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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1635 lines
51 KiB
C
1635 lines
51 KiB
C
/* GStreamer
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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:gstbaseaudioencoder
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* @short_description: Base class for audio encoders
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* @see_also: #GstBaseTransform
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*
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* This base class is for audio encoders turning raw audio samples into
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* encoded audio data.
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*
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* GstBaseAudioEncoder 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, GstBaseAudioEncoder calls @start when the encoder element
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* is activated, which allows subclass to perform any global setup.
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* </para></listitem>
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* <listitem><para>
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* GstBaseAudioEncoder calls @set_format to inform subclass of the format
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* of input audio data that it is about to receive. Subclass should
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* setup for encoding and configure various base class context parameters
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* appropriately, notably those directing desired input data handling.
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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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* GstBaseAudioEncoder 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, GstBaseAudioEncoder
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* provides a GstBaseAudioEncoderContext that provides required context,
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* e.g. describing the format of input audio data.
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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 sample data (as directed by the context's
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* frame_samples and frame_max) and provides this 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 encoded data, subclass should call
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* @gst_base_audio_encoder_finish_frame to have encoded data pushed
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* downstream. Alternatively, it might also call to indicate dropped
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* (non-encoded) samples.
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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.
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* </para></listitem>
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* <listitem><para>
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* During the parsing process GstBaseAudioEncoderClass 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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* GstBaseAudioEncoder 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_encoder_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 either favor tracking upstream timestamps
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* (at the possible expense of jitter) or aim to arrange for a perfect stream of
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* output timestamps, depending on #GstBaseAudioEncoder:perfect-ts.
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* However, in the latter case, the input may not be so perfect or ideal, which
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* is handled as follows. An input timestamp is compared with the expected
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* timestamp as dictated by input sample stream and if the deviation is less
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* than #GstBaseAudioEncoder:tolerance, the deviation is discarded.
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* Otherwise, it is considered a discontuinity and subsequent output timestamp
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* is resynced to the new position after performing configured discontinuity
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* processing. In the non-perfect-ts case, an upstream variation exceeding
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* tolerance only leads to marking DISCONT on subsequent outgoing
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* (while timestamps are adjusted to upstream regardless of variation).
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* While DISCONT is also marked in the perfect-ts case, this one optionally
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* (see #GstBaseAudioEncoder:hard-resync)
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* performs some additional steps, such as clipping of (early) input samples
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* or draining all currently remaining input data, depending on the direction
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* of the discontuinity.
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*
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* If perfect timestamps are arranged, it is also possible to request baseclass
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* (usually set by subclass) to provide additional buffer metadata (in OFFSET
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* and OFFSET_END) fields according to granule defined semantics currently
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* needed by oggmux. Specifically, OFFSET is set to granulepos (= sample count
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* including buffer) and OFFSET_END to corresponding timestamp (as determined
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* by same sample count and sample rate).
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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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* Inform base class of buffer processing needs using context's
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* frame_samples and frame_bytes.
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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, e.g. those controlling timestamp behaviour
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* and discontinuity processing.
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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_encoder_finish_frame.
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* </para></listitem>
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* </itemizedlist>
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*
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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 <stdlib.h>
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#include <string.h>
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#include "gstbaseaudioencoder.h"
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#include <gst/audio/audio.h>
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GST_DEBUG_CATEGORY_STATIC (gst_base_audio_encoder_debug);
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#define GST_CAT_DEFAULT gst_base_audio_encoder_debug
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#define GST_BASE_AUDIO_ENCODER_GET_PRIVATE(obj) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_ENCODER, \
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GstBaseAudioEncoderPrivate))
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enum
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{
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PROP_0,
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PROP_PERFECT_TS,
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PROP_GRANULE,
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PROP_HARD_RESYNC,
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PROP_TOLERANCE
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};
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#define DEFAULT_PERFECT_TS FALSE
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#define DEFAULT_GRANULE FALSE
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#define DEFAULT_HARD_RESYNC FALSE
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#define DEFAULT_TOLERANCE 40000000
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struct _GstBaseAudioEncoderPrivate
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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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* kept nearly constant for perfect_ts,
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* otherwise resyncs to upstream ts */
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GstClockTime base_ts;
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/* corresponding base granulepos */
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gint64 base_gp;
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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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/* currently collected sample data */
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GstAdapter *adapter;
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/* offset in adapter up to which already supplied to encoder */
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gint offset;
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/* mark outgoing discont */
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gboolean discont;
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/* to guess duration of drained data */
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GstClockTime last_duration;
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/* subclass provided data in processing round */
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gboolean got_data;
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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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/* output bps estimatation */
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/* global in samples seen */
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guint64 samples_in;
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/* global bytes sent out */
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guint64 bytes_out;
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/* context storage */
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GstBaseAudioEncoderContext ctx;
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};
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static GstElementClass *parent_class = NULL;
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static void gst_base_audio_encoder_class_init (GstBaseAudioEncoderClass *
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klass);
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static void gst_base_audio_encoder_init (GstBaseAudioEncoder * parse,
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GstBaseAudioEncoderClass * klass);
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GType
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gst_base_audio_encoder_get_type (void)
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{
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static GType base_audio_encoder_type = 0;
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if (!base_audio_encoder_type) {
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static const GTypeInfo base_audio_encoder_info = {
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sizeof (GstBaseAudioEncoderClass),
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(GBaseInitFunc) NULL,
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(GBaseFinalizeFunc) NULL,
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(GClassInitFunc) gst_base_audio_encoder_class_init,
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NULL,
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NULL,
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sizeof (GstBaseAudioEncoder),
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0,
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(GInstanceInitFunc) gst_base_audio_encoder_init,
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};
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const GInterfaceInfo preset_interface_info = {
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NULL, /* interface_init */
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NULL, /* interface_finalize */
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NULL /* interface_data */
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};
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base_audio_encoder_type = g_type_register_static (GST_TYPE_ELEMENT,
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"GstBaseAudioEncoder", &base_audio_encoder_info, G_TYPE_FLAG_ABSTRACT);
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g_type_add_interface_static (base_audio_encoder_type, GST_TYPE_PRESET,
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&preset_interface_info);
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}
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return base_audio_encoder_type;
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}
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static void gst_base_audio_encoder_finalize (GObject * object);
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static void gst_base_audio_encoder_reset (GstBaseAudioEncoder * enc,
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gboolean full);
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static void gst_base_audio_encoder_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_encoder_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static gboolean gst_base_audio_encoder_sink_activate_push (GstPad * pad,
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gboolean active);
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static gboolean gst_base_audio_encoder_sink_event (GstPad * pad,
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GstEvent * event);
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static gboolean gst_base_audio_encoder_sink_setcaps (GstPad * pad,
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GstCaps * caps);
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static GstFlowReturn gst_base_audio_encoder_chain (GstPad * pad,
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GstBuffer * buffer);
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static gboolean gst_base_audio_encoder_src_query (GstPad * pad,
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GstQuery * query);
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static gboolean gst_base_audio_encoder_sink_query (GstPad * pad,
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GstQuery * query);
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static const GstQueryType *gst_base_audio_encoder_get_query_types (GstPad *
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pad);
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static GstCaps *gst_base_audio_encoder_sink_getcaps (GstPad * pad);
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static void
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gst_base_audio_encoder_class_init (GstBaseAudioEncoderClass * klass)
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{
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GObjectClass *gobject_class;
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gobject_class = G_OBJECT_CLASS (klass);
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parent_class = g_type_class_peek_parent (klass);
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GST_DEBUG_CATEGORY_INIT (gst_base_audio_encoder_debug, "baseaudioencoder", 0,
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"baseaudioencoder element");
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g_type_class_add_private (klass, sizeof (GstBaseAudioEncoderPrivate));
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gobject_class->set_property = gst_base_audio_encoder_set_property;
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gobject_class->get_property = gst_base_audio_encoder_get_property;
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_base_audio_encoder_finalize);
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/* properties */
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g_object_class_install_property (gobject_class, PROP_PERFECT_TS,
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g_param_spec_boolean ("perfect-ts", "Perfect Timestamps",
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"Favour perfect timestamps over tracking upstream timestamps",
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DEFAULT_PERFECT_TS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_GRANULE,
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g_param_spec_boolean ("granule", "Granule Marking",
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"Apply granule semantics to buffer metadata (implies perfect-ts)",
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DEFAULT_GRANULE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_HARD_RESYNC,
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g_param_spec_boolean ("hard-resync", "Hard Resync",
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"Perform clipping and sample flushing upon discontinuity",
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DEFAULT_HARD_RESYNC, 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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"Consider discontinuity if timestamp jitter/imperfection exceeds tolerance",
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0, G_MAXINT64, DEFAULT_TOLERANCE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_base_audio_encoder_init (GstBaseAudioEncoder * enc,
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GstBaseAudioEncoderClass * bclass)
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{
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GstPadTemplate *pad_template;
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GST_DEBUG_OBJECT (enc, "gst_base_audio_encoder_init");
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enc->priv = GST_BASE_AUDIO_ENCODER_GET_PRIVATE (enc);
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/* only push mode supported */
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pad_template =
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gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "sink");
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g_return_if_fail (pad_template != NULL);
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enc->sinkpad = gst_pad_new_from_template (pad_template, "sink");
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gst_pad_set_event_function (enc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_event));
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gst_pad_set_setcaps_function (enc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_setcaps));
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gst_pad_set_getcaps_function (enc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_getcaps));
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gst_pad_set_query_function (enc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_query));
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gst_pad_set_chain_function (enc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_base_audio_encoder_chain));
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gst_pad_set_activatepush_function (enc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_activate_push));
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gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
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GST_DEBUG_OBJECT (enc, "sinkpad created");
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/* and we don't mind upstream traveling stuff that much ... */
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pad_template =
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gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "src");
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g_return_if_fail (pad_template != NULL);
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enc->srcpad = gst_pad_new_from_template (pad_template, "src");
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gst_pad_set_query_function (enc->srcpad,
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GST_DEBUG_FUNCPTR (gst_base_audio_encoder_src_query));
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gst_pad_set_query_type_function (enc->srcpad,
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GST_DEBUG_FUNCPTR (gst_base_audio_encoder_get_query_types));
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gst_pad_use_fixed_caps (enc->srcpad);
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gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
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GST_DEBUG_OBJECT (enc, "src created");
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enc->priv->adapter = gst_adapter_new ();
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enc->ctx = &enc->priv->ctx;
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/* property default */
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enc->perfect_ts = DEFAULT_PERFECT_TS;
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enc->hard_resync = DEFAULT_HARD_RESYNC;
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enc->tolerance = DEFAULT_TOLERANCE;
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/* init state */
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gst_base_audio_encoder_reset (enc, TRUE);
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GST_DEBUG_OBJECT (enc, "init ok");
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}
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static void
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gst_base_audio_encoder_reset (GstBaseAudioEncoder * enc, gboolean full)
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{
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GST_OBJECT_LOCK (enc);
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if (full) {
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enc->priv->active = FALSE;
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enc->priv->samples_in = 0;
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enc->priv->bytes_out = 0;
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memset (enc->ctx, 0, sizeof (enc->ctx));
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enc->ctx->state.bpf = 0;
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enc->ctx->state.rate = 0;
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enc->ctx->min_latency = 0;
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enc->ctx->max_latency = 0;
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g_free (enc->ctx->state.channel_pos);
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enc->ctx->state.channel_pos = NULL;
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}
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gst_segment_init (&enc->segment, GST_FORMAT_TIME);
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gst_adapter_clear (enc->priv->adapter);
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enc->priv->got_data = FALSE;
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enc->priv->drained = TRUE;
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enc->priv->offset = 0;
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enc->priv->base_ts = GST_CLOCK_TIME_NONE;
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enc->priv->base_gp = -1;
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enc->priv->samples = 0;
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enc->priv->discont = FALSE;
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GST_OBJECT_UNLOCK (enc);
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}
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static void
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gst_base_audio_encoder_finalize (GObject * object)
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{
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GstBaseAudioEncoder *enc = GST_BASE_AUDIO_ENCODER (object);
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g_object_unref (enc->priv->adapter);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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/**
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* gst_base_audio_encoder_finish_frame:
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* @enc: a #GstBaseAudioEncoder
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* @buffer: encoded data
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* @samples: number of samples (per channel) represented by encoded data
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*
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* Collects encoded data and/or pushes encoded data downstream.
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* Source pad caps must be set when this is called. Depending on the nature
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* of the (framing of) the format, subclass can decide whether to push
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* encoded data directly or to collect various "frames" in a single buffer.
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* Note that the latter behaviour is recommended whenever the format is allowed,
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* as it incurs no additional latency and avoids otherwise generating a
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* a multitude of (small) output buffers. If not explicitly pushed,
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* any available encoded data is pushed at the end of each processing cycle,
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* i.e. which encodes as much data as available input data allows.
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*
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* If @samples < 0, then best estimate is all samples provided to encoder
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* (subclass) so far. @buf may be NULL, in which case next number of @samples
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* are considered discarded, e.g. as a result of discontinuous transmission,
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* and a discontinuity is marked (note that @buf == NULL => push == TRUE).
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*
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* Returns: a #GstFlowReturn that should be escalated to caller (of caller)
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*/
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GstFlowReturn
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gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc, GstBuffer * buf,
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gint samples)
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{
|
|
GstBaseAudioEncoderClass *klass;
|
|
GstBaseAudioEncoderPrivate *priv;
|
|
GstBaseAudioEncoderContext *ctx;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
|
|
klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
|
|
priv = enc->priv;
|
|
ctx = enc->ctx;
|
|
|
|
/* subclass should know what it is producing by now */
|
|
g_return_val_if_fail (GST_PAD_CAPS (enc->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);
|
|
|
|
GST_LOG_OBJECT (enc, "accepting %d bytes encoded data as %d samples",
|
|
buf ? GST_BUFFER_SIZE (buf) : -1, samples);
|
|
|
|
/* mark subclass still alive and providing */
|
|
priv->got_data = TRUE;
|
|
|
|
/* remove corresponding samples from input */
|
|
if (samples < 0)
|
|
samples = (enc->priv->offset / ctx->state.bpf);
|
|
|
|
if (G_LIKELY (samples)) {
|
|
/* track upstream ts if so configured */
|
|
if (!enc->perfect_ts) {
|
|
guint64 ts, distance;
|
|
|
|
ts = gst_adapter_prev_timestamp (priv->adapter, &distance);
|
|
g_assert (distance % ctx->state.bpf == 0);
|
|
distance /= ctx->state.bpf;
|
|
GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past prev_ts %"
|
|
GST_TIME_FORMAT, distance, GST_TIME_ARGS (ts));
|
|
GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past base_ts %"
|
|
GST_TIME_FORMAT, priv->samples, GST_TIME_ARGS (priv->base_ts));
|
|
/* when draining adapter might be empty and no ts to offer */
|
|
if (GST_CLOCK_TIME_IS_VALID (ts) && ts != priv->base_ts) {
|
|
GstClockTimeDiff diff;
|
|
GstClockTime old_ts, next_ts;
|
|
|
|
/* passed into another buffer;
|
|
* mild check for discontinuity and only mark if so */
|
|
next_ts = ts +
|
|
gst_util_uint64_scale (distance, GST_SECOND, ctx->state.rate);
|
|
old_ts = priv->base_ts +
|
|
gst_util_uint64_scale (priv->samples, GST_SECOND, ctx->state.rate);
|
|
diff = GST_CLOCK_DIFF (next_ts, old_ts);
|
|
GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
|
|
/* only mark discontinuity if beyond tolerance */
|
|
if (G_UNLIKELY (diff < -enc->tolerance || diff > enc->tolerance)) {
|
|
GST_DEBUG_OBJECT (enc, "marked discont");
|
|
priv->discont = TRUE;
|
|
}
|
|
GST_LOG_OBJECT (enc, "new upstream ts %" GST_TIME_FORMAT
|
|
" at distance %" G_GUINT64_FORMAT, GST_TIME_ARGS (ts), distance);
|
|
/* re-sync to upstream ts */
|
|
priv->base_ts = ts;
|
|
priv->samples = distance;
|
|
}
|
|
}
|
|
/* advance sample view */
|
|
if (G_UNLIKELY (samples * ctx->state.bpf > priv->offset)) {
|
|
if (G_LIKELY (!priv->force)) {
|
|
/* no way we can let this pass */
|
|
g_assert_not_reached ();
|
|
/* really no way */
|
|
goto overflow;
|
|
} else {
|
|
priv->offset = 0;
|
|
if (samples * ctx->state.bpf >= gst_adapter_available (priv->adapter))
|
|
gst_adapter_clear (priv->adapter);
|
|
else
|
|
gst_adapter_flush (priv->adapter, samples * ctx->state.bpf);
|
|
}
|
|
} else {
|
|
gst_adapter_flush (priv->adapter, samples * ctx->state.bpf);
|
|
priv->offset -= samples * ctx->state.bpf;
|
|
/* avoid subsequent stray prev_ts */
|
|
if (G_UNLIKELY (gst_adapter_available (priv->adapter) == 0))
|
|
gst_adapter_clear (priv->adapter);
|
|
}
|
|
/* sample count advanced below after buffer handling */
|
|
}
|
|
|
|
/* collect output */
|
|
if (G_LIKELY (buf)) {
|
|
GST_LOG_OBJECT (enc, "taking %d bytes for output", GST_BUFFER_SIZE (buf));
|
|
buf = gst_buffer_make_metadata_writable (buf);
|
|
|
|
/* decorate */
|
|
gst_buffer_set_caps (buf, GST_PAD_CAPS (enc->srcpad));
|
|
if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
|
|
/* FIXME ? lookahead could lead to weird ts and duration ?
|
|
* (particularly if not in perfect mode) */
|
|
/* mind sample rounding and produce perfect output */
|
|
GST_BUFFER_TIMESTAMP (buf) = priv->base_ts +
|
|
gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
|
|
ctx->state.rate);
|
|
GST_DEBUG_OBJECT (enc, "out samples %d", samples);
|
|
if (G_LIKELY (samples > 0)) {
|
|
priv->samples += samples;
|
|
GST_BUFFER_DURATION (buf) = priv->base_ts +
|
|
gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
|
|
ctx->state.rate) - GST_BUFFER_TIMESTAMP (buf);
|
|
priv->last_duration = GST_BUFFER_DURATION (buf);
|
|
} else {
|
|
/* duration forecast in case of handling remainder;
|
|
* the last one is probably like the previous one ... */
|
|
GST_BUFFER_DURATION (buf) = priv->last_duration;
|
|
}
|
|
if (priv->base_gp >= 0) {
|
|
/* pamper oggmux */
|
|
/* FIXME: in longer run, muxer should take care of this ... */
|
|
/* offset_end = granulepos for ogg muxer */
|
|
GST_BUFFER_OFFSET_END (buf) = priv->base_gp + priv->samples -
|
|
enc->ctx->lookahead;
|
|
/* offset = timestamp corresponding to granulepos for ogg muxer */
|
|
GST_BUFFER_OFFSET (buf) =
|
|
GST_FRAMES_TO_CLOCK_TIME (GST_BUFFER_OFFSET_END (buf),
|
|
ctx->state.rate);
|
|
} else {
|
|
GST_BUFFER_OFFSET (buf) = priv->bytes_out;
|
|
GST_BUFFER_OFFSET_END (buf) = priv->bytes_out + GST_BUFFER_SIZE (buf);
|
|
}
|
|
}
|
|
|
|
priv->bytes_out += GST_BUFFER_SIZE (buf);
|
|
|
|
if (G_UNLIKELY (priv->discont)) {
|
|
GST_LOG_OBJECT (enc, "marking discont");
|
|
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
|
|
priv->discont = FALSE;
|
|
}
|
|
|
|
if (klass->pre_push) {
|
|
/* last chance for subclass to do some dirty stuff */
|
|
ret = klass->pre_push (enc, &buf);
|
|
if (ret != GST_FLOW_OK || !buf) {
|
|
GST_DEBUG_OBJECT (enc, "subclass returned %s, buf %p",
|
|
gst_flow_get_name (ret), buf);
|
|
if (buf)
|
|
gst_buffer_unref (buf);
|
|
goto exit;
|
|
}
|
|
}
|
|
|
|
GST_LOG_OBJECT (enc, "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)));
|
|
|
|
ret = gst_pad_push (enc->srcpad, buf);
|
|
GST_LOG_OBJECT (enc, "buffer pushed: %s", gst_flow_get_name (ret));
|
|
} else {
|
|
/* merely advance samples, most work for that already done above */
|
|
priv->samples += samples;
|
|
}
|
|
|
|
exit:
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
overflow:
|
|
{
|
|
GST_ELEMENT_ERROR (enc, STREAM, ENCODE,
|
|
("received more encoded samples %d than provided %d",
|
|
samples, priv->offset / ctx->state.bpf), (NULL));
|
|
if (buf)
|
|
gst_buffer_unref (buf);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
/* adapter tracking idea:
|
|
* - start of adapter corresponds with what has already been encoded
|
|
* (i.e. really returned by encoder subclass)
|
|
* - start + offset is what needs to be fed to subclass next */
|
|
static GstFlowReturn
|
|
gst_base_audio_encoder_push_buffers (GstBaseAudioEncoder * enc, gboolean force)
|
|
{
|
|
GstBaseAudioEncoderClass *klass;
|
|
GstBaseAudioEncoderPrivate *priv;
|
|
GstBaseAudioEncoderContext *ctx;
|
|
gint av, need;
|
|
GstBuffer *buf;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
|
|
klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
|
|
|
|
g_return_val_if_fail (klass->handle_frame != NULL, GST_FLOW_ERROR);
|
|
|
|
priv = enc->priv;
|
|
ctx = enc->ctx;
|
|
|
|
while (ret == GST_FLOW_OK) {
|
|
|
|
buf = NULL;
|
|
av = gst_adapter_available (priv->adapter);
|
|
|
|
g_assert (priv->offset <= av);
|
|
av -= priv->offset;
|
|
|
|
need = ctx->frame_samples > 0 ? ctx->frame_samples * ctx->state.bpf : av;
|
|
GST_LOG_OBJECT (enc, "available: %d, needed: %d, force: %d",
|
|
av, need, force);
|
|
|
|
if ((need > av) || !av) {
|
|
if (G_UNLIKELY (force)) {
|
|
priv->force = TRUE;
|
|
need = av;
|
|
} else {
|
|
break;
|
|
}
|
|
} else {
|
|
priv->force = FALSE;
|
|
}
|
|
|
|
/* if we have some extra metadata,
|
|
* provide for integer multiple of frames to allow for better granularity
|
|
* of processing */
|
|
if (ctx->frame_samples > 0 && need) {
|
|
if (ctx->frame_max > 1)
|
|
need = need * MIN ((av / need), ctx->frame_max);
|
|
else if (ctx->frame_max == 0)
|
|
need = need * (av / need);
|
|
}
|
|
|
|
if (need) {
|
|
buf = gst_buffer_new ();
|
|
GST_BUFFER_DATA (buf) = (guint8 *)
|
|
gst_adapter_peek (priv->adapter, priv->offset + need) + priv->offset;
|
|
GST_BUFFER_SIZE (buf) = need;
|
|
}
|
|
|
|
GST_LOG_OBJECT (enc, "providing subclass with %d bytes at offset %d",
|
|
need, priv->offset);
|
|
|
|
/* mark this already as consumed,
|
|
* which it should be when subclass gives us data in exchange for samples */
|
|
priv->offset += need;
|
|
priv->samples_in += need / ctx->state.bpf;
|
|
|
|
priv->got_data = FALSE;
|
|
ret = klass->handle_frame (enc, buf);
|
|
|
|
if (G_LIKELY (buf))
|
|
gst_buffer_unref (buf);
|
|
|
|
/* no data to feed, no leftover provided, then bail out */
|
|
if (G_UNLIKELY (!buf && !priv->got_data)) {
|
|
priv->drained = TRUE;
|
|
GST_LOG_OBJECT (enc, "no more data drained from subclass");
|
|
break;
|
|
}
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_audio_encoder_drain (GstBaseAudioEncoder * enc)
|
|
{
|
|
if (enc->priv->drained)
|
|
return GST_FLOW_OK;
|
|
else
|
|
return gst_base_audio_encoder_push_buffers (enc, TRUE);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_audio_encoder_chain (GstPad * pad, GstBuffer * buffer)
|
|
{
|
|
GstBaseAudioEncoderClass *bclass;
|
|
GstBaseAudioEncoder *enc;
|
|
GstBaseAudioEncoderPrivate *priv;
|
|
GstBaseAudioEncoderContext *ctx;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
gboolean discont;
|
|
|
|
enc = GST_BASE_AUDIO_ENCODER (GST_OBJECT_PARENT (pad));
|
|
bclass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
|
|
|
|
priv = enc->priv;
|
|
ctx = enc->ctx;
|
|
|
|
/* should know what is coming by now */
|
|
if (!ctx->state.bpf)
|
|
goto not_negotiated;
|
|
|
|
GST_LOG_OBJECT (enc,
|
|
"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)));
|
|
|
|
/* input shoud be whole number of sample frames */
|
|
if (GST_BUFFER_SIZE (buffer) % ctx->state.bpf)
|
|
goto wrong_buffer;
|
|
|
|
#ifndef GST_DISABLE_GST_DEBUG
|
|
{
|
|
GstClockTime duration;
|
|
GstClockTimeDiff diff;
|
|
|
|
/* verify buffer duration */
|
|
duration = gst_util_uint64_scale (GST_BUFFER_SIZE (buffer), GST_SECOND,
|
|
ctx->state.rate * ctx->state.bpf);
|
|
diff = GST_CLOCK_DIFF (duration, GST_BUFFER_DURATION (buffer));
|
|
if (GST_BUFFER_DURATION (buffer) != GST_CLOCK_TIME_NONE &&
|
|
(diff > GST_SECOND / ctx->state.rate / 2 ||
|
|
diff < -GST_SECOND / ctx->state.rate / 2)) {
|
|
GST_DEBUG_OBJECT (enc, "incoming buffer had incorrect duration %"
|
|
GST_TIME_FORMAT ", expected duration %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)),
|
|
GST_TIME_ARGS (duration));
|
|
}
|
|
}
|
|
#endif
|
|
|
|
discont = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT);
|
|
if (G_UNLIKELY (discont)) {
|
|
GST_LOG_OBJECT (buffer, "marked discont");
|
|
enc->priv->discont = discont;
|
|
}
|
|
|
|
/* clip to segment */
|
|
/* NOTE: slightly painful linking -laudio only for this one ... */
|
|
buffer = gst_audio_buffer_clip (buffer, &enc->segment, ctx->state.rate,
|
|
ctx->state.bpf);
|
|
if (G_UNLIKELY (!buffer)) {
|
|
GST_DEBUG_OBJECT (buffer, "no data after clipping to segment");
|
|
goto done;
|
|
}
|
|
|
|
GST_LOG_OBJECT (enc,
|
|
"buffer after segment clipping has 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)));
|
|
|
|
if (!GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
|
|
priv->base_ts = GST_BUFFER_TIMESTAMP (buffer);
|
|
GST_DEBUG_OBJECT (enc, "new base ts %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (priv->base_ts));
|
|
if (enc->granule) {
|
|
priv->base_gp =
|
|
GST_CLOCK_TIME_TO_FRAMES (priv->base_ts, enc->ctx->state.rate);
|
|
GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT,
|
|
GST_TIME_ARGS (priv->base_gp));
|
|
}
|
|
}
|
|
|
|
/* check for continuity;
|
|
* checked elsewhere in non-perfect case */
|
|
if (enc->perfect_ts) {
|
|
GstClockTimeDiff diff = 0;
|
|
GstClockTime next_ts = 0;
|
|
|
|
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer) &&
|
|
GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
|
|
guint64 samples;
|
|
|
|
samples = priv->samples +
|
|
gst_adapter_available (priv->adapter) / ctx->state.bpf;
|
|
next_ts = priv->base_ts +
|
|
gst_util_uint64_scale (samples, GST_SECOND, ctx->state.rate);
|
|
GST_LOG_OBJECT (enc, "buffer is %" G_GUINT64_FORMAT
|
|
" 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, GST_BUFFER_TIMESTAMP (buffer));
|
|
GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
|
|
/* if within tolerance,
|
|
* discard buffer ts and carry on producing perfect stream,
|
|
* otherwise clip or resync to ts */
|
|
if (G_UNLIKELY (diff < -enc->tolerance || diff > enc->tolerance)) {
|
|
GST_DEBUG_OBJECT (enc, "marked discont");
|
|
discont = TRUE;
|
|
}
|
|
}
|
|
|
|
/* do some fancy tweaking in hard resync case */
|
|
if (discont && enc->hard_resync) {
|
|
if (diff < 0) {
|
|
guint64 diff_bytes;
|
|
|
|
GST_WARNING_OBJECT (enc, "Buffer is older than expected ts %"
|
|
GST_TIME_FORMAT ". Clipping buffer", GST_TIME_ARGS (next_ts));
|
|
|
|
diff_bytes =
|
|
GST_CLOCK_TIME_TO_FRAMES (-diff, ctx->state.rate) * ctx->state.bpf;
|
|
if (diff_bytes >= GST_BUFFER_SIZE (buffer)) {
|
|
gst_buffer_unref (buffer);
|
|
goto done;
|
|
}
|
|
buffer = gst_buffer_make_metadata_writable (buffer);
|
|
GST_BUFFER_DATA (buffer) += diff_bytes;
|
|
GST_BUFFER_SIZE (buffer) -= diff_bytes;
|
|
|
|
GST_BUFFER_TIMESTAMP (buffer) += diff;
|
|
/* care even less about duration after this */
|
|
} else {
|
|
/* drain stuff prior to resync */
|
|
gst_base_audio_encoder_drain (enc);
|
|
}
|
|
}
|
|
/* now re-sync ts */
|
|
priv->base_ts += diff;
|
|
if (priv->base_gp >= 0)
|
|
priv->base_gp =
|
|
GST_CLOCK_TIME_TO_FRAMES (priv->base_ts, enc->ctx->state.rate);
|
|
priv->discont |= discont;
|
|
}
|
|
|
|
gst_adapter_push (enc->priv->adapter, buffer);
|
|
/* new stuff, so we can push subclass again */
|
|
enc->priv->drained = FALSE;
|
|
|
|
ret = gst_base_audio_encoder_push_buffers (enc, FALSE);
|
|
|
|
done:
|
|
GST_LOG_OBJECT (enc, "chain leaving");
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
not_negotiated:
|
|
{
|
|
GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
|
|
("encoder not initialized"));
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_NOT_NEGOTIATED;
|
|
}
|
|
wrong_buffer:
|
|
{
|
|
GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
|
|
("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buffer),
|
|
ctx->state.bpf));
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
#define CHECK_VALUE(res, var, val) \
|
|
if (!res) \
|
|
goto refuse_caps; \
|
|
if (var != val) \
|
|
changed = TRUE; \
|
|
var = val;
|
|
|
|
static gboolean
|
|
gst_base_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
|
|
{
|
|
GstBaseAudioEncoder *enc;
|
|
GstBaseAudioEncoderClass *klass;
|
|
GstBaseAudioEncoderContext *ctx;
|
|
GstAudioState *state;
|
|
gboolean res = TRUE, changed = FALSE;
|
|
GstStructure *s;
|
|
gboolean vb;
|
|
gint vi;
|
|
|
|
enc = GST_BASE_AUDIO_ENCODER (GST_PAD_PARENT (pad));
|
|
klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
|
|
|
|
/* subclass must do something here ... */
|
|
g_return_val_if_fail (klass->set_format != NULL, FALSE);
|
|
|
|
ctx = enc->ctx;
|
|
state = &ctx->state;
|
|
|
|
GST_DEBUG_OBJECT (enc, "caps: %" GST_PTR_FORMAT, caps);
|
|
|
|
if (!gst_caps_is_fixed (caps))
|
|
goto refuse_caps;
|
|
|
|
s = gst_caps_get_structure (caps, 0);
|
|
/* parse caps here to save subclass the trouble */
|
|
if (gst_structure_has_name (s, "audio/x-raw-int"))
|
|
state->is_int = TRUE;
|
|
else if (gst_structure_has_name (s, "audio/x-raw-float"))
|
|
state->is_int = FALSE;
|
|
else
|
|
goto refuse_caps;
|
|
|
|
res = gst_structure_get_int (s, "rate", &vi);
|
|
CHECK_VALUE (res, state->rate, vi);
|
|
res &= gst_structure_get_int (s, "channels", &vi);
|
|
CHECK_VALUE (res, state->channels, vi);
|
|
res &= gst_structure_get_int (s, "width", &vi);
|
|
CHECK_VALUE (res, state->width, vi);
|
|
res &= (!state->is_int || gst_structure_get_int (s, "depth", &vi));
|
|
CHECK_VALUE (res, state->depth, vi);
|
|
res &= gst_structure_get_int (s, "endianness", &vi);
|
|
CHECK_VALUE (res, state->endian, vi);
|
|
res &= (!state->is_int || gst_structure_get_boolean (s, "signed", &vb));
|
|
CHECK_VALUE (res, state->sign, vb);
|
|
|
|
state->bpf = (state->width / 8) * state->channels;
|
|
GST_LOG_OBJECT (enc, "bpf: %d", state->bpf);
|
|
if (!state->bpf)
|
|
goto refuse_caps;
|
|
|
|
g_free (state->channel_pos);
|
|
state->channel_pos = gst_audio_get_channel_positions (s);
|
|
|
|
if (changed) {
|
|
GstClockTime old_min_latency;
|
|
GstClockTime old_max_latency;
|
|
|
|
/* drain any pending old data stuff */
|
|
gst_base_audio_encoder_drain (enc);
|
|
|
|
/* context defaults */
|
|
enc->ctx->frame_samples = 0;
|
|
enc->ctx->frame_max = 0;
|
|
enc->ctx->lookahead = 0;
|
|
|
|
/* element might report latency */
|
|
GST_OBJECT_LOCK (enc);
|
|
old_min_latency = ctx->min_latency;
|
|
old_max_latency = ctx->max_latency;
|
|
GST_OBJECT_UNLOCK (enc);
|
|
|
|
if (klass->set_format)
|
|
res = klass->set_format (enc, state);
|
|
|
|
/* notify if new latency */
|
|
GST_OBJECT_LOCK (enc);
|
|
if ((ctx->min_latency > 0 && ctx->min_latency != old_min_latency) ||
|
|
(ctx->max_latency > 0 && ctx->max_latency != old_max_latency)) {
|
|
GST_OBJECT_UNLOCK (enc);
|
|
/* post latency message on the bus */
|
|
gst_element_post_message (GST_ELEMENT (enc),
|
|
gst_message_new_latency (GST_OBJECT (enc)));
|
|
GST_OBJECT_LOCK (enc);
|
|
}
|
|
GST_OBJECT_UNLOCK (enc);
|
|
} else {
|
|
GST_DEBUG_OBJECT (enc, "new audio format identical to configured format");
|
|
}
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
refuse_caps:
|
|
{
|
|
GST_WARNING_OBJECT (enc, "rejected caps %" GST_PTR_FORMAT, caps);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_base_audio_encoder_proxy_getcaps:
|
|
* @enc: a #GstBaseAudioEncoder
|
|
* @caps: initial
|
|
*
|
|
* Returns caps that express @caps (or sink template caps if @caps == NULL)
|
|
* restricted to channel/rate combinations supported by downstream elements
|
|
* (e.g. muxers).
|
|
*
|
|
* Returns: a #GstCaps owned by caller
|
|
*/
|
|
GstCaps *
|
|
gst_base_audio_encoder_proxy_getcaps (GstBaseAudioEncoder * enc, GstCaps * caps)
|
|
{
|
|
const GstCaps *templ_caps;
|
|
GstCaps *allowed = NULL;
|
|
GstCaps *fcaps, *filter_caps;
|
|
gint i, j;
|
|
|
|
/* we want to be able to communicate to upstream elements like audioconvert
|
|
* and audioresample any rate/channel restrictions downstream (e.g. muxer
|
|
* only accepting certain sample rates) */
|
|
templ_caps = caps ? caps : gst_pad_get_pad_template_caps (enc->sinkpad);
|
|
allowed = gst_pad_get_allowed_caps (enc->srcpad);
|
|
if (!allowed || gst_caps_is_empty (allowed) || gst_caps_is_any (allowed)) {
|
|
fcaps = gst_caps_copy (templ_caps);
|
|
goto done;
|
|
}
|
|
|
|
GST_LOG_OBJECT (enc, "template caps %" GST_PTR_FORMAT, templ_caps);
|
|
GST_LOG_OBJECT (enc, "allowed caps %" GST_PTR_FORMAT, allowed);
|
|
|
|
filter_caps = gst_caps_new_empty ();
|
|
|
|
for (i = 0; i < gst_caps_get_size (templ_caps); i++) {
|
|
GQuark q_name;
|
|
|
|
q_name = gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));
|
|
|
|
/* pick rate + channel fields from allowed caps */
|
|
for (j = 0; j < gst_caps_get_size (allowed); j++) {
|
|
const GstStructure *allowed_s = gst_caps_get_structure (allowed, j);
|
|
const GValue *val;
|
|
GstStructure *s;
|
|
|
|
s = gst_structure_id_empty_new (q_name);
|
|
if ((val = gst_structure_get_value (allowed_s, "rate")))
|
|
gst_structure_set_value (s, "rate", val);
|
|
if ((val = gst_structure_get_value (allowed_s, "channels")))
|
|
gst_structure_set_value (s, "channels", val);
|
|
|
|
gst_caps_merge_structure (filter_caps, s);
|
|
}
|
|
}
|
|
|
|
fcaps = gst_caps_intersect (filter_caps, templ_caps);
|
|
gst_caps_unref (filter_caps);
|
|
|
|
done:
|
|
gst_caps_replace (&allowed, NULL);
|
|
|
|
GST_LOG_OBJECT (enc, "proxy caps %" GST_PTR_FORMAT, fcaps);
|
|
|
|
return fcaps;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_base_audio_encoder_sink_getcaps (GstPad * pad)
|
|
{
|
|
GstBaseAudioEncoder *enc;
|
|
GstBaseAudioEncoderClass *klass;
|
|
GstCaps *caps;
|
|
|
|
enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
|
|
klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
|
|
g_assert (pad == enc->sinkpad);
|
|
|
|
if (klass->getcaps)
|
|
caps = klass->getcaps (enc);
|
|
else
|
|
caps = gst_base_audio_encoder_proxy_getcaps (enc, NULL);
|
|
gst_object_unref (enc);
|
|
|
|
GST_LOG_OBJECT (enc, "returning caps %" GST_PTR_FORMAT, caps);
|
|
|
|
return caps;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_encoder_sink_eventfunc (GstBaseAudioEncoder * enc,
|
|
GstEvent * event)
|
|
{
|
|
GstBaseAudioEncoderClass *klass;
|
|
gboolean handled = FALSE;
|
|
|
|
klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_NEWSEGMENT:
|
|
{
|
|
GstFormat format;
|
|
gdouble rate, arate;
|
|
gint64 start, stop, time;
|
|
gboolean update;
|
|
|
|
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
|
|
&start, &stop, &time);
|
|
|
|
if (format == GST_FORMAT_TIME) {
|
|
GST_DEBUG_OBJECT (enc, "received TIME NEW_SEGMENT %" GST_TIME_FORMAT
|
|
" -- %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT
|
|
", accum %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (enc->segment.start),
|
|
GST_TIME_ARGS (enc->segment.stop),
|
|
GST_TIME_ARGS (enc->segment.time),
|
|
GST_TIME_ARGS (enc->segment.accum));
|
|
} else {
|
|
GST_DEBUG_OBJECT (enc, "received NEW_SEGMENT %" G_GINT64_FORMAT
|
|
" -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT
|
|
", accum %" G_GINT64_FORMAT,
|
|
enc->segment.start, enc->segment.stop,
|
|
enc->segment.time, enc->segment.accum);
|
|
GST_DEBUG_OBJECT (enc, "unsupported format; ignoring");
|
|
break;
|
|
}
|
|
|
|
/* finish current segment */
|
|
gst_base_audio_encoder_drain (enc);
|
|
/* reset partially for new segment */
|
|
gst_base_audio_encoder_reset (enc, FALSE);
|
|
/* and follow along with segment */
|
|
gst_segment_set_newsegment_full (&enc->segment, update, rate, arate,
|
|
format, start, stop, time);
|
|
break;
|
|
}
|
|
|
|
case GST_EVENT_FLUSH_START:
|
|
break;
|
|
|
|
case GST_EVENT_FLUSH_STOP:
|
|
/* discard any pending stuff */
|
|
// TODO route through drain ?
|
|
if (!enc->priv->drained && klass->flush)
|
|
klass->flush (enc);
|
|
/* and get (re)set for the sequel */
|
|
gst_base_audio_encoder_reset (enc, FALSE);
|
|
break;
|
|
|
|
case GST_EVENT_EOS:
|
|
gst_base_audio_encoder_drain (enc);
|
|
break;
|
|
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return handled;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_encoder_sink_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstBaseAudioEncoder *enc;
|
|
GstBaseAudioEncoderClass *klass;
|
|
gboolean handled = FALSE;
|
|
gboolean ret = TRUE;
|
|
|
|
enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
|
|
klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
|
|
|
|
GST_DEBUG_OBJECT (enc, "received event %d, %s", GST_EVENT_TYPE (event),
|
|
GST_EVENT_TYPE_NAME (event));
|
|
|
|
if (klass->event)
|
|
handled = klass->event (enc, event);
|
|
|
|
if (!handled)
|
|
handled = gst_base_audio_encoder_sink_eventfunc (enc, event);
|
|
|
|
if (!handled)
|
|
ret = gst_pad_event_default (pad, event);
|
|
|
|
GST_DEBUG_OBJECT (enc, "event handled");
|
|
|
|
gst_object_unref (enc);
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_encoder_convert_sink (GstPad * pad, GstFormat src_format,
|
|
gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
|
|
{
|
|
GstBaseAudioEncoder *enc;
|
|
gboolean res = FALSE;
|
|
guint scale = 1;
|
|
gint bytes_per_sample, rate, byterate;
|
|
|
|
enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
|
|
|
|
bytes_per_sample = enc->ctx->state.bpf;
|
|
rate = enc->ctx->state.rate;
|
|
byterate = bytes_per_sample * rate;
|
|
|
|
if (G_UNLIKELY (bytes_per_sample == 0 || rate == 0)) {
|
|
GST_DEBUG_OBJECT (enc, "not enough metadata yet to convert");
|
|
goto exit;
|
|
}
|
|
|
|
switch (src_format) {
|
|
case GST_FORMAT_BYTES:
|
|
switch (*dest_format) {
|
|
case GST_FORMAT_DEFAULT:
|
|
*dest_value = src_value / bytes_per_sample;
|
|
res = TRUE;
|
|
break;
|
|
case GST_FORMAT_TIME:
|
|
*dest_value =
|
|
gst_util_uint64_scale_int (src_value, GST_SECOND, byterate);
|
|
res = TRUE;
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
}
|
|
break;
|
|
case GST_FORMAT_DEFAULT:
|
|
switch (*dest_format) {
|
|
case GST_FORMAT_BYTES:
|
|
*dest_value = src_value * bytes_per_sample;
|
|
res = TRUE;
|
|
break;
|
|
case GST_FORMAT_TIME:
|
|
*dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND, rate);
|
|
res = TRUE;
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
}
|
|
break;
|
|
case GST_FORMAT_TIME:
|
|
switch (*dest_format) {
|
|
case GST_FORMAT_BYTES:
|
|
scale = bytes_per_sample;
|
|
/* fallthrough */
|
|
case GST_FORMAT_DEFAULT:
|
|
*dest_value = gst_util_uint64_scale_int (src_value,
|
|
scale * rate, GST_SECOND);
|
|
res = TRUE;
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
}
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
}
|
|
|
|
exit:
|
|
gst_object_unref (enc);
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_encoder_sink_query (GstPad * pad, GstQuery * query)
|
|
{
|
|
gboolean res = TRUE;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
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_encoder_convert_sink (pad, 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:
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_encoder_convert_src (GstPad * pad, GstFormat src_format,
|
|
gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
|
|
{
|
|
GstBaseAudioEncoder *enc;
|
|
gboolean res = FALSE;
|
|
gint64 avg;
|
|
|
|
enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
|
|
|
|
if (enc->priv->samples_in == 0 ||
|
|
enc->priv->bytes_out == 0 || enc->ctx->state.rate == 0) {
|
|
GST_DEBUG_OBJECT (enc, "not enough metadata yet to convert");
|
|
goto exit;
|
|
}
|
|
|
|
avg = (enc->priv->bytes_out * enc->ctx->state.rate) / (enc->priv->samples_in);
|
|
|
|
switch (src_format) {
|
|
case GST_FORMAT_BYTES:
|
|
switch (*dest_format) {
|
|
case GST_FORMAT_TIME:
|
|
*dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND, avg);
|
|
res = TRUE;
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
}
|
|
break;
|
|
case GST_FORMAT_TIME:
|
|
switch (*dest_format) {
|
|
case GST_FORMAT_BYTES:
|
|
*dest_value = gst_util_uint64_scale_int (src_value, avg, GST_SECOND);
|
|
res = TRUE;
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
}
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
}
|
|
|
|
exit:
|
|
gst_object_unref (enc);
|
|
return res;
|
|
}
|
|
|
|
static const GstQueryType *
|
|
gst_base_audio_encoder_get_query_types (GstPad * pad)
|
|
{
|
|
static const GstQueryType gst_base_audio_encoder_src_query_types[] = {
|
|
GST_QUERY_POSITION,
|
|
GST_QUERY_DURATION,
|
|
GST_QUERY_CONVERT,
|
|
GST_QUERY_LATENCY,
|
|
0
|
|
};
|
|
|
|
return gst_base_audio_encoder_src_query_types;
|
|
}
|
|
|
|
/* FIXME ? are any of these queries (other than latency) an encoder'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_encoder_src_query (GstPad * pad, GstQuery * query)
|
|
{
|
|
GstBaseAudioEncoder *enc;
|
|
GstPad *peerpad;
|
|
gboolean res = FALSE;
|
|
|
|
enc = GST_BASE_AUDIO_ENCODER (GST_PAD_PARENT (pad));
|
|
peerpad = gst_pad_get_peer (GST_PAD (enc->sinkpad));
|
|
|
|
GST_LOG_OBJECT (enc, "handling query: %" GST_PTR_FORMAT, query);
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_POSITION:
|
|
{
|
|
GstFormat fmt, req_fmt;
|
|
gint64 pos, val;
|
|
|
|
if ((res = gst_pad_peer_query (pad, query))) {
|
|
GST_LOG_OBJECT (enc, "returning peer response");
|
|
break;
|
|
}
|
|
|
|
if (!peerpad) {
|
|
GST_LOG_OBJECT (enc, "no peer");
|
|
break;
|
|
}
|
|
|
|
gst_query_parse_position (query, &req_fmt, NULL);
|
|
fmt = GST_FORMAT_TIME;
|
|
if (!(res = gst_pad_query_position (peerpad, &fmt, &pos)))
|
|
break;
|
|
|
|
if ((res = gst_pad_query_convert (peerpad, fmt, pos, &req_fmt, &val))) {
|
|
gst_query_set_position (query, req_fmt, val);
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_DURATION:
|
|
{
|
|
GstFormat fmt, req_fmt;
|
|
gint64 dur, val;
|
|
|
|
if ((res = gst_pad_peer_query (pad, query))) {
|
|
GST_LOG_OBJECT (enc, "returning peer response");
|
|
break;
|
|
}
|
|
|
|
if (!peerpad) {
|
|
GST_LOG_OBJECT (enc, "no peer");
|
|
break;
|
|
}
|
|
|
|
gst_query_parse_duration (query, &req_fmt, NULL);
|
|
fmt = GST_FORMAT_TIME;
|
|
if (!(res = gst_pad_query_duration (peerpad, &fmt, &dur)))
|
|
break;
|
|
|
|
if ((res = gst_pad_query_convert (peerpad, fmt, dur, &req_fmt, &val))) {
|
|
gst_query_set_duration (query, req_fmt, val);
|
|
}
|
|
break;
|
|
}
|
|
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_encoder_convert_src (pad, 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 (pad, query))) {
|
|
gboolean live;
|
|
GstClockTime min_latency, max_latency;
|
|
|
|
gst_query_parse_latency (query, &live, &min_latency, &max_latency);
|
|
GST_DEBUG_OBJECT (enc, "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 (enc);
|
|
/* add our latency */
|
|
if (min_latency != -1)
|
|
min_latency += enc->ctx->min_latency;
|
|
if (max_latency != -1)
|
|
max_latency += enc->ctx->max_latency;
|
|
GST_OBJECT_UNLOCK (enc);
|
|
|
|
gst_query_set_latency (query, live, min_latency, max_latency);
|
|
}
|
|
}
|
|
default:
|
|
res = gst_pad_query_default (pad, query);
|
|
break;
|
|
}
|
|
|
|
gst_object_unref (peerpad);
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_base_audio_encoder_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstBaseAudioEncoder *enc;
|
|
|
|
enc = GST_BASE_AUDIO_ENCODER (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_PERFECT_TS:
|
|
if (enc->granule && !g_value_get_boolean (value))
|
|
GST_WARNING_OBJECT (enc, "perfect-ts can not be set FALSE");
|
|
else
|
|
enc->perfect_ts = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_HARD_RESYNC:
|
|
enc->hard_resync = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_TOLERANCE:
|
|
enc->tolerance = g_value_get_int64 (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_audio_encoder_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstBaseAudioEncoder *enc;
|
|
|
|
enc = GST_BASE_AUDIO_ENCODER (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_PERFECT_TS:
|
|
g_value_set_boolean (value, enc->perfect_ts);
|
|
break;
|
|
case PROP_GRANULE:
|
|
g_value_set_boolean (value, enc->granule);
|
|
break;
|
|
case PROP_HARD_RESYNC:
|
|
g_value_set_boolean (value, enc->hard_resync);
|
|
break;
|
|
case PROP_TOLERANCE:
|
|
g_value_set_int64 (value, enc->tolerance);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_audio_encoder_activate (GstBaseAudioEncoder * enc, gboolean active)
|
|
{
|
|
GstBaseAudioEncoderClass *klass;
|
|
gboolean result = FALSE;
|
|
|
|
klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
|
|
|
|
g_return_val_if_fail (!enc->granule || enc->perfect_ts, FALSE);
|
|
|
|
GST_DEBUG_OBJECT (enc, "activate %d", active);
|
|
|
|
if (active) {
|
|
if (!enc->priv->active && klass->start)
|
|
result = klass->start (enc);
|
|
} else {
|
|
/* We must make sure streaming has finished before resetting things
|
|
* and calling the ::stop vfunc */
|
|
GST_PAD_STREAM_LOCK (enc->sinkpad);
|
|
GST_PAD_STREAM_UNLOCK (enc->sinkpad);
|
|
|
|
if (enc->priv->active && klass->stop)
|
|
result = klass->stop (enc);
|
|
|
|
/* clean up */
|
|
gst_base_audio_encoder_reset (enc, TRUE);
|
|
}
|
|
GST_DEBUG_OBJECT (enc, "activate return: %d", result);
|
|
return result;
|
|
}
|
|
|
|
|
|
static gboolean
|
|
gst_base_audio_encoder_sink_activate_push (GstPad * pad, gboolean active)
|
|
{
|
|
gboolean result = TRUE;
|
|
GstBaseAudioEncoder *enc;
|
|
|
|
enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
|
|
|
|
GST_DEBUG_OBJECT (enc, "sink activate push %d", active);
|
|
|
|
result = gst_base_audio_encoder_activate (enc, active);
|
|
|
|
if (result)
|
|
enc->priv->active = active;
|
|
|
|
GST_DEBUG_OBJECT (enc, "sink activate push return: %d", result);
|
|
|
|
gst_object_unref (enc);
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_base_audio_encoder_add_streamheader:
|
|
* @caps: a #GstCaps
|
|
* @buf: header buffers
|
|
*
|
|
* Adds given buffers to an array of buffers set as streamheader field
|
|
* on the given @caps. List of buffer arguments must be NULL-terminated.
|
|
*
|
|
* Returns: input caps with a streamheader field added, or NULL if some error
|
|
*/
|
|
GstCaps *
|
|
gst_base_audio_encoder_add_streamheader (GstCaps * caps, GstBuffer * buf, ...)
|
|
{
|
|
GstStructure *structure = NULL;
|
|
va_list va;
|
|
GValue array = { 0 };
|
|
GValue value = { 0 };
|
|
|
|
g_return_val_if_fail (caps != NULL, NULL);
|
|
g_return_val_if_fail (gst_caps_is_fixed (caps), NULL);
|
|
|
|
caps = gst_caps_make_writable (caps);
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
|
|
g_value_init (&array, GST_TYPE_ARRAY);
|
|
|
|
va_start (va, buf);
|
|
/* put buffers in a fixed list */
|
|
while (buf) {
|
|
g_assert (gst_buffer_is_metadata_writable (buf));
|
|
|
|
/* mark buffer */
|
|
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
|
|
|
|
g_value_init (&value, GST_TYPE_BUFFER);
|
|
buf = gst_buffer_copy (buf);
|
|
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
|
|
gst_value_set_buffer (&value, buf);
|
|
gst_buffer_unref (buf);
|
|
gst_value_array_append_value (&array, &value);
|
|
g_value_unset (&value);
|
|
|
|
buf = va_arg (va, GstBuffer *);
|
|
}
|
|
|
|
gst_structure_set_value (structure, "streamheader", &array);
|
|
g_value_unset (&array);
|
|
|
|
return caps;
|
|
}
|