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024f61e75a
This will lead to deadlocks
1947 lines
59 KiB
C
1947 lines
59 KiB
C
/* GStreamer
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* Copyright (C) 2009 Igalia S.L.
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* Author: Iago Toral Quiroga <itoral@igalia.com>
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* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
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* Copyright (C) 2011 Nokia Corporation. All rights reserved.
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* Contact: Stefan Kost <stefan.kost@nokia.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:gstbaseaudiodecoder
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* @short_description: Base class for audio decoders
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* @see_also: #GstBaseTransform
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*
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* This base class is for audio decoders turning encoded data into
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* raw audio samples.
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*
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* GstBaseAudioDecoder and subclass should cooperate as follows.
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* <orderedlist>
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* <listitem>
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* <itemizedlist><title>Configuration</title>
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* <listitem><para>
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* Initially, GstBaseAudioDecoder calls @start when the decoder element
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* is activated, which allows subclass to perform any global setup.
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* Base class context parameters can already be set according to subclass
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* capabilities (or possibly upon receive more information in subsequent
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* @set_format).
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* </para></listitem>
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* <listitem><para>
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* GstBaseAudioDecoder calls @set_format to inform subclass of the format
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* of input audio data that it is about to receive.
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* While unlikely, it might be called more than once, if changing input
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* parameters require reconfiguration.
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* </para></listitem>
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* <listitem><para>
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* GstBaseAudioDecoder calls @stop at end of all processing.
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* </para></listitem>
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* </itemizedlist>
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* </listitem>
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* As of configuration stage, and throughout processing, GstBaseAudioDecoder
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* provides a GstBaseAudioDecoderContext that provides required context,
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* e.g. describing the format of output audio data
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* (valid when output caps have been caps) or current parsing state.
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* Conversely, subclass can and should configure context to inform
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* base class of its expectation w.r.t. buffer handling.
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* <listitem>
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* <itemizedlist>
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* <title>Data processing</title>
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* <listitem><para>
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* Base class gathers input data, and optionally allows subclass
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* to parse this into subsequently manageable (as defined by subclass)
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* chunks. Such chunks are subsequently referred to as 'frames',
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* though they may or may not correspond to 1 (or more) audio format frame.
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* </para></listitem>
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* <listitem><para>
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* Input frame is provided to subclass' @handle_frame.
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* </para></listitem>
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* <listitem><para>
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* If codec processing results in decoded data, subclass should call
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* @gst_base_audio_decoder_finish_frame to have decoded data pushed
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* downstream.
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* </para></listitem>
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* <listitem><para>
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* Just prior to actually pushing a buffer downstream,
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* it is passed to @pre_push. Subclass should either use this callback
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* to arrange for additional downstream pushing or otherwise ensure such
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* custom pushing occurs after at least a method call has finished since
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* setting src pad caps.
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* </para></listitem>
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* <listitem><para>
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* During the parsing process GstBaseAudioDecoderClass will handle both
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* srcpad and sinkpad events. Sink events will be passed to subclass
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* if @event callback has been provided.
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* </para></listitem>
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* </itemizedlist>
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* </listitem>
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* <listitem>
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* <itemizedlist><title>Shutdown phase</title>
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* <listitem><para>
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* GstBaseAudioDecoder class calls @stop to inform the subclass that data
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* parsing will be stopped.
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* </para></listitem>
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* </itemizedlist>
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* </listitem>
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* </orderedlist>
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*
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* Subclass is responsible for providing pad template caps for
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* source and sink pads. The pads need to be named "sink" and "src". It also
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* needs to set the fixed caps on srcpad, when the format is ensured. This
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* is typically when base class calls subclass' @set_format function, though
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* it might be delayed until calling @gst_base_audio_decoder_finish_frame.
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*
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* In summary, above process should have subclass concentrating on
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* codec data processing while leaving other matters to base class,
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* such as most notably timestamp handling. While it may exert more control
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* in this area (see e.g. @pre_push), it is very much not recommended.
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*
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* In particular, base class will try to arrange for perfect output timestamps
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* as much as possible while tracking upstream timestamps.
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* To this end, if deviation between the next ideal expected perfect timestamp
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* and upstream exceeds #GstBaseAudioDecoder:tolerance, then resync to upstream
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* occurs (which would happen always if the tolerance mechanism is disabled).
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*
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* In non-live pipelines, baseclass can also (configurably) arrange for
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* output buffer aggregation which may help to redue large(r) numbers of
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* small(er) buffers being pushed and processed downstream.
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*
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* On the other hand, it should be noted that baseclass only provides limited
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* seeking support (upon explicit subclass request), as full-fledged support
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* should rather be left to upstream demuxer, parser or alike. This simple
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* approach caters for seeking and duration reporting using estimated input
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* bitrates.
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*
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* Things that subclass need to take care of:
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* <itemizedlist>
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* <listitem><para>Provide pad templates</para></listitem>
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* <listitem><para>
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* Set source pad caps when appropriate
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* </para></listitem>
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* <listitem><para>
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* Set user-configurable properties to sane defaults for format and
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* implementing codec at hand, and convey some subclass capabilities and
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* expectations in context.
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* </para></listitem>
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* <listitem><para>
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* Accept data in @handle_frame and provide encoded results to
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* @gst_base_audio_decoder_finish_frame. If it is prepared to perform
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* PLC, it should also accept NULL data in @handle_frame and provide for
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* data for indicated duration.
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* </para></listitem>
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* </itemizedlist>
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstbaseaudiodecoder.h"
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#include <gst/audio/audio.h>
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#include <gst/base/gstadapter.h>
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#include <gst/pbutils/descriptions.h>
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#include <string.h>
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GST_DEBUG_CATEGORY (baseaudiodecoder_debug);
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#define GST_CAT_DEFAULT baseaudiodecoder_debug
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#define GST_BASE_AUDIO_DECODER_GET_PRIVATE(obj) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_DECODER, \
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GstBaseAudioDecoderPrivate))
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enum
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{
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LAST_SIGNAL
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};
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enum
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{
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PROP_0,
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PROP_LATENCY,
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PROP_TOLERANCE,
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PROP_PLC
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};
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#define DEFAULT_LATENCY 0
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#define DEFAULT_TOLERANCE 0
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#define DEFAULT_PLC FALSE
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struct _GstBaseAudioDecoderPrivate
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{
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/* activation status */
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gboolean active;
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/* input base/first ts as basis for output ts */
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GstClockTime base_ts;
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/* input samples processed and sent downstream so far (w.r.t. base_ts) */
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guint64 samples;
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/* collected input data */
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GstAdapter *adapter;
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/* tracking input ts for changes */
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GstClockTime prev_ts;
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/* frames obtained from input */
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GQueue frames;
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/* collected output data */
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GstAdapter *adapter_out;
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/* ts and duration for output data collected above */
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GstClockTime out_ts, out_dur;
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/* mark outgoing discont */
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gboolean discont;
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/* subclass gave all it could already */
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gboolean drained;
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/* subclass currently being forcibly drained */
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gboolean force;
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/* input bps estimatation */
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/* global in bytes seen */
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guint64 bytes_in;
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/* global samples sent out */
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guint64 samples_out;
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/* bytes flushed during parsing */
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guint sync_flush;
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/* error count */
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gint error_count;
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/* codec id tag */
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GstTagList *taglist;
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/* whether circumstances allow output aggregation */
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gint agg;
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/* reverse playback queues */
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/* collect input */
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GList *gather;
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/* to-be-decoded */
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GList *decode;
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/* reversed output */
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GList *queued;
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/* context storage */
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GstBaseAudioDecoderContext ctx;
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/* pending serialized sink events, will be sent from finish_frame() */
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GList *pending_events;
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};
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static void gst_base_audio_decoder_finalize (GObject * object);
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static void gst_base_audio_decoder_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_base_audio_decoder_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static void gst_base_audio_decoder_clear_queues (GstBaseAudioDecoder * dec);
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static GstFlowReturn gst_base_audio_decoder_chain_reverse (GstBaseAudioDecoder *
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dec, GstBuffer * buf);
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static GstStateChangeReturn gst_base_audio_decoder_change_state (GstElement *
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element, GstStateChange transition);
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static gboolean gst_base_audio_decoder_sink_event (GstPad * pad,
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GstEvent * event);
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static gboolean gst_base_audio_decoder_src_event (GstPad * pad,
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GstEvent * event);
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static gboolean gst_base_audio_decoder_sink_setcaps (GstPad * pad,
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GstCaps * caps);
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static gboolean gst_base_audio_decoder_src_setcaps (GstPad * pad,
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GstCaps * caps);
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static GstFlowReturn gst_base_audio_decoder_chain (GstPad * pad,
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GstBuffer * buf);
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static gboolean gst_base_audio_decoder_src_query (GstPad * pad,
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GstQuery * query);
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static gboolean gst_base_audio_decoder_sink_query (GstPad * pad,
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GstQuery * query);
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static const GstQueryType *gst_base_audio_decoder_get_query_types (GstPad *
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pad);
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static void gst_base_audio_decoder_reset (GstBaseAudioDecoder * dec,
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gboolean full);
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GST_BOILERPLATE (GstBaseAudioDecoder, gst_base_audio_decoder, GstElement,
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GST_TYPE_ELEMENT);
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static void
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gst_base_audio_decoder_base_init (gpointer g_class)
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{
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}
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static void
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gst_base_audio_decoder_class_init (GstBaseAudioDecoderClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *element_class;
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gobject_class = G_OBJECT_CLASS (klass);
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element_class = GST_ELEMENT_CLASS (klass);
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g_type_class_add_private (klass, sizeof (GstBaseAudioDecoderPrivate));
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GST_DEBUG_CATEGORY_INIT (baseaudiodecoder_debug, "baseaudiodecoder", 0,
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"baseaudiodecoder element");
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gobject_class->set_property = gst_base_audio_decoder_set_property;
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gobject_class->get_property = gst_base_audio_decoder_get_property;
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gobject_class->finalize = gst_base_audio_decoder_finalize;
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element_class->change_state = gst_base_audio_decoder_change_state;
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/* Properties */
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g_object_class_install_property (gobject_class, PROP_LATENCY,
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g_param_spec_int64 ("latency", "Latency",
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"Aggregate output data to a minimum of latency time (ns)",
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0, G_MAXINT64, DEFAULT_LATENCY,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_TOLERANCE,
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g_param_spec_int64 ("tolerance", "Tolerance",
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"Perfect ts while timestamp jitter/imperfection within tolerance (ns)",
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0, G_MAXINT64, DEFAULT_TOLERANCE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_PLC,
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g_param_spec_boolean ("plc", "Packet Loss Concealment",
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"Perform packet loss concealment (if supported)",
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DEFAULT_PLC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_base_audio_decoder_init (GstBaseAudioDecoder * dec,
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GstBaseAudioDecoderClass * klass)
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{
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GstPadTemplate *pad_template;
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GST_DEBUG_OBJECT (dec, "gst_base_audio_decoder_init");
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dec->priv = GST_BASE_AUDIO_DECODER_GET_PRIVATE (dec);
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/* Setup sink pad */
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pad_template =
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gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink");
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g_return_if_fail (pad_template != NULL);
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dec->sinkpad = gst_pad_new_from_template (pad_template, "sink");
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gst_pad_set_event_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_base_audio_decoder_sink_event));
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gst_pad_set_setcaps_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_base_audio_decoder_sink_setcaps));
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gst_pad_set_chain_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_base_audio_decoder_chain));
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gst_pad_set_query_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (gst_base_audio_decoder_sink_query));
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gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
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GST_DEBUG_OBJECT (dec, "sinkpad created");
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/* Setup source pad */
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pad_template =
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gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "src");
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g_return_if_fail (pad_template != NULL);
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dec->srcpad = gst_pad_new_from_template (pad_template, "src");
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gst_pad_set_setcaps_function (dec->srcpad,
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GST_DEBUG_FUNCPTR (gst_base_audio_decoder_src_setcaps));
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gst_pad_set_event_function (dec->srcpad,
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GST_DEBUG_FUNCPTR (gst_base_audio_decoder_src_event));
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gst_pad_set_query_function (dec->srcpad,
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GST_DEBUG_FUNCPTR (gst_base_audio_decoder_src_query));
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gst_pad_set_query_type_function (dec->srcpad,
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GST_DEBUG_FUNCPTR (gst_base_audio_decoder_get_query_types));
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gst_pad_use_fixed_caps (dec->srcpad);
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gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
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GST_DEBUG_OBJECT (dec, "srcpad created");
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dec->priv->adapter = gst_adapter_new ();
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dec->priv->adapter_out = gst_adapter_new ();
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g_queue_init (&dec->priv->frames);
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dec->ctx = &dec->priv->ctx;
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g_static_rec_mutex_init (&dec->stream_lock);
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/* property default */
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dec->latency = DEFAULT_LATENCY;
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dec->tolerance = DEFAULT_TOLERANCE;
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/* init state */
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gst_base_audio_decoder_reset (dec, TRUE);
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GST_DEBUG_OBJECT (dec, "init ok");
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}
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static void
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gst_base_audio_decoder_reset (GstBaseAudioDecoder * dec, gboolean full)
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{
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GST_DEBUG_OBJECT (dec, "gst_base_audio_decoder_reset");
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GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
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if (full) {
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dec->priv->active = FALSE;
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dec->priv->bytes_in = 0;
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dec->priv->samples_out = 0;
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dec->priv->agg = -1;
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dec->priv->error_count = 0;
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gst_base_audio_decoder_clear_queues (dec);
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g_free (dec->ctx->state.channel_pos);
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memset (dec->ctx, 0, sizeof (dec->ctx));
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if (dec->priv->taglist) {
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gst_tag_list_free (dec->priv->taglist);
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dec->priv->taglist = NULL;
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}
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gst_segment_init (&dec->segment, GST_FORMAT_TIME);
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g_list_foreach (dec->priv->pending_events, (GFunc) gst_event_unref, NULL);
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g_list_free (dec->priv->pending_events);
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dec->priv->pending_events = NULL;
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}
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g_queue_foreach (&dec->priv->frames, (GFunc) gst_buffer_unref, NULL);
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g_queue_clear (&dec->priv->frames);
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gst_adapter_clear (dec->priv->adapter);
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gst_adapter_clear (dec->priv->adapter_out);
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dec->priv->out_ts = GST_CLOCK_TIME_NONE;
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dec->priv->out_dur = 0;
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dec->priv->prev_ts = GST_CLOCK_TIME_NONE;
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dec->priv->drained = TRUE;
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dec->priv->base_ts = GST_CLOCK_TIME_NONE;
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dec->priv->samples = 0;
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dec->priv->discont = TRUE;
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dec->priv->sync_flush = FALSE;
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GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
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}
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static void
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gst_base_audio_decoder_finalize (GObject * object)
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{
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GstBaseAudioDecoder *dec;
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g_return_if_fail (GST_IS_BASE_AUDIO_DECODER (object));
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dec = GST_BASE_AUDIO_DECODER (object);
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if (dec->priv->adapter) {
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g_object_unref (dec->priv->adapter);
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}
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if (dec->priv->adapter_out) {
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g_object_unref (dec->priv->adapter_out);
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}
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g_static_rec_mutex_free (&dec->stream_lock);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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/* automagically perform sanity checking of src caps;
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* also extracts output data format */
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static gboolean
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gst_base_audio_decoder_src_setcaps (GstPad * pad, GstCaps * caps)
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{
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GstBaseAudioDecoder *dec;
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GstAudioState *state;
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gboolean res = TRUE, changed;
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dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
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state = &dec->ctx->state;
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GST_DEBUG_OBJECT (dec, "setting src caps %" GST_PTR_FORMAT, caps);
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GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
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/* parse caps here to check subclass;
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|
* also makes us aware of output format */
|
|
if (!gst_caps_is_fixed (caps))
|
|
goto refuse_caps;
|
|
|
|
/* adjust ts tracking to new sample rate */
|
|
if (GST_CLOCK_TIME_IS_VALID (dec->priv->base_ts) && state->rate) {
|
|
dec->priv->base_ts +=
|
|
GST_FRAMES_TO_CLOCK_TIME (dec->priv->samples, state->rate);
|
|
dec->priv->samples = 0;
|
|
}
|
|
|
|
if (!gst_base_audio_parse_caps (caps, state, &changed))
|
|
goto refuse_caps;
|
|
|
|
done:
|
|
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
|
|
|
|
gst_object_unref (dec);
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
refuse_caps:
|
|
{
|
|
GST_WARNING_OBJECT (dec, "rejected caps %" GST_PTR_FORMAT, caps);
|
|
res = FALSE;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
|
|
{
|
|
GstBaseAudioDecoder *dec;
|
|
GstBaseAudioDecoderClass *klass;
|
|
gboolean res = TRUE;
|
|
|
|
dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
|
|
klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
|
|
|
|
GST_DEBUG_OBJECT (dec, "caps: %" GST_PTR_FORMAT, caps);
|
|
|
|
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
|
|
/* NOTE pbutils only needed here */
|
|
/* TODO maybe (only) upstream demuxer/parser etc should handle this ? */
|
|
if (dec->priv->taglist)
|
|
gst_tag_list_free (dec->priv->taglist);
|
|
dec->priv->taglist = gst_tag_list_new ();
|
|
gst_pb_utils_add_codec_description_to_tag_list (dec->priv->taglist,
|
|
GST_TAG_AUDIO_CODEC, caps);
|
|
|
|
if (klass->set_format)
|
|
res = klass->set_format (dec, caps);
|
|
|
|
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
|
|
|
|
g_object_unref (dec);
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_base_audio_decoder_setup (GstBaseAudioDecoder * dec)
|
|
{
|
|
GstQuery *query;
|
|
gboolean res;
|
|
|
|
/* check if in live pipeline, then latency messing is no-no */
|
|
query = gst_query_new_latency ();
|
|
res = gst_pad_peer_query (dec->sinkpad, query);
|
|
if (res) {
|
|
gst_query_parse_latency (query, &res, NULL, NULL);
|
|
res = !res;
|
|
}
|
|
gst_query_unref (query);
|
|
|
|
/* normalize to bool */
|
|
dec->priv->agg = ! !res;
|
|
}
|
|
|
|
/* mini aggregator combining output buffers into fewer larger ones,
|
|
* if so allowed/configured */
|
|
static GstFlowReturn
|
|
gst_base_audio_decoder_output (GstBaseAudioDecoder * dec, GstBuffer * buf)
|
|
{
|
|
GstBaseAudioDecoderClass *klass;
|
|
GstBaseAudioDecoderPrivate *priv;
|
|
GstBaseAudioDecoderContext *ctx;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstBuffer *inbuf = NULL;
|
|
|
|
klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
|
|
priv = dec->priv;
|
|
ctx = dec->ctx;
|
|
|
|
if (G_UNLIKELY (priv->agg < 0))
|
|
gst_base_audio_decoder_setup (dec);
|
|
|
|
if (G_LIKELY (buf)) {
|
|
g_return_val_if_fail (ctx->state.bpf != 0, GST_FLOW_ERROR);
|
|
|
|
GST_LOG_OBJECT (dec, "output buffer of size %d with ts %" GST_TIME_FORMAT
|
|
", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
|
|
|
|
/* clip buffer */
|
|
buf = gst_audio_buffer_clip (buf, &dec->segment, ctx->state.rate,
|
|
ctx->state.bpf);
|
|
if (G_UNLIKELY (!buf)) {
|
|
GST_DEBUG_OBJECT (dec, "no data after clipping to segment");
|
|
} else {
|
|
GST_LOG_OBJECT (dec,
|
|
"buffer after segment clipping has size %d with ts %" GST_TIME_FORMAT
|
|
", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
|
|
}
|
|
} else {
|
|
GST_DEBUG_OBJECT (dec, "no output buffer");
|
|
}
|
|
|
|
again:
|
|
inbuf = NULL;
|
|
if (priv->agg && dec->latency > 0) {
|
|
gint av;
|
|
gboolean assemble = FALSE;
|
|
const GstClockTimeDiff tol = 10 * GST_MSECOND;
|
|
GstClockTimeDiff diff = -100 * GST_MSECOND;
|
|
|
|
av = gst_adapter_available (priv->adapter_out);
|
|
if (G_UNLIKELY (!buf)) {
|
|
/* forcibly send current */
|
|
assemble = TRUE;
|
|
GST_LOG_OBJECT (dec, "forcing fragment flush");
|
|
} else if (av && (!GST_BUFFER_TIMESTAMP_IS_VALID (buf) ||
|
|
!GST_CLOCK_TIME_IS_VALID (priv->out_ts) ||
|
|
((diff = GST_CLOCK_DIFF (GST_BUFFER_TIMESTAMP (buf),
|
|
priv->out_ts + priv->out_dur)) > tol) || diff < -tol)) {
|
|
assemble = TRUE;
|
|
GST_LOG_OBJECT (dec, "buffer %d ms apart from current fragment",
|
|
(gint) (diff / GST_MSECOND));
|
|
} else {
|
|
/* add or start collecting */
|
|
if (!av) {
|
|
GST_LOG_OBJECT (dec, "starting new fragment");
|
|
priv->out_ts = GST_BUFFER_TIMESTAMP (buf);
|
|
} else {
|
|
GST_LOG_OBJECT (dec, "adding to fragment");
|
|
}
|
|
gst_adapter_push (priv->adapter_out, buf);
|
|
priv->out_dur += GST_BUFFER_DURATION (buf);
|
|
av += GST_BUFFER_SIZE (buf);
|
|
buf = NULL;
|
|
}
|
|
if (priv->out_dur > dec->latency)
|
|
assemble = TRUE;
|
|
if (av && assemble) {
|
|
GST_LOG_OBJECT (dec, "assembling fragment");
|
|
inbuf = buf;
|
|
buf = gst_adapter_take_buffer (priv->adapter_out, av);
|
|
GST_BUFFER_TIMESTAMP (buf) = priv->out_ts;
|
|
GST_BUFFER_DURATION (buf) = priv->out_dur;
|
|
priv->out_ts = GST_CLOCK_TIME_NONE;
|
|
priv->out_dur = 0;
|
|
}
|
|
}
|
|
|
|
if (G_LIKELY (buf)) {
|
|
|
|
/* decorate */
|
|
gst_buffer_set_caps (buf, GST_PAD_CAPS (dec->srcpad));
|
|
|
|
if (G_UNLIKELY (priv->discont)) {
|
|
GST_LOG_OBJECT (dec, "marking discont");
|
|
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
|
|
priv->discont = FALSE;
|
|
}
|
|
|
|
if (G_LIKELY (GST_BUFFER_TIMESTAMP_IS_VALID (buf))) {
|
|
/* duration should always be valid for raw audio */
|
|
g_assert (GST_BUFFER_DURATION_IS_VALID (buf));
|
|
dec->segment.last_stop =
|
|
GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
|
|
}
|
|
|
|
if (klass->pre_push) {
|
|
/* last chance for subclass to do some dirty stuff */
|
|
ret = klass->pre_push (dec, &buf);
|
|
if (ret != GST_FLOW_OK || !buf) {
|
|
GST_DEBUG_OBJECT (dec, "subclass returned %s, buf %p",
|
|
gst_flow_get_name (ret), buf);
|
|
if (buf)
|
|
gst_buffer_unref (buf);
|
|
goto exit;
|
|
}
|
|
}
|
|
|
|
GST_LOG_OBJECT (dec, "pushing buffer of size %d with ts %" GST_TIME_FORMAT
|
|
", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
|
|
|
|
if (dec->segment.rate > 0.0) {
|
|
ret = gst_pad_push (dec->srcpad, buf);
|
|
GST_LOG_OBJECT (dec, "buffer pushed: %s", gst_flow_get_name (ret));
|
|
} else {
|
|
ret = GST_FLOW_OK;
|
|
priv->queued = g_list_prepend (priv->queued, buf);
|
|
GST_LOG_OBJECT (dec, "buffer queued");
|
|
}
|
|
|
|
exit:
|
|
if (inbuf) {
|
|
buf = inbuf;
|
|
goto again;
|
|
}
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
GstFlowReturn
|
|
gst_base_audio_decoder_finish_frame (GstBaseAudioDecoder * dec, GstBuffer * buf,
|
|
gint frames)
|
|
{
|
|
GstBaseAudioDecoderPrivate *priv;
|
|
GstBaseAudioDecoderContext *ctx;
|
|
gint samples = 0;
|
|
GstClockTime ts, next_ts;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
|
|
/* subclass should know what it is producing by now */
|
|
g_return_val_if_fail (buf == NULL || GST_PAD_CAPS (dec->srcpad) != NULL,
|
|
GST_FLOW_ERROR);
|
|
/* subclass should not hand us no data */
|
|
g_return_val_if_fail (buf == NULL || GST_BUFFER_SIZE (buf) > 0,
|
|
GST_FLOW_ERROR);
|
|
/* no dummy calls please */
|
|
g_return_val_if_fail (frames != 0, GST_FLOW_ERROR);
|
|
|
|
priv = dec->priv;
|
|
ctx = dec->ctx;
|
|
|
|
GST_LOG_OBJECT (dec, "accepting %d bytes == %d samples for %d frames",
|
|
buf ? GST_BUFFER_SIZE (buf) : -1,
|
|
buf ? GST_BUFFER_SIZE (buf) / ctx->state.bpf : -1, frames);
|
|
|
|
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
|
|
|
|
if (priv->pending_events) {
|
|
GList *pending_events, *l;
|
|
|
|
pending_events = priv->pending_events;
|
|
priv->pending_events = NULL;
|
|
|
|
GST_DEBUG_OBJECT (dec, "Pushing pending events");
|
|
for (l = priv->pending_events; l; l = l->next)
|
|
gst_pad_push_event (dec->srcpad, l->data);
|
|
g_list_free (pending_events);
|
|
}
|
|
|
|
/* output shoud be whole number of sample frames */
|
|
if (G_LIKELY (buf && ctx->state.bpf)) {
|
|
if (GST_BUFFER_SIZE (buf) % ctx->state.bpf)
|
|
goto wrong_buffer;
|
|
/* per channel least */
|
|
samples = GST_BUFFER_SIZE (buf) / ctx->state.bpf;
|
|
}
|
|
|
|
/* frame and ts book-keeping */
|
|
if (G_UNLIKELY (frames < 0)) {
|
|
if (G_UNLIKELY (-frames - 1 > priv->frames.length))
|
|
goto overflow;
|
|
frames = priv->frames.length + frames + 1;
|
|
} else if (G_UNLIKELY (frames > priv->frames.length)) {
|
|
if (G_LIKELY (!priv->force)) {
|
|
/* no way we can let this pass */
|
|
g_assert_not_reached ();
|
|
/* really no way */
|
|
goto overflow;
|
|
}
|
|
}
|
|
|
|
if (G_LIKELY (priv->frames.length))
|
|
ts = GST_BUFFER_TIMESTAMP (priv->frames.head->data);
|
|
else
|
|
ts = GST_CLOCK_TIME_NONE;
|
|
|
|
GST_DEBUG_OBJECT (dec, "leading frame ts %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (ts));
|
|
|
|
while (priv->frames.length && frames) {
|
|
gst_buffer_unref (g_queue_pop_head (&priv->frames));
|
|
dec->ctx->delay = dec->priv->frames.length;
|
|
frames--;
|
|
}
|
|
|
|
/* lock on */
|
|
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
|
|
priv->base_ts = ts;
|
|
GST_DEBUG_OBJECT (dec, "base_ts now %" GST_TIME_FORMAT, GST_TIME_ARGS (ts));
|
|
}
|
|
|
|
if (G_UNLIKELY (!buf))
|
|
goto exit;
|
|
|
|
/* slightly convoluted approach caters for perfect ts if subclass desires */
|
|
if (GST_CLOCK_TIME_IS_VALID (ts)) {
|
|
if (dec->tolerance > 0) {
|
|
GstClockTimeDiff diff;
|
|
|
|
g_assert (GST_CLOCK_TIME_IS_VALID (priv->base_ts));
|
|
next_ts = priv->base_ts +
|
|
gst_util_uint64_scale (samples, GST_SECOND, ctx->state.rate);
|
|
GST_LOG_OBJECT (dec, "buffer is %d samples past base_ts %" GST_TIME_FORMAT
|
|
", expected ts %" GST_TIME_FORMAT, samples,
|
|
GST_TIME_ARGS (priv->base_ts), GST_TIME_ARGS (next_ts));
|
|
diff = GST_CLOCK_DIFF (next_ts, ts);
|
|
GST_LOG_OBJECT (dec, "ts diff %d ms", (gint) (diff / GST_MSECOND));
|
|
/* if within tolerance,
|
|
* discard buffer ts and carry on producing perfect stream,
|
|
* otherwise resync to ts */
|
|
if (G_UNLIKELY (diff < -dec->tolerance || diff > dec->tolerance)) {
|
|
GST_DEBUG_OBJECT (dec, "base_ts resync");
|
|
priv->base_ts = ts;
|
|
priv->samples = 0;
|
|
}
|
|
} else {
|
|
GST_DEBUG_OBJECT (dec, "base_ts resync");
|
|
priv->base_ts = ts;
|
|
priv->samples = 0;
|
|
}
|
|
}
|
|
|
|
/* delayed one-shot stuff until confirmed data */
|
|
if (priv->taglist) {
|
|
GST_DEBUG_OBJECT (dec, "codec tag %" GST_PTR_FORMAT, priv->taglist);
|
|
if (gst_tag_list_is_empty (priv->taglist)) {
|
|
gst_tag_list_free (priv->taglist);
|
|
} else {
|
|
gst_element_found_tags (GST_ELEMENT (dec), priv->taglist);
|
|
}
|
|
priv->taglist = NULL;
|
|
}
|
|
|
|
buf = gst_buffer_make_metadata_writable (buf);
|
|
if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
|
|
GST_BUFFER_TIMESTAMP (buf) =
|
|
priv->base_ts +
|
|
GST_FRAMES_TO_CLOCK_TIME (priv->samples, ctx->state.rate);
|
|
GST_BUFFER_DURATION (buf) = priv->base_ts +
|
|
GST_FRAMES_TO_CLOCK_TIME (priv->samples + samples, ctx->state.rate) -
|
|
GST_BUFFER_TIMESTAMP (buf);
|
|
} else {
|
|
GST_BUFFER_TIMESTAMP (buf) = GST_CLOCK_TIME_NONE;
|
|
GST_BUFFER_DURATION (buf) =
|
|
GST_FRAMES_TO_CLOCK_TIME (samples, ctx->state.rate);
|
|
}
|
|
priv->samples += samples;
|
|
priv->samples_out += samples;
|
|
|
|
/* we got data, so note things are looking up */
|
|
if (G_UNLIKELY (dec->priv->error_count))
|
|
dec->priv->error_count--;
|
|
|
|
exit:
|
|
ret = gst_base_audio_decoder_output (dec, buf);
|
|
|
|
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
wrong_buffer:
|
|
{
|
|
GST_ELEMENT_ERROR (dec, STREAM, ENCODE, (NULL),
|
|
("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buf),
|
|
ctx->state.bpf));
|
|
gst_buffer_unref (buf);
|
|
ret = GST_FLOW_ERROR;
|
|
goto exit;
|
|
}
|
|
overflow:
|
|
{
|
|
GST_ELEMENT_ERROR (dec, STREAM, ENCODE,
|
|
("received more decoded frames %d than provided %d", frames,
|
|
priv->frames.length), (NULL));
|
|
if (buf)
|
|
gst_buffer_unref (buf);
|
|
ret = GST_FLOW_ERROR;
|
|
goto exit;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_audio_decoder_handle_frame (GstBaseAudioDecoder * dec,
|
|
GstBaseAudioDecoderClass * klass, GstBuffer * buffer)
|
|
{
|
|
if (G_LIKELY (buffer)) {
|
|
/* keep around for admin */
|
|
GST_LOG_OBJECT (dec, "tracking frame size %d, ts %" GST_TIME_FORMAT,
|
|
GST_BUFFER_SIZE (buffer),
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
|
|
g_queue_push_tail (&dec->priv->frames, buffer);
|
|
dec->ctx->delay = dec->priv->frames.length;
|
|
dec->priv->bytes_in += GST_BUFFER_SIZE (buffer);
|
|
} else {
|
|
GST_LOG_OBJECT (dec, "providing subclass with NULL frame");
|
|
}
|
|
|
|
return klass->handle_frame (dec, buffer);
|
|
}
|
|
|
|
/* maybe subclass configurable instead, but this allows for a whole lot of
|
|
* raw samples, so at least quite some encoded ... */
|
|
#define GST_BASE_AUDIO_DECODER_MAX_SYNC 10 * 8 * 2 * 1024
|
|
|
|
static GstFlowReturn
|
|
gst_base_audio_decoder_push_buffers (GstBaseAudioDecoder * dec, gboolean force)
|
|
{
|
|
GstBaseAudioDecoderClass *klass;
|
|
GstBaseAudioDecoderPrivate *priv;
|
|
GstBaseAudioDecoderContext *ctx;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstBuffer *buffer;
|
|
gint av, flush;
|
|
|
|
klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
|
|
priv = dec->priv;
|
|
ctx = dec->ctx;
|
|
|
|
g_return_val_if_fail (klass->handle_frame != NULL, GST_FLOW_ERROR);
|
|
|
|
av = gst_adapter_available (priv->adapter);
|
|
GST_DEBUG_OBJECT (dec, "available: %d", av);
|
|
|
|
while (ret == GST_FLOW_OK) {
|
|
|
|
flush = 0;
|
|
ctx->eos = force;
|
|
|
|
if (G_LIKELY (av)) {
|
|
gint len;
|
|
GstClockTime ts;
|
|
|
|
/* parse if needed */
|
|
if (klass->parse) {
|
|
gint offset = 0;
|
|
|
|
/* limited (legacy) parsing; avoid whole of baseparse */
|
|
GST_DEBUG_OBJECT (dec, "parsing available: %d", av);
|
|
/* piggyback sync state on discont */
|
|
ctx->sync = !priv->discont;
|
|
ret = klass->parse (dec, priv->adapter, &offset, &len);
|
|
|
|
g_assert (offset <= av);
|
|
if (offset) {
|
|
/* jumped a bit */
|
|
GST_DEBUG_OBJECT (dec, "setting DISCONT");
|
|
gst_adapter_flush (priv->adapter, offset);
|
|
flush = offset;
|
|
/* avoid parsing indefinitely */
|
|
priv->sync_flush += offset;
|
|
if (priv->sync_flush > GST_BASE_AUDIO_DECODER_MAX_SYNC)
|
|
goto parse_failed;
|
|
}
|
|
|
|
if (ret == GST_FLOW_UNEXPECTED) {
|
|
GST_LOG_OBJECT (dec, "no frame yet");
|
|
ret = GST_FLOW_OK;
|
|
break;
|
|
} else if (ret == GST_FLOW_OK) {
|
|
GST_LOG_OBJECT (dec, "frame at offset %d of length %d", offset, len);
|
|
g_assert (offset + len <= av);
|
|
priv->sync_flush = 0;
|
|
} else {
|
|
break;
|
|
}
|
|
} else {
|
|
len = av;
|
|
}
|
|
/* track upstream ts, but do not get stuck if nothing new upstream */
|
|
ts = gst_adapter_prev_timestamp (priv->adapter, NULL);
|
|
if (ts == priv->prev_ts) {
|
|
GST_LOG_OBJECT (dec, "ts == prev_ts; discarding");
|
|
ts = GST_CLOCK_TIME_NONE;
|
|
} else {
|
|
priv->prev_ts = ts;
|
|
}
|
|
buffer = gst_adapter_take_buffer (priv->adapter, len);
|
|
buffer = gst_buffer_make_metadata_writable (buffer);
|
|
GST_BUFFER_TIMESTAMP (buffer) = ts;
|
|
flush += len;
|
|
} else {
|
|
if (!force)
|
|
break;
|
|
buffer = NULL;
|
|
}
|
|
|
|
ret = gst_base_audio_decoder_handle_frame (dec, klass, buffer);
|
|
|
|
/* do not keep pushing it ... */
|
|
if (G_UNLIKELY (!av)) {
|
|
priv->drained = TRUE;
|
|
break;
|
|
}
|
|
|
|
av -= flush;
|
|
g_assert (av >= 0);
|
|
}
|
|
|
|
GST_LOG_OBJECT (dec, "done pushing to subclass");
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
parse_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("failed to parse stream"));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_audio_decoder_drain (GstBaseAudioDecoder * dec)
|
|
{
|
|
GstFlowReturn ret;
|
|
|
|
if (dec->priv->drained)
|
|
return GST_FLOW_OK;
|
|
else {
|
|
/* dispatch reverse pending buffers */
|
|
/* chain eventually calls upon drain as well, but by that time
|
|
* gather list should be clear, so ok ... */
|
|
if (dec->segment.rate < 0.0 && dec->priv->gather)
|
|
gst_base_audio_decoder_chain_reverse (dec, NULL);
|
|
/* have subclass give all it can */
|
|
ret = gst_base_audio_decoder_push_buffers (dec, TRUE);
|
|
/* ensure all output sent */
|
|
ret = gst_base_audio_decoder_output (dec, NULL);
|
|
/* everything should be away now */
|
|
if (dec->priv->frames.length) {
|
|
/* not fatal/impossible though if subclass/codec eats stuff */
|
|
GST_WARNING_OBJECT (dec, "still %d frames left after draining",
|
|
dec->priv->frames.length);
|
|
g_queue_foreach (&dec->priv->frames, (GFunc) gst_buffer_unref, NULL);
|
|
g_queue_clear (&dec->priv->frames);
|
|
}
|
|
/* discard (unparsed) leftover */
|
|
gst_adapter_clear (dec->priv->adapter);
|
|
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
/* hard == FLUSH, otherwise discont */
|
|
static GstFlowReturn
|
|
gst_base_audio_decoder_flush (GstBaseAudioDecoder * dec, gboolean hard)
|
|
{
|
|
GstBaseAudioDecoderClass *klass;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
|
|
klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
|
|
|
|
GST_LOG_OBJECT (dec, "flush hard %d", hard);
|
|
|
|
if (!hard) {
|
|
ret = gst_base_audio_decoder_drain (dec);
|
|
} else {
|
|
gst_base_audio_decoder_clear_queues (dec);
|
|
gst_segment_init (&dec->segment, GST_FORMAT_TIME);
|
|
dec->priv->error_count = 0;
|
|
}
|
|
/* only bother subclass with flushing if known it is already alive
|
|
* and kicking out stuff */
|
|
if (klass->flush && dec->priv->samples_out > 0)
|
|
klass->flush (dec, hard);
|
|
/* and get (re)set for the sequel */
|
|
gst_base_audio_decoder_reset (dec, FALSE);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_audio_decoder_chain_forward (GstBaseAudioDecoder * dec,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstFlowReturn ret;
|
|
|
|
/* grab buffer */
|
|
gst_adapter_push (dec->priv->adapter, buffer);
|
|
buffer = NULL;
|
|
/* new stuff, so we can push subclass again */
|
|
dec->priv->drained = FALSE;
|
|
|
|
/* hand to subclass */
|
|
ret = gst_base_audio_decoder_push_buffers (dec, FALSE);
|
|
|
|
GST_LOG_OBJECT (dec, "chain-done");
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_base_audio_decoder_clear_queues (GstBaseAudioDecoder * dec)
|
|
{
|
|
GstBaseAudioDecoderPrivate *priv = dec->priv;
|
|
|
|
g_list_foreach (priv->queued, (GFunc) gst_mini_object_unref, NULL);
|
|
g_list_free (priv->queued);
|
|
priv->queued = NULL;
|
|
g_list_foreach (priv->gather, (GFunc) gst_mini_object_unref, NULL);
|
|
g_list_free (priv->gather);
|
|
priv->gather = NULL;
|
|
g_list_foreach (priv->decode, (GFunc) gst_mini_object_unref, NULL);
|
|
g_list_free (priv->decode);
|
|
priv->decode = NULL;
|
|
}
|
|
|
|
/*
|
|
* Input:
|
|
* Buffer decoding order: 7 8 9 4 5 6 3 1 2 EOS
|
|
* Discont flag: D D D D
|
|
*
|
|
* - Each Discont marks a discont in the decoding order.
|
|
*
|
|
* for vorbis, each buffer is a keyframe when we have the previous
|
|
* buffer. This means that to decode buffer 7, we need buffer 6, which
|
|
* arrives out of order.
|
|
*
|
|
* we first gather buffers in the gather queue until we get a DISCONT. We
|
|
* prepend each incomming buffer so that they are in reversed order.
|
|
*
|
|
* gather queue: 9 8 7
|
|
* decode queue:
|
|
* output queue:
|
|
*
|
|
* When a DISCONT is received (buffer 4), we move the gather queue to the
|
|
* decode queue. This is simply done be taking the head of the gather queue
|
|
* and prepending it to the decode queue. This yields:
|
|
*
|
|
* gather queue:
|
|
* decode queue: 7 8 9
|
|
* output queue:
|
|
*
|
|
* Then we decode each buffer in the decode queue in order and put the output
|
|
* buffer in the output queue. The first buffer (7) will not produce any output
|
|
* because it needs the previous buffer (6) which did not arrive yet. This
|
|
* yields:
|
|
*
|
|
* gather queue:
|
|
* decode queue: 7 8 9
|
|
* output queue: 9 8
|
|
*
|
|
* Then we remove the consumed buffers from the decode queue. Buffer 7 is not
|
|
* completely consumed, we need to keep it around for when we receive buffer
|
|
* 6. This yields:
|
|
*
|
|
* gather queue:
|
|
* decode queue: 7
|
|
* output queue: 9 8
|
|
*
|
|
* Then we accumulate more buffers:
|
|
*
|
|
* gather queue: 6 5 4
|
|
* decode queue: 7
|
|
* output queue:
|
|
*
|
|
* prepending to the decode queue on DISCONT yields:
|
|
*
|
|
* gather queue:
|
|
* decode queue: 4 5 6 7
|
|
* output queue:
|
|
*
|
|
* after decoding and keeping buffer 4:
|
|
*
|
|
* gather queue:
|
|
* decode queue: 4
|
|
* output queue: 7 6 5
|
|
*
|
|
* Etc..
|
|
*/
|
|
static GstFlowReturn
|
|
gst_base_audio_decoder_flush_decode (GstBaseAudioDecoder * dec)
|
|
{
|
|
GstBaseAudioDecoderPrivate *priv = dec->priv;
|
|
GstFlowReturn res = GST_FLOW_OK;
|
|
GList *walk;
|
|
|
|
walk = priv->decode;
|
|
|
|
GST_DEBUG_OBJECT (dec, "flushing buffers to decoder");
|
|
|
|
/* clear buffer and decoder state */
|
|
gst_base_audio_decoder_flush (dec, FALSE);
|
|
|
|
while (walk) {
|
|
GList *next;
|
|
GstBuffer *buf = GST_BUFFER_CAST (walk->data);
|
|
|
|
GST_DEBUG_OBJECT (dec, "decoding buffer %p, ts %" GST_TIME_FORMAT,
|
|
buf, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
|
|
|
|
next = g_list_next (walk);
|
|
/* decode buffer, resulting data prepended to output queue */
|
|
gst_buffer_ref (buf);
|
|
res = gst_base_audio_decoder_chain_forward (dec, buf);
|
|
|
|
/* if we generated output, we can discard the buffer, else we
|
|
* keep it in the queue */
|
|
if (priv->queued) {
|
|
GST_DEBUG_OBJECT (dec, "decoded buffer to %p", priv->queued->data);
|
|
priv->decode = g_list_delete_link (priv->decode, walk);
|
|
gst_buffer_unref (buf);
|
|
} else {
|
|
GST_DEBUG_OBJECT (dec, "buffer did not decode, keeping");
|
|
}
|
|
walk = next;
|
|
}
|
|
|
|
/* drain any aggregation (or otherwise) leftover */
|
|
gst_base_audio_decoder_drain (dec);
|
|
|
|
/* now send queued data downstream */
|
|
while (priv->queued) {
|
|
GstBuffer *buf = GST_BUFFER_CAST (priv->queued->data);
|
|
|
|
if (G_LIKELY (res == GST_FLOW_OK)) {
|
|
GST_DEBUG_OBJECT (dec, "pushing buffer %p of size %u, "
|
|
"time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, buf,
|
|
GST_BUFFER_SIZE (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
|
|
/* should be already, but let's be sure */
|
|
buf = gst_buffer_make_metadata_writable (buf);
|
|
/* avoid stray DISCONT from forward processing,
|
|
* which have no meaning in reverse pushing */
|
|
GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
|
|
res = gst_pad_push (dec->srcpad, buf);
|
|
} else {
|
|
gst_buffer_unref (buf);
|
|
}
|
|
|
|
priv->queued = g_list_delete_link (priv->queued, priv->queued);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_audio_decoder_chain_reverse (GstBaseAudioDecoder * dec,
|
|
GstBuffer * buf)
|
|
{
|
|
GstBaseAudioDecoderPrivate *priv = dec->priv;
|
|
GstFlowReturn result = GST_FLOW_OK;
|
|
|
|
/* if we have a discont, move buffers to the decode list */
|
|
if (!buf || GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
|
|
GST_DEBUG_OBJECT (dec, "received discont");
|
|
while (priv->gather) {
|
|
GstBuffer *gbuf;
|
|
|
|
gbuf = GST_BUFFER_CAST (priv->gather->data);
|
|
/* remove from the gather list */
|
|
priv->gather = g_list_delete_link (priv->gather, priv->gather);
|
|
/* copy to decode queue */
|
|
priv->decode = g_list_prepend (priv->decode, gbuf);
|
|
}
|
|
/* decode stuff in the decode queue */
|
|
gst_base_audio_decoder_flush_decode (dec);
|
|
}
|
|
|
|
if (G_LIKELY (buf)) {
|
|
GST_DEBUG_OBJECT (dec, "gathering buffer %p of size %u, "
|
|
"time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, buf,
|
|
GST_BUFFER_SIZE (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
|
|
|
|
/* add buffer to gather queue */
|
|
priv->gather = g_list_prepend (priv->gather, buf);
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_audio_decoder_chain (GstPad * pad, GstBuffer * buffer)
|
|
{
|
|
GstBaseAudioDecoder *dec;
|
|
GstFlowReturn ret;
|
|
|
|
dec = GST_BASE_AUDIO_DECODER (GST_PAD_PARENT (pad));
|
|
|
|
GST_LOG_OBJECT (dec,
|
|
"received buffer of size %d with ts %" GST_TIME_FORMAT
|
|
", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
|
|
|
|
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
|
|
|
|
if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
|
|
gint64 samples, ts;
|
|
|
|
/* track present position */
|
|
ts = dec->priv->base_ts;
|
|
samples = dec->priv->samples;
|
|
|
|
GST_DEBUG_OBJECT (dec, "handling discont");
|
|
gst_base_audio_decoder_flush (dec, FALSE);
|
|
dec->priv->discont = TRUE;
|
|
|
|
/* buffer may claim DISCONT loudly, if it can't tell us where we are now,
|
|
* we'll stick to where we were ...
|
|
* Particularly useful/needed for upstream BYTE based */
|
|
if (dec->segment.rate > 0.0 && !GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
|
|
GST_DEBUG_OBJECT (dec, "... but restoring previous ts tracking");
|
|
dec->priv->base_ts = ts;
|
|
dec->priv->samples = samples;
|
|
}
|
|
}
|
|
|
|
if (dec->segment.rate > 0.0)
|
|
ret = gst_base_audio_decoder_chain_forward (dec, buffer);
|
|
else
|
|
ret = gst_base_audio_decoder_chain_reverse (dec, buffer);
|
|
|
|
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* perform upstream byte <-> time conversion (duration, seeking)
|
|
* if subclass allows and if enough data for moderately decent conversion */
|
|
static inline gboolean
|
|
gst_base_audio_decoder_do_byte (GstBaseAudioDecoder * dec)
|
|
{
|
|
return dec->ctx->do_byte_time && dec->ctx->state.bpf &&
|
|
dec->ctx->state.rate <= dec->priv->samples_out;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_decoder_sink_eventfunc (GstBaseAudioDecoder * dec,
|
|
GstEvent * event)
|
|
{
|
|
gboolean handled = FALSE;
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_NEWSEGMENT:
|
|
{
|
|
GstFormat format;
|
|
gdouble rate, arate;
|
|
gint64 start, stop, time;
|
|
gboolean update;
|
|
|
|
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
|
|
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
|
|
&start, &stop, &time);
|
|
|
|
if (format == GST_FORMAT_TIME) {
|
|
GST_DEBUG_OBJECT (dec, "received TIME NEW_SEGMENT %" GST_TIME_FORMAT
|
|
" -- %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT
|
|
", rate %g, applied_rate %g",
|
|
GST_TIME_ARGS (start), GST_TIME_ARGS (stop), GST_TIME_ARGS (time),
|
|
rate, arate);
|
|
} else {
|
|
GstFormat dformat = GST_FORMAT_TIME;
|
|
|
|
GST_DEBUG_OBJECT (dec, "received NEW_SEGMENT %" G_GINT64_FORMAT
|
|
" -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT
|
|
", rate %g, applied_rate %g", start, stop, time, rate, arate);
|
|
/* handle newsegment resulting from legacy simple seeking */
|
|
/* note that we need to convert this whether or not enough data
|
|
* to handle initial newsegment */
|
|
if (dec->ctx->do_byte_time &&
|
|
gst_pad_query_convert (dec->sinkpad, GST_FORMAT_BYTES, start,
|
|
&dformat, &start)) {
|
|
/* best attempt convert */
|
|
/* as these are only estimates, stop is kept open-ended to avoid
|
|
* premature cutting */
|
|
GST_DEBUG_OBJECT (dec, "converted to TIME start %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (start));
|
|
format = GST_FORMAT_TIME;
|
|
time = start;
|
|
stop = GST_CLOCK_TIME_NONE;
|
|
/* replace event */
|
|
gst_event_unref (event);
|
|
event = gst_event_new_new_segment_full (update, rate, arate,
|
|
GST_FORMAT_TIME, start, stop, time);
|
|
} else {
|
|
GST_DEBUG_OBJECT (dec, "unsupported format; ignoring");
|
|
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* finish current segment */
|
|
gst_base_audio_decoder_drain (dec);
|
|
|
|
if (update) {
|
|
/* time progressed without data, see if we can fill the gap with
|
|
* some concealment data */
|
|
GST_DEBUG_OBJECT (dec,
|
|
"segment update: plc %d, do_plc %d, last_stop %" GST_TIME_FORMAT,
|
|
dec->plc, dec->ctx->do_plc, GST_TIME_ARGS (dec->segment.last_stop));
|
|
if (dec->plc && dec->ctx->do_plc && dec->segment.rate > 0.0 &&
|
|
dec->segment.last_stop < start) {
|
|
GstBaseAudioDecoderClass *klass;
|
|
GstBuffer *buf;
|
|
|
|
klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
|
|
/* hand subclass empty frame with duration that needs covering */
|
|
buf = gst_buffer_new ();
|
|
GST_BUFFER_DURATION (buf) = start - dec->segment.last_stop;
|
|
/* best effort, not much error handling */
|
|
gst_base_audio_decoder_handle_frame (dec, klass, buf);
|
|
}
|
|
} else {
|
|
/* prepare for next one */
|
|
gst_base_audio_decoder_flush (dec, FALSE);
|
|
/* and that's where we time from,
|
|
* in case upstream does not come up with anything better
|
|
* (e.g. upstream BYTE) */
|
|
if (format != GST_FORMAT_TIME) {
|
|
dec->priv->base_ts = start;
|
|
dec->priv->samples = 0;
|
|
}
|
|
}
|
|
|
|
/* and follow along with segment */
|
|
gst_segment_set_newsegment_full (&dec->segment, update, rate, arate,
|
|
format, start, stop, time);
|
|
|
|
dec->priv->pending_events =
|
|
g_list_append (dec->priv->pending_events, event);
|
|
handled = TRUE;
|
|
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
|
|
break;
|
|
}
|
|
|
|
case GST_EVENT_FLUSH_START:
|
|
break;
|
|
|
|
case GST_EVENT_FLUSH_STOP:
|
|
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
|
|
/* prepare for fresh start */
|
|
gst_base_audio_decoder_flush (dec, TRUE);
|
|
|
|
g_list_foreach (dec->priv->pending_events, (GFunc) gst_event_unref, NULL);
|
|
g_list_free (dec->priv->pending_events);
|
|
dec->priv->pending_events = NULL;
|
|
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
|
|
break;
|
|
|
|
case GST_EVENT_EOS:
|
|
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
|
|
gst_base_audio_decoder_drain (dec);
|
|
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
|
|
break;
|
|
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return handled;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_decoder_sink_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstBaseAudioDecoder *dec;
|
|
GstBaseAudioDecoderClass *klass;
|
|
gboolean handled = FALSE;
|
|
gboolean ret = TRUE;
|
|
|
|
dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
|
|
klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
|
|
|
|
GST_DEBUG_OBJECT (dec, "received event %d, %s", GST_EVENT_TYPE (event),
|
|
GST_EVENT_TYPE_NAME (event));
|
|
|
|
if (klass->event)
|
|
handled = klass->event (dec, event);
|
|
|
|
if (!handled)
|
|
handled = gst_base_audio_decoder_sink_eventfunc (dec, event);
|
|
|
|
if (!handled) {
|
|
/* Forward non-serialized events and EOS/FLUSH_STOP immediately.
|
|
* For EOS this is required because no buffer or serialized event
|
|
* will come after EOS and nothing could trigger another
|
|
* _finish_frame() call.
|
|
*
|
|
* For FLUSH_STOP this is required because it is expected
|
|
* to be forwarded immediately and no buffers are queued anyway.
|
|
*/
|
|
if (!GST_EVENT_IS_SERIALIZED (event)
|
|
|| GST_EVENT_TYPE (event) == GST_EVENT_EOS
|
|
|| GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP) {
|
|
ret = gst_pad_event_default (pad, event);
|
|
} else {
|
|
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
|
|
dec->priv->pending_events =
|
|
g_list_append (dec->priv->pending_events, event);
|
|
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
|
|
ret = TRUE;
|
|
}
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (dec, "event handled");
|
|
|
|
gst_object_unref (dec);
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_decoder_do_seek (GstBaseAudioDecoder * dec, GstEvent * event)
|
|
{
|
|
GstSeekFlags flags;
|
|
GstSeekType start_type, end_type;
|
|
GstFormat format;
|
|
gdouble rate;
|
|
gint64 start, start_time, end_time;
|
|
GstSegment seek_segment;
|
|
guint32 seqnum;
|
|
|
|
gst_event_parse_seek (event, &rate, &format, &flags, &start_type,
|
|
&start_time, &end_type, &end_time);
|
|
|
|
/* we'll handle plain open-ended flushing seeks with the simple approach */
|
|
if (rate != 1.0) {
|
|
GST_DEBUG_OBJECT (dec, "unsupported seek: rate");
|
|
return FALSE;
|
|
}
|
|
|
|
if (start_type != GST_SEEK_TYPE_SET) {
|
|
GST_DEBUG_OBJECT (dec, "unsupported seek: start time");
|
|
return FALSE;
|
|
}
|
|
|
|
if (end_type != GST_SEEK_TYPE_NONE ||
|
|
(end_type == GST_SEEK_TYPE_SET && end_time != GST_CLOCK_TIME_NONE)) {
|
|
GST_DEBUG_OBJECT (dec, "unsupported seek: end time");
|
|
return FALSE;
|
|
}
|
|
|
|
if (!(flags & GST_SEEK_FLAG_FLUSH)) {
|
|
GST_DEBUG_OBJECT (dec, "unsupported seek: not flushing");
|
|
return FALSE;
|
|
}
|
|
|
|
memcpy (&seek_segment, &dec->segment, sizeof (seek_segment));
|
|
gst_segment_set_seek (&seek_segment, rate, format, flags, start_type,
|
|
start_time, end_type, end_time, NULL);
|
|
start_time = seek_segment.last_stop;
|
|
|
|
format = GST_FORMAT_BYTES;
|
|
if (!gst_pad_query_convert (dec->sinkpad, GST_FORMAT_TIME, start_time,
|
|
&format, &start)) {
|
|
GST_DEBUG_OBJECT (dec, "conversion failed");
|
|
return FALSE;
|
|
}
|
|
|
|
seqnum = gst_event_get_seqnum (event);
|
|
event = gst_event_new_seek (1.0, GST_FORMAT_BYTES, flags,
|
|
GST_SEEK_TYPE_SET, start, GST_SEEK_TYPE_NONE, -1);
|
|
gst_event_set_seqnum (event, seqnum);
|
|
|
|
GST_DEBUG_OBJECT (dec, "seeking to %" GST_TIME_FORMAT " at byte offset %"
|
|
G_GINT64_FORMAT, GST_TIME_ARGS (start_time), start);
|
|
|
|
return gst_pad_push_event (dec->sinkpad, event);
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_decoder_src_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstBaseAudioDecoder *dec;
|
|
gboolean res = FALSE;
|
|
|
|
dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
|
|
|
|
GST_DEBUG_OBJECT (dec, "received event %d, %s", GST_EVENT_TYPE (event),
|
|
GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEEK:
|
|
{
|
|
GstFormat format, tformat;
|
|
gdouble rate;
|
|
GstSeekFlags flags;
|
|
GstSeekType cur_type, stop_type;
|
|
gint64 cur, stop;
|
|
gint64 tcur, tstop;
|
|
guint32 seqnum;
|
|
|
|
gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
|
|
&stop_type, &stop);
|
|
seqnum = gst_event_get_seqnum (event);
|
|
|
|
/* upstream gets a chance first */
|
|
if ((res = gst_pad_push_event (dec->sinkpad, event)))
|
|
break;
|
|
|
|
/* if upstream fails for a time seek, maybe we can help if allowed */
|
|
if (format == GST_FORMAT_TIME) {
|
|
if (gst_base_audio_decoder_do_byte (dec))
|
|
res = gst_base_audio_decoder_do_seek (dec, event);
|
|
break;
|
|
}
|
|
|
|
/* ... though a non-time seek can be aided as well */
|
|
/* First bring the requested format to time */
|
|
tformat = GST_FORMAT_TIME;
|
|
if (!(res = gst_pad_query_convert (pad, format, cur, &tformat, &tcur)))
|
|
goto convert_error;
|
|
if (!(res = gst_pad_query_convert (pad, format, stop, &tformat, &tstop)))
|
|
goto convert_error;
|
|
|
|
/* then seek with time on the peer */
|
|
event = gst_event_new_seek (rate, GST_FORMAT_TIME,
|
|
flags, cur_type, tcur, stop_type, tstop);
|
|
gst_event_set_seqnum (event, seqnum);
|
|
|
|
res = gst_pad_push_event (dec->sinkpad, event);
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_push_event (dec->sinkpad, event);
|
|
break;
|
|
}
|
|
done:
|
|
gst_object_unref (dec);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
convert_error:
|
|
{
|
|
GST_DEBUG_OBJECT (dec, "cannot convert start/stop for seek");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_decoder_sink_query (GstPad * pad, GstQuery * query)
|
|
{
|
|
gboolean res = TRUE;
|
|
GstBaseAudioDecoder *dec;
|
|
|
|
dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_FORMATS:
|
|
{
|
|
gst_query_set_formats (query, 2, GST_FORMAT_TIME, GST_FORMAT_BYTES);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
case GST_QUERY_CONVERT:
|
|
{
|
|
GstFormat src_fmt, dest_fmt;
|
|
gint64 src_val, dest_val;
|
|
|
|
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
|
|
if (!(res = gst_base_audio_encoded_audio_convert (&dec->ctx->state,
|
|
dec->priv->bytes_in, dec->priv->samples_out,
|
|
src_fmt, src_val, &dest_fmt, &dest_val)))
|
|
goto error;
|
|
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_query_default (pad, query);
|
|
break;
|
|
}
|
|
|
|
error:
|
|
gst_object_unref (dec);
|
|
return res;
|
|
}
|
|
|
|
static const GstQueryType *
|
|
gst_base_audio_decoder_get_query_types (GstPad * pad)
|
|
{
|
|
static const GstQueryType gst_base_audio_decoder_src_query_types[] = {
|
|
GST_QUERY_POSITION,
|
|
GST_QUERY_DURATION,
|
|
GST_QUERY_CONVERT,
|
|
GST_QUERY_LATENCY,
|
|
0
|
|
};
|
|
|
|
return gst_base_audio_decoder_src_query_types;
|
|
}
|
|
|
|
/* FIXME ? are any of these queries (other than latency) a decoder's business ??
|
|
* also, the conversion stuff might seem to make sense, but seems to not mind
|
|
* segment stuff etc at all
|
|
* Supposedly that's backward compatibility ... */
|
|
static gboolean
|
|
gst_base_audio_decoder_src_query (GstPad * pad, GstQuery * query)
|
|
{
|
|
GstBaseAudioDecoder *dec;
|
|
GstPad *peerpad;
|
|
gboolean res = FALSE;
|
|
|
|
dec = GST_BASE_AUDIO_DECODER (GST_PAD_PARENT (pad));
|
|
peerpad = gst_pad_get_peer (GST_PAD (dec->sinkpad));
|
|
|
|
GST_LOG_OBJECT (dec, "handling query: %" GST_PTR_FORMAT, query);
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_DURATION:
|
|
{
|
|
GstFormat format;
|
|
|
|
/* upstream in any case */
|
|
if ((res = gst_pad_query_default (pad, query)))
|
|
break;
|
|
|
|
gst_query_parse_duration (query, &format, NULL);
|
|
/* try answering TIME by converting from BYTE if subclass allows */
|
|
if (format == GST_FORMAT_TIME && gst_base_audio_decoder_do_byte (dec)) {
|
|
gint64 value;
|
|
|
|
format = GST_FORMAT_BYTES;
|
|
if (gst_pad_query_peer_duration (dec->sinkpad, &format, &value)) {
|
|
GST_LOG_OBJECT (dec, "upstream size %" G_GINT64_FORMAT, value);
|
|
format = GST_FORMAT_TIME;
|
|
if (gst_pad_query_convert (dec->sinkpad, GST_FORMAT_BYTES, value,
|
|
&format, &value)) {
|
|
gst_query_set_duration (query, GST_FORMAT_TIME, value);
|
|
res = TRUE;
|
|
}
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_POSITION:
|
|
{
|
|
GstFormat format;
|
|
gint64 time, value;
|
|
|
|
if ((res = gst_pad_peer_query (dec->sinkpad, query))) {
|
|
GST_LOG_OBJECT (dec, "returning peer response");
|
|
break;
|
|
}
|
|
|
|
/* we start from the last seen time */
|
|
time = dec->segment.last_stop;
|
|
/* correct for the segment values */
|
|
time = gst_segment_to_stream_time (&dec->segment, GST_FORMAT_TIME, time);
|
|
|
|
GST_LOG_OBJECT (dec,
|
|
"query %p: our time: %" GST_TIME_FORMAT, query, GST_TIME_ARGS (time));
|
|
|
|
/* and convert to the final format */
|
|
gst_query_parse_position (query, &format, NULL);
|
|
if (!(res = gst_pad_query_convert (pad, GST_FORMAT_TIME, time,
|
|
&format, &value)))
|
|
break;
|
|
|
|
gst_query_set_position (query, format, value);
|
|
|
|
GST_LOG_OBJECT (dec,
|
|
"query %p: we return %" G_GINT64_FORMAT " (format %u)", query, value,
|
|
format);
|
|
break;
|
|
}
|
|
case GST_QUERY_FORMATS:
|
|
{
|
|
gst_query_set_formats (query, 3,
|
|
GST_FORMAT_TIME, GST_FORMAT_BYTES, GST_FORMAT_DEFAULT);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
case GST_QUERY_CONVERT:
|
|
{
|
|
GstFormat src_fmt, dest_fmt;
|
|
gint64 src_val, dest_val;
|
|
|
|
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
|
|
if (!(res = gst_base_audio_raw_audio_convert (&dec->ctx->state,
|
|
src_fmt, src_val, &dest_fmt, &dest_val)))
|
|
break;
|
|
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
|
|
break;
|
|
}
|
|
case GST_QUERY_LATENCY:
|
|
{
|
|
if ((res = gst_pad_peer_query (dec->sinkpad, query))) {
|
|
gboolean live;
|
|
GstClockTime min_latency, max_latency;
|
|
|
|
gst_query_parse_latency (query, &live, &min_latency, &max_latency);
|
|
GST_DEBUG_OBJECT (dec, "Peer latency: live %d, min %"
|
|
GST_TIME_FORMAT " max %" GST_TIME_FORMAT, live,
|
|
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
|
|
|
|
GST_OBJECT_LOCK (dec);
|
|
/* add our latency */
|
|
if (min_latency != -1)
|
|
min_latency += dec->ctx->min_latency;
|
|
if (max_latency != -1)
|
|
max_latency += dec->ctx->max_latency;
|
|
GST_OBJECT_UNLOCK (dec);
|
|
|
|
gst_query_set_latency (query, live, min_latency, max_latency);
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_query_default (pad, query);
|
|
break;
|
|
}
|
|
|
|
gst_object_unref (peerpad);
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_decoder_stop (GstBaseAudioDecoder * dec)
|
|
{
|
|
GstBaseAudioDecoderClass *klass;
|
|
gboolean ret = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (dec, "gst_base_audio_decoder_stop");
|
|
|
|
klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
|
|
|
|
if (klass->stop) {
|
|
ret = klass->stop (dec);
|
|
}
|
|
|
|
/* clean up */
|
|
gst_base_audio_decoder_reset (dec, TRUE);
|
|
|
|
if (ret)
|
|
dec->priv->active = FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_decoder_start (GstBaseAudioDecoder * dec)
|
|
{
|
|
GstBaseAudioDecoderClass *klass;
|
|
gboolean ret = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (dec, "gst_base_audio_decoder_start");
|
|
|
|
klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
|
|
|
|
/* arrange clean state */
|
|
gst_base_audio_decoder_reset (dec, TRUE);
|
|
|
|
if (klass->start) {
|
|
ret = klass->start (dec);
|
|
}
|
|
|
|
if (ret)
|
|
dec->priv->active = TRUE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_base_audio_decoder_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstBaseAudioDecoder *dec;
|
|
|
|
dec = GST_BASE_AUDIO_DECODER (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_LATENCY:
|
|
g_value_set_int64 (value, dec->latency);
|
|
break;
|
|
case PROP_TOLERANCE:
|
|
g_value_set_int64 (value, dec->tolerance);
|
|
break;
|
|
case PROP_PLC:
|
|
g_value_set_boolean (value, dec->plc);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_audio_decoder_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstBaseAudioDecoder *dec;
|
|
|
|
dec = GST_BASE_AUDIO_DECODER (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_LATENCY:
|
|
dec->latency = g_value_get_int64 (value);
|
|
break;
|
|
case PROP_TOLERANCE:
|
|
dec->tolerance = g_value_get_int64 (value);
|
|
break;
|
|
case PROP_PLC:
|
|
dec->plc = g_value_get_boolean (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_base_audio_decoder_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstBaseAudioDecoder *codec;
|
|
GstStateChangeReturn ret;
|
|
|
|
codec = GST_BASE_AUDIO_DECODER (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
if (!gst_base_audio_decoder_start (codec)) {
|
|
goto start_failed;
|
|
}
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = parent_class->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
if (!gst_base_audio_decoder_stop (codec)) {
|
|
goto stop_failed;
|
|
}
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
|
|
start_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (codec, LIBRARY, INIT, (NULL), ("Failed to start codec"));
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
stop_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (codec, LIBRARY, INIT, (NULL), ("Failed to stop codec"));
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
}
|
|
|
|
GstFlowReturn
|
|
_gst_base_audio_decoder_error (GstBaseAudioDecoder * dec, gint weight,
|
|
GQuark domain, gint code, gchar * txt, gchar * dbg, const gchar * file,
|
|
const gchar * function, gint line)
|
|
{
|
|
if (txt)
|
|
GST_WARNING_OBJECT (dec, "error: %s", txt);
|
|
if (dbg)
|
|
GST_WARNING_OBJECT (dec, "error: %s", dbg);
|
|
dec->priv->error_count += weight;
|
|
dec->priv->discont = TRUE;
|
|
if (dec->ctx->max_errors < dec->priv->error_count) {
|
|
gst_element_message_full (GST_ELEMENT (dec), GST_MESSAGE_ERROR,
|
|
domain, code, txt, dbg, file, function, line);
|
|
return GST_FLOW_ERROR;
|
|
} else {
|
|
return GST_FLOW_OK;
|
|
}
|
|
}
|