typically called from within the aforementioned vfuncs (which in turn
* are called with the lock already held).
*
* Returns: TRUE if setting the output format succeeded, FALSE otherwise
*/
gboolean
gst_nonstream_audio_decoder_set_output_format (GstNonstreamAudioDecoder * dec,
GstAudioInfo const *audio_info)
{
GstCaps *caps;
GstCaps *templ_caps;
gboolean caps_ok;
gboolean res = TRUE;
g_return_val_if_fail (GST_IS_NONSTREAM_AUDIO_DECODER (dec), FALSE);
caps = gst_audio_info_to_caps (audio_info);
if (caps == NULL) {
GST_WARNING_OBJECT (dec, "Could not create caps out of audio info");
return FALSE;
}
templ_caps = gst_pad_get_pad_template_caps (dec->srcpad);
caps_ok = gst_caps_is_subset (caps, templ_caps);
if (caps_ok) {
dec->output_audio_info = *audio_info;
dec->output_format_changed = TRUE;
GST_INFO_OBJECT (dec, "setting output format to %" GST_PTR_FORMAT,
(gpointer) caps);
} else {
GST_WARNING_OBJECT (dec,
"requested output format %" GST_PTR_FORMAT " does not match template %"
GST_PTR_FORMAT, (gpointer) caps, (gpointer) templ_caps);
res = FALSE;
}
gst_caps_unref (caps);
gst_caps_unref (templ_caps);
return res;
}
/**
* gst_nonstream_audio_decoder_set_output_format_simple:
* @dec: a #GstNonstreamAudioDecoder
* @sample_rate: Output sample rate to use, in Hz
* @sample_format: Output sample format to use
* @num_channels: Number of output channels to use
*
* Convenience function; sets the output caps by means of common parameters.
*
* Internally, this fills a GstAudioInfo structure and calls
* gst_nonstream_audio_decoder_set_output_format().
*
* Returns: TRUE if setting the output format succeeded, FALSE otherwise
*/
gboolean
gst_nonstream_audio_decoder_set_output_format_simple (GstNonstreamAudioDecoder *
dec, guint sample_rate, GstAudioFormat sample_format, guint num_channels)
{
GstAudioInfo output_audio_info;
gst_audio_info_init (&output_audio_info);
gst_audio_info_set_format (&output_audio_info,
sample_format, sample_rate, num_channels, NULL);
return gst_nonstream_audio_decoder_set_output_format (dec,
&output_audio_info);
}
/**
* gst_nonstream_audio_decoder_get_downstream_info:
* @dec: a #GstNonstreamAudioDecoder
* @format: #GstAudioFormat value to fill with a sample format
* @sample_rate: Integer to fill with a sample rate
* @num_channels: Integer to fill with a channel count
*
* Gets sample format, sample rate, channel count from the allowed srcpad caps.
*
* This is useful for when the subclass wishes to adjust one or more output
* parameters to whatever downstream is supporting. For example, the output
* sample rate is often a freely adjustable value in module players.
*
* This function tries to find a value inside the srcpad peer's caps for
* @format, @sample_rate, @num_chnanels . Any of these can be NULL; they
* (and the corresponding downstream caps) are then skipped while retrieving
* information. Non-fixated caps are fixated first; the value closest to
* their present value is then chosen. For example, if the variables pointed
* to by the arguments are GST_AUDIO_FORMAT_16, 48000 Hz, and 2 channels,
* and the downstream caps are:
*
* "audio/x-raw, format={S16LE,S32LE}, rate=[1,32000], channels=[1,MAX]"
*
* Then @format and @channels stay the same, while @sample_rate is set to 32000 Hz.
* This way, the initial values the the variables pointed to by the arguments
* are set to can be used as default output values. Note that if no downstream
* caps can be retrieved, then this function does nothing, therefore it is
* necessary to ensure that @format, @sample_rate, and @channels have valid
* initial values.
*
* Decoder lock is not held by this function, so it can be called from within
* any of the class vfuncs.
*/
void
gst_nonstream_audio_decoder_get_downstream_info (GstNonstreamAudioDecoder * dec,
GstAudioFormat * format, gint * sample_rate, gint * num_channels)
{
GstCaps *allowed_srccaps;
guint structure_nr, num_structures;
gboolean ds_format_found = FALSE, ds_rate_found = FALSE, ds_channels_found =
FALSE;
g_return_if_fail (GST_IS_NONSTREAM_AUDIO_DECODER (dec));
allowed_srccaps = gst_pad_get_allowed_caps (dec->srcpad);
if (allowed_srccaps == NULL) {
GST_INFO_OBJECT (dec,
"no downstream caps available - not modifying arguments");
return;
}
num_structures = gst_caps_get_size (allowed_srccaps);
GST_DEBUG_OBJECT (dec, "%u structure(s) in downstream caps", num_structures);
for (structure_nr = 0; structure_nr < num_structures; ++structure_nr) {
GstStructure *structure;
ds_format_found = FALSE;
ds_rate_found = FALSE;
ds_channels_found = FALSE;
structure = gst_caps_get_structure (allowed_srccaps, structure_nr);
/* If all formats which need to be queried are present in the structure,
* check its contents */
if (((format == NULL) || gst_structure_has_field (structure, "format")) &&
((sample_rate == NULL) || gst_structure_has_field (structure, "rate"))
&& ((num_channels == NULL)
|| gst_structure_has_field (structure, "channels"))) {
gint fixated_sample_rate;
gint fixated_num_channels;
GstAudioFormat fixated_format = 0;
GstStructure *fixated_str;
gboolean passed = TRUE;
/* Make a copy of the structure, since we need to modify
* (fixate) values inside */
fixated_str = gst_structure_copy (structure);
/* Try to fixate and retrieve the sample format */
if (passed && (format != NULL)) {
passed = FALSE;
if ((gst_structure_get_field_type (fixated_str,
"format") == G_TYPE_STRING)
|| gst_structure_fixate_field_string (fixated_str, "format",
gst_audio_format_to_string (*format))) {
gchar const *fmt_str =
gst_structure_get_string (fixated_str, "format");
if (fmt_str
&& ((fixated_format =
gst_audio_format_from_string (fmt_str)) !=
GST_AUDIO_FORMAT_UNKNOWN)) {
GST_DEBUG_OBJECT (dec, "found fixated format: %s", fmt_str);
ds_format_found = TRUE;
passed = TRUE;
}
}
}
/* Try to fixate and retrieve the sample rate */
if (passed && (sample_rate != NULL)) {
passed = FALSE;
if ((gst_structure_get_field_type (fixated_str, "rate") == G_TYPE_INT)
|| gst_structure_fixate_field_nearest_int (fixated_str, "rate",
*sample_rate)) {
if (gst_structure_get_int (fixated_str, "rate", &fixated_sample_rate)) {
GST_DEBUG_OBJECT (dec, "found fixated sample rate: %d",
fixated_sample_rate);
ds_rate_found = TRUE;
passed = TRUE;
}
}
}
/* Try to fixate and retrieve the channel count */
if (passed && (num_channels != NULL)) {
passed = FALSE;
if ((gst_structure_get_field_type (fixated_str,
"channels") == G_TYPE_INT)
|| gst_structure_fixate_field_nearest_int (fixated_str, "channels",
*num_channels)) {
if (gst_structure_get_int (fixated_str, "channels",
&fixated_num_channels)) {
GST_DEBUG_OBJECT (dec, "found fixated channel count: %d",
fixated_num_channels);
ds_channels_found = TRUE;
passed = TRUE;
}
}
}
gst_structure_free (fixated_str);
if (ds_format_found && ds_rate_found && ds_channels_found) {
*format = fixated_format;
*sample_rate = fixated_sample_rate;
*num_channels = fixated_num_channels;
break;
}
}
}
gst_caps_unref (allowed_srccaps);
if ((format != NULL) && !ds_format_found)
GST_INFO_OBJECT (dec,
"downstream did not specify format - using default (%s)",
gst_audio_format_to_string (*format));
if ((sample_rate != NULL) && !ds_rate_found)
GST_INFO_OBJECT (dec,
"downstream did not specify sample rate - using default (%d Hz)",
*sample_rate);
if ((num_channels != NULL) && !ds_channels_found)
GST_INFO_OBJECT (dec,
"downstream did not specify number of channels - using default (%d channels)",
*num_channels);
}
/**
* gst_nonstream_audio_decoder_allocate_output_buffer:
* @dec: Decoder instance
* @size: Size of the output buffer, in bytes
*
* Allocates an output buffer with the internally configured buffer pool.
*
* This function may only be called from within @load_from_buffer,
* @load_from_custom, and @decode.
*
* Returns: Newly allocated output buffer, or NULL if allocation failed
*/
GstBuffer *
gst_nonstream_audio_decoder_allocate_output_buffer (GstNonstreamAudioDecoder *
dec, gsize size)
{
if (G_UNLIKELY (dec->output_format_changed ||
(GST_AUDIO_INFO_IS_VALID (&(dec->output_audio_info))
&& gst_pad_check_reconfigure (dec->srcpad))
)) {
/* renegotiate if necessary, before allocating,
* to make sure the right allocator and the right allocation
* params are used */
if (!gst_nonstream_audio_decoder_negotiate (dec)) {
GST_ERROR_OBJECT (dec,
"could not allocate output buffer because negotiation failed");
return NULL;
}
}
return gst_buffer_new_allocate (dec->allocator, size,
&(dec->allocation_params));
}
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