/* * Initially based on gst-omx/omx/gstomxvideodec.c * * Copyright (C) 2011, Hewlett-Packard Development Company, L.P. * Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>, Collabora Ltd. * * Copyright (C) 2012, Collabora Ltd. * Author: Sebastian Dröge <sebastian.droege@collabora.co.uk> * * Copyright (C) 2015, Sebastian Dröge <sebastian@centricular.com> * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation * version 2.1 of the License. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with this library; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include <gst/gst.h> #include <gst/audio/audio.h> #include <string.h> #ifdef HAVE_ORC #include <orc/orc.h> #else #define orc_memcpy memcpy #endif #include "gstamcaudiodec.h" #include "gstamc-constants.h" GST_DEBUG_CATEGORY_STATIC (gst_amc_audio_dec_debug_category); #define GST_CAT_DEFAULT gst_amc_audio_dec_debug_category #define GST_AUDIO_DECODER_ERROR_FROM_ERROR(el, err) G_STMT_START { \ gchar *__dbg = g_strdup (err->message); \ GstAudioDecoder *__dec = GST_AUDIO_DECODER (el); \ GST_WARNING_OBJECT (el, "error: %s", __dbg); \ _gst_audio_decoder_error (__dec, 1, \ err->domain, err->code, \ NULL, __dbg, __FILE__, GST_FUNCTION, __LINE__); \ g_clear_error (&err); \ } G_STMT_END /* prototypes */ static void gst_amc_audio_dec_finalize (GObject * object); static GstStateChangeReturn gst_amc_audio_dec_change_state (GstElement * element, GstStateChange transition); static gboolean gst_amc_audio_dec_open (GstAudioDecoder * decoder); static gboolean gst_amc_audio_dec_close (GstAudioDecoder * decoder); static gboolean gst_amc_audio_dec_start (GstAudioDecoder * decoder); static gboolean gst_amc_audio_dec_stop (GstAudioDecoder * decoder); static gboolean gst_amc_audio_dec_set_format (GstAudioDecoder * decoder, GstCaps * caps); static void gst_amc_audio_dec_flush (GstAudioDecoder * decoder, gboolean hard); static GstFlowReturn gst_amc_audio_dec_handle_frame (GstAudioDecoder * decoder, GstBuffer * buffer); static GstFlowReturn gst_amc_audio_dec_drain (GstAmcAudioDec * self); enum { PROP_0 }; /* class initialization */ static void gst_amc_audio_dec_class_init (GstAmcAudioDecClass * klass); static void gst_amc_audio_dec_init (GstAmcAudioDec * self); static void gst_amc_audio_dec_base_init (gpointer g_class); static GstAudioDecoderClass *parent_class = NULL; GType gst_amc_audio_dec_get_type (void) { static volatile gsize type = 0; if (g_once_init_enter (&type)) { GType _type; static const GTypeInfo info = { sizeof (GstAmcAudioDecClass), gst_amc_audio_dec_base_init, NULL, (GClassInitFunc) gst_amc_audio_dec_class_init, NULL, NULL, sizeof (GstAmcAudioDec), 0, (GInstanceInitFunc) gst_amc_audio_dec_init, NULL }; _type = g_type_register_static (GST_TYPE_AUDIO_DECODER, "GstAmcAudioDec", &info, 0); GST_DEBUG_CATEGORY_INIT (gst_amc_audio_dec_debug_category, "amcaudiodec", 0, "Android MediaCodec audio decoder"); g_once_init_leave (&type, _type); } return type; } static const gchar * caps_to_mime (GstCaps * caps) { GstStructure *s; const gchar *name; s = gst_caps_get_structure (caps, 0); if (!s) return NULL; name = gst_structure_get_name (s); if (strcmp (name, "audio/mpeg") == 0) { gint mpegversion; if (!gst_structure_get_int (s, "mpegversion", &mpegversion)) return NULL; if (mpegversion == 1) { gint layer; if (!gst_structure_get_int (s, "layer", &layer) || layer == 3) return "audio/mpeg"; else if (layer == 2) return "audio/mpeg-L2"; } else if (mpegversion == 2 || mpegversion == 4) { return "audio/mp4a-latm"; } } else if (strcmp (name, "audio/AMR") == 0) { return "audio/3gpp"; } else if (strcmp (name, "audio/AMR-WB") == 0) { return "audio/amr-wb"; } else if (strcmp (name, "audio/x-alaw") == 0) { return "audio/g711-alaw"; } else if (strcmp (name, "audio/x-mulaw") == 0) { return "audio/g711-mlaw"; } else if (strcmp (name, "audio/x-vorbis") == 0) { return "audio/vorbis"; } return NULL; } static void gst_amc_audio_dec_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); GstAmcAudioDecClass *amcaudiodec_class = GST_AMC_AUDIO_DEC_CLASS (g_class); const GstAmcCodecInfo *codec_info; GstPadTemplate *templ; GstCaps *sink_caps, *src_caps; gchar *longname; codec_info = g_type_get_qdata (G_TYPE_FROM_CLASS (g_class), gst_amc_codec_info_quark); /* This happens for the base class and abstract subclasses */ if (!codec_info) return; amcaudiodec_class->codec_info = codec_info; gst_amc_codec_info_to_caps (codec_info, &sink_caps, &src_caps); /* Add pad templates */ templ = gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, sink_caps); gst_element_class_add_pad_template (element_class, templ); gst_caps_unref (sink_caps); templ = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS, src_caps); gst_element_class_add_pad_template (element_class, templ); gst_caps_unref (src_caps); longname = g_strdup_printf ("Android MediaCodec %s", codec_info->name); gst_element_class_set_metadata (element_class, codec_info->name, "Codec/Decoder/Audio", longname, "Sebastian Dröge <sebastian.droege@collabora.co.uk>"); g_free (longname); } static void gst_amc_audio_dec_class_init (GstAmcAudioDecClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstAudioDecoderClass *audiodec_class = GST_AUDIO_DECODER_CLASS (klass); parent_class = g_type_class_peek_parent (klass); gobject_class->finalize = gst_amc_audio_dec_finalize; element_class->change_state = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_change_state); audiodec_class->start = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_start); audiodec_class->stop = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_stop); audiodec_class->open = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_open); audiodec_class->close = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_close); audiodec_class->flush = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_flush); audiodec_class->set_format = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_set_format); audiodec_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_handle_frame); } static void gst_amc_audio_dec_init (GstAmcAudioDec * self) { gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (self), TRUE); gst_audio_decoder_set_drainable (GST_AUDIO_DECODER (self), TRUE); g_mutex_init (&self->drain_lock); g_cond_init (&self->drain_cond); self->output_adapter = gst_adapter_new (); } static gboolean gst_amc_audio_dec_open (GstAudioDecoder * decoder) { GstAmcAudioDec *self = GST_AMC_AUDIO_DEC (decoder); GstAmcAudioDecClass *klass = GST_AMC_AUDIO_DEC_GET_CLASS (self); GError *err = NULL; GST_DEBUG_OBJECT (self, "Opening decoder"); self->codec = gst_amc_codec_new (klass->codec_info->name, &err); if (!self->codec) { GST_ELEMENT_ERROR_FROM_ERROR (self, err); return FALSE; } self->started = FALSE; self->flushing = TRUE; GST_DEBUG_OBJECT (self, "Opened decoder"); return TRUE; } static gboolean gst_amc_audio_dec_close (GstAudioDecoder * decoder) { GstAmcAudioDec *self = GST_AMC_AUDIO_DEC (decoder); GST_DEBUG_OBJECT (self, "Closing decoder"); if (self->codec) { GError *err = NULL; gst_amc_codec_release (self->codec, &err); if (err) GST_ELEMENT_WARNING_FROM_ERROR (self, err); gst_amc_codec_free (self->codec); } self->codec = NULL; self->started = FALSE; self->flushing = TRUE; GST_DEBUG_OBJECT (self, "Closed decoder"); return TRUE; } static void gst_amc_audio_dec_finalize (GObject * object) { GstAmcAudioDec *self = GST_AMC_AUDIO_DEC (object); if (self->output_adapter) gst_object_unref (self->output_adapter); self->output_adapter = NULL; g_mutex_clear (&self->drain_lock); g_cond_clear (&self->drain_cond); G_OBJECT_CLASS (parent_class)->finalize (object); } static GstStateChangeReturn gst_amc_audio_dec_change_state (GstElement * element, GstStateChange transition) { GstAmcAudioDec *self; GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; GError *err = NULL; g_return_val_if_fail (GST_IS_AMC_AUDIO_DEC (element), GST_STATE_CHANGE_FAILURE); self = GST_AMC_AUDIO_DEC (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: self->downstream_flow_ret = GST_FLOW_OK; self->draining = FALSE; self->started = FALSE; break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; case GST_STATE_CHANGE_PAUSED_TO_READY: self->flushing = TRUE; gst_amc_codec_flush (self->codec, &err); if (err) GST_ELEMENT_WARNING_FROM_ERROR (self, err); g_mutex_lock (&self->drain_lock); self->draining = FALSE; g_cond_broadcast (&self->drain_cond); g_mutex_unlock (&self->drain_lock); break; default: break; } if (ret == GST_STATE_CHANGE_FAILURE) return ret; ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); if (ret == GST_STATE_CHANGE_FAILURE) return ret; switch (transition) { case GST_STATE_CHANGE_PLAYING_TO_PAUSED: break; case GST_STATE_CHANGE_PAUSED_TO_READY: self->downstream_flow_ret = GST_FLOW_FLUSHING; self->started = FALSE; break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } return ret; } static gboolean gst_amc_audio_dec_set_src_caps (GstAmcAudioDec * self, GstAmcFormat * format) { gint rate, channels; guint32 channel_mask = 0; GstAudioChannelPosition to[64]; GError *err = NULL; if (!gst_amc_format_get_int (format, "sample-rate", &rate, &err) || !gst_amc_format_get_int (format, "channel-count", &channels, &err)) { GST_ERROR_OBJECT (self, "Failed to get output format metadata: %s", err->message); g_clear_error (&err); return FALSE; } if (rate == 0 || channels == 0) { GST_ERROR_OBJECT (self, "Rate or channels not set"); return FALSE; } /* Not always present */ if (gst_amc_format_contains_key (format, "channel-mask", NULL)) gst_amc_format_get_int (format, "channel-mask", (gint *) & channel_mask, NULL); gst_amc_audio_channel_mask_to_positions (channel_mask, channels, self->positions); memcpy (to, self->positions, sizeof (to)); gst_audio_channel_positions_to_valid_order (to, channels); self->needs_reorder = (memcmp (self->positions, to, sizeof (GstAudioChannelPosition) * channels) != 0); if (self->needs_reorder) gst_audio_get_channel_reorder_map (channels, self->positions, to, self->reorder_map); gst_audio_info_init (&self->info); gst_audio_info_set_format (&self->info, GST_AUDIO_FORMAT_S16, rate, channels, to); if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (self), &self->info)) return FALSE; self->input_caps_changed = FALSE; return TRUE; } static void gst_amc_audio_dec_loop (GstAmcAudioDec * self) { GstFlowReturn flow_ret = GST_FLOW_OK; gboolean is_eos; GstAmcBuffer *buf; GstAmcBufferInfo buffer_info; gint idx; GError *err = NULL; GST_AUDIO_DECODER_STREAM_LOCK (self); retry: /*if (self->input_caps_changed) { idx = INFO_OUTPUT_FORMAT_CHANGED; } else { */ GST_DEBUG_OBJECT (self, "Waiting for available output buffer"); GST_AUDIO_DECODER_STREAM_UNLOCK (self); /* Wait at most 100ms here, some codecs don't fail dequeueing if * the codec is flushing, causing deadlocks during shutdown */ idx = gst_amc_codec_dequeue_output_buffer (self->codec, &buffer_info, 100000, &err); GST_AUDIO_DECODER_STREAM_LOCK (self); /*} */ if (idx < 0) { if (self->flushing) { g_clear_error (&err); goto flushing; } switch (idx) { case INFO_OUTPUT_BUFFERS_CHANGED: /* Handled internally */ g_assert_not_reached (); break; case INFO_OUTPUT_FORMAT_CHANGED:{ GstAmcFormat *format; gchar *format_string; GST_DEBUG_OBJECT (self, "Output format has changed"); format = gst_amc_codec_get_output_format (self->codec, &err); if (!format) goto format_error; format_string = gst_amc_format_to_string (format, &err); if (err) { gst_amc_format_free (format); goto format_error; } GST_DEBUG_OBJECT (self, "Got new output format: %s", format_string); g_free (format_string); if (!gst_amc_audio_dec_set_src_caps (self, format)) { gst_amc_format_free (format); goto format_error; } gst_amc_format_free (format); goto retry; } case INFO_TRY_AGAIN_LATER: GST_DEBUG_OBJECT (self, "Dequeueing output buffer timed out"); goto retry; case G_MININT: GST_ERROR_OBJECT (self, "Failure dequeueing output buffer"); goto dequeue_error; default: g_assert_not_reached (); break; } goto retry; } GST_DEBUG_OBJECT (self, "Got output buffer at index %d: offset %d size %d time %" G_GINT64_FORMAT " flags 0x%08x", idx, buffer_info.offset, buffer_info.size, buffer_info.presentation_time_us, buffer_info.flags); is_eos = ! !(buffer_info.flags & BUFFER_FLAG_END_OF_STREAM); buf = gst_amc_codec_get_output_buffer (self->codec, idx, &err); if (err) goto failed_to_get_output_buffer; else if (!buf) goto got_null_output_buffer; if (buffer_info.size > 0) { GstBuffer *outbuf; GstMapInfo minfo; /* This sometimes happens at EOS or if the input is not properly framed, * let's handle it gracefully by allocating a new buffer for the current * caps and filling it */ if (buffer_info.size % self->info.bpf != 0) goto invalid_buffer_size; outbuf = gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (self), buffer_info.size); if (!outbuf) goto failed_allocate; gst_buffer_map (outbuf, &minfo, GST_MAP_WRITE); if (self->needs_reorder) { gint i, n_samples, c, n_channels; gint *reorder_map = self->reorder_map; gint16 *dest, *source; dest = (gint16 *) minfo.data; source = (gint16 *) (buf->data + buffer_info.offset); n_samples = buffer_info.size / self->info.bpf; n_channels = self->info.channels; for (i = 0; i < n_samples; i++) { for (c = 0; c < n_channels; c++) { dest[i * n_channels + reorder_map[c]] = source[i * n_channels + c]; } } } else { orc_memcpy (minfo.data, buf->data + buffer_info.offset, buffer_info.size); } gst_buffer_unmap (outbuf, &minfo); if (self->spf != -1) { gst_adapter_push (self->output_adapter, outbuf); } else { flow_ret = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf, 1); } } gst_amc_buffer_free (buf); buf = NULL; if (self->spf != -1) { GstBuffer *outbuf; guint avail = gst_adapter_available (self->output_adapter); guint nframes; /* On EOS we take the complete adapter content, no matter * if it is a multiple of the codec frame size or not. * Otherwise we take a multiple of codec frames and push * them downstream */ avail /= self->info.bpf; if (!is_eos) { nframes = avail / self->spf; avail = nframes * self->spf; } else { nframes = (avail + self->spf - 1) / self->spf; } avail *= self->info.bpf; if (avail > 0) { outbuf = gst_adapter_take_buffer (self->output_adapter, avail); flow_ret = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf, nframes); } } if (!gst_amc_codec_release_output_buffer (self->codec, idx, FALSE, &err)) { if (self->flushing) { g_clear_error (&err); goto flushing; } goto failed_release; } if (is_eos || flow_ret == GST_FLOW_EOS) { GST_AUDIO_DECODER_STREAM_UNLOCK (self); g_mutex_lock (&self->drain_lock); if (self->draining) { GST_DEBUG_OBJECT (self, "Drained"); self->draining = FALSE; g_cond_broadcast (&self->drain_cond); } else if (flow_ret == GST_FLOW_OK) { GST_DEBUG_OBJECT (self, "Component signalled EOS"); flow_ret = GST_FLOW_EOS; } g_mutex_unlock (&self->drain_lock); GST_AUDIO_DECODER_STREAM_LOCK (self); } else { GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret)); } self->downstream_flow_ret = flow_ret; if (flow_ret != GST_FLOW_OK) goto flow_error; GST_AUDIO_DECODER_STREAM_UNLOCK (self); return; dequeue_error: { GST_ELEMENT_ERROR_FROM_ERROR (self, err); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_ERROR; GST_AUDIO_DECODER_STREAM_UNLOCK (self); g_mutex_lock (&self->drain_lock); self->draining = FALSE; g_cond_broadcast (&self->drain_cond); g_mutex_unlock (&self->drain_lock); return; } format_error: { if (err) GST_ELEMENT_ERROR_FROM_ERROR (self, err); else GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL), ("Failed to handle format")); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_ERROR; GST_AUDIO_DECODER_STREAM_UNLOCK (self); g_mutex_lock (&self->drain_lock); self->draining = FALSE; g_cond_broadcast (&self->drain_cond); g_mutex_unlock (&self->drain_lock); return; } failed_release: { GST_AUDIO_DECODER_ERROR_FROM_ERROR (self, err); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_ERROR; GST_AUDIO_DECODER_STREAM_UNLOCK (self); g_mutex_lock (&self->drain_lock); self->draining = FALSE; g_cond_broadcast (&self->drain_cond); g_mutex_unlock (&self->drain_lock); return; } flushing: { GST_DEBUG_OBJECT (self, "Flushing -- stopping task"); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_FLUSHING; GST_AUDIO_DECODER_STREAM_UNLOCK (self); return; } flow_error: { if (flow_ret == GST_FLOW_EOS) { GST_DEBUG_OBJECT (self, "EOS"); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); } else if (flow_ret < GST_FLOW_EOS) { GST_ELEMENT_ERROR (self, STREAM, FAILED, ("Internal data stream error."), ("stream stopped, reason %s", gst_flow_get_name (flow_ret))); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); } else if (flow_ret == GST_FLOW_FLUSHING) { GST_DEBUG_OBJECT (self, "Flushing -- stopping task"); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); } GST_AUDIO_DECODER_STREAM_UNLOCK (self); g_mutex_lock (&self->drain_lock); self->draining = FALSE; g_cond_broadcast (&self->drain_cond); g_mutex_unlock (&self->drain_lock); return; } failed_to_get_output_buffer: { GST_AUDIO_DECODER_ERROR_FROM_ERROR (self, err); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_ERROR; GST_AUDIO_DECODER_STREAM_UNLOCK (self); g_mutex_lock (&self->drain_lock); self->draining = FALSE; g_cond_broadcast (&self->drain_cond); g_mutex_unlock (&self->drain_lock); return; } got_null_output_buffer: { GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Got no output buffer")); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_ERROR; GST_AUDIO_DECODER_STREAM_UNLOCK (self); g_mutex_lock (&self->drain_lock); self->draining = FALSE; g_cond_broadcast (&self->drain_cond); g_mutex_unlock (&self->drain_lock); return; } invalid_buffer_size: { GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL), ("Invalid buffer size %u (bfp %d)", buffer_info.size, self->info.bpf)); gst_amc_codec_release_output_buffer (self->codec, idx, FALSE, &err); if (err && !self->flushing) GST_ELEMENT_WARNING_FROM_ERROR (self, err); g_clear_error (&err); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_ERROR; GST_AUDIO_DECODER_STREAM_UNLOCK (self); g_mutex_lock (&self->drain_lock); self->draining = FALSE; g_cond_broadcast (&self->drain_cond); g_mutex_unlock (&self->drain_lock); return; } failed_allocate: { GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to allocate output buffer")); gst_amc_codec_release_output_buffer (self->codec, idx, FALSE, &err); if (err && !self->flushing) GST_ELEMENT_WARNING_FROM_ERROR (self, err); g_clear_error (&err); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_ERROR; GST_AUDIO_DECODER_STREAM_UNLOCK (self); g_mutex_lock (&self->drain_lock); self->draining = FALSE; g_cond_broadcast (&self->drain_cond); g_mutex_unlock (&self->drain_lock); return; } } static gboolean gst_amc_audio_dec_start (GstAudioDecoder * decoder) { GstAmcAudioDec *self; self = GST_AMC_AUDIO_DEC (decoder); self->last_upstream_ts = 0; self->drained = TRUE; self->downstream_flow_ret = GST_FLOW_OK; self->started = FALSE; self->flushing = TRUE; return TRUE; } static gboolean gst_amc_audio_dec_stop (GstAudioDecoder * decoder) { GstAmcAudioDec *self; GError *err = NULL; self = GST_AMC_AUDIO_DEC (decoder); GST_DEBUG_OBJECT (self, "Stopping decoder"); self->flushing = TRUE; if (self->started) { gst_amc_codec_flush (self->codec, &err); if (err) GST_ELEMENT_WARNING_FROM_ERROR (self, err); gst_amc_codec_stop (self->codec, &err); if (err) GST_ELEMENT_WARNING_FROM_ERROR (self, err); self->started = FALSE; } gst_pad_stop_task (GST_AUDIO_DECODER_SRC_PAD (decoder)); memset (self->positions, 0, sizeof (self->positions)); gst_adapter_flush (self->output_adapter, gst_adapter_available (self->output_adapter)); g_list_foreach (self->codec_datas, (GFunc) g_free, NULL); g_list_free (self->codec_datas); self->codec_datas = NULL; self->downstream_flow_ret = GST_FLOW_FLUSHING; self->drained = TRUE; g_mutex_lock (&self->drain_lock); self->draining = FALSE; g_cond_broadcast (&self->drain_cond); g_mutex_unlock (&self->drain_lock); GST_DEBUG_OBJECT (self, "Stopped decoder"); return TRUE; } static gboolean gst_amc_audio_dec_set_format (GstAudioDecoder * decoder, GstCaps * caps) { GstAmcAudioDec *self; GstStructure *s; GstAmcFormat *format; const gchar *mime; gboolean is_format_change = FALSE; gboolean needs_disable = FALSE; gchar *format_string; gint rate, channels; GError *err = NULL; self = GST_AMC_AUDIO_DEC (decoder); GST_DEBUG_OBJECT (self, "Setting new caps %" GST_PTR_FORMAT, caps); /* Check if the caps change is a real format change or if only irrelevant * parts of the caps have changed or nothing at all. */ is_format_change |= (!self->input_caps || !gst_caps_is_equal (self->input_caps, caps)); needs_disable = self->started; /* If the component is not started and a real format change happens * we have to restart the component. If no real format change * happened we can just exit here. */ if (needs_disable && !is_format_change) { /* Framerate or something minor changed */ self->input_caps_changed = TRUE; GST_DEBUG_OBJECT (self, "Already running and caps did not change the format"); return TRUE; } if (needs_disable && is_format_change) { gst_amc_audio_dec_drain (self); GST_AUDIO_DECODER_STREAM_UNLOCK (self); gst_amc_audio_dec_stop (GST_AUDIO_DECODER (self)); GST_AUDIO_DECODER_STREAM_LOCK (self); gst_amc_audio_dec_close (GST_AUDIO_DECODER (self)); if (!gst_amc_audio_dec_open (GST_AUDIO_DECODER (self))) { GST_ERROR_OBJECT (self, "Failed to open codec again"); return FALSE; } if (!gst_amc_audio_dec_start (GST_AUDIO_DECODER (self))) { GST_ERROR_OBJECT (self, "Failed to start codec again"); } } /* srcpad task is not running at this point */ mime = caps_to_mime (caps); if (!mime) { GST_ERROR_OBJECT (self, "Failed to convert caps to mime"); return FALSE; } s = gst_caps_get_structure (caps, 0); if (!gst_structure_get_int (s, "rate", &rate) || !gst_structure_get_int (s, "channels", &channels)) { GST_ERROR_OBJECT (self, "Failed to get rate/channels"); return FALSE; } format = gst_amc_format_new_audio (mime, rate, channels, &err); if (!format) { GST_ELEMENT_ERROR_FROM_ERROR (self, err); return FALSE; } /* FIXME: These buffers needs to be valid until the codec is stopped again */ g_list_foreach (self->codec_datas, (GFunc) gst_buffer_unref, NULL); g_list_free (self->codec_datas); self->codec_datas = NULL; if (gst_structure_has_field (s, "codec_data")) { const GValue *h = gst_structure_get_value (s, "codec_data"); GstBuffer *codec_data = gst_value_get_buffer (h); GstMapInfo minfo; guint8 *data; gst_buffer_map (codec_data, &minfo, GST_MAP_READ); data = g_memdup (minfo.data, minfo.size); self->codec_datas = g_list_prepend (self->codec_datas, data); gst_amc_format_set_buffer (format, "csd-0", data, minfo.size, &err); if (err) GST_ELEMENT_WARNING_FROM_ERROR (self, err); gst_buffer_unmap (codec_data, &minfo); } else if (gst_structure_has_field (s, "streamheader")) { const GValue *sh = gst_structure_get_value (s, "streamheader"); gint nsheaders = gst_value_array_get_size (sh); GstBuffer *buf; const GValue *h; gint i, j; gchar *fname; GstMapInfo minfo; guint8 *data; for (i = 0, j = 0; i < nsheaders; i++) { h = gst_value_array_get_value (sh, i); buf = gst_value_get_buffer (h); if (strcmp (mime, "audio/vorbis") == 0) { guint8 header_type; gst_buffer_extract (buf, 0, &header_type, 1); /* Only use the identification and setup packets */ if (header_type != 0x01 && header_type != 0x05) continue; } fname = g_strdup_printf ("csd-%d", j); gst_buffer_map (buf, &minfo, GST_MAP_READ); data = g_memdup (minfo.data, minfo.size); self->codec_datas = g_list_prepend (self->codec_datas, data); gst_amc_format_set_buffer (format, fname, data, minfo.size, &err); if (err) GST_ELEMENT_WARNING_FROM_ERROR (self, err); gst_buffer_unmap (buf, &minfo); g_free (fname); j++; } } format_string = gst_amc_format_to_string (format, &err); if (err) GST_ELEMENT_WARNING_FROM_ERROR (self, err); GST_DEBUG_OBJECT (self, "Configuring codec with format: %s", GST_STR_NULL (format_string)); g_free (format_string); if (!gst_amc_codec_configure (self->codec, format, NULL, 0, &err)) { GST_ERROR_OBJECT (self, "Failed to configure codec"); GST_ELEMENT_ERROR_FROM_ERROR (self, err); return FALSE; } gst_amc_format_free (format); if (!gst_amc_codec_start (self->codec, &err)) { GST_ERROR_OBJECT (self, "Failed to start codec"); GST_ELEMENT_ERROR_FROM_ERROR (self, err); return FALSE; } self->spf = -1; /* TODO: Implement for other codecs too */ if (gst_structure_has_name (s, "audio/mpeg")) { gint mpegversion = -1; gst_structure_get_int (s, "mpegversion", &mpegversion); if (mpegversion == 1) { gint layer = -1, mpegaudioversion = -1; gst_structure_get_int (s, "layer", &layer); gst_structure_get_int (s, "mpegaudioversion", &mpegaudioversion); if (layer == 1) self->spf = 384; else if (layer == 2) self->spf = 1152; else if (layer == 3 && mpegaudioversion != -1) self->spf = (mpegaudioversion == 1 ? 1152 : 576); } } self->started = TRUE; self->input_caps_changed = TRUE; /* Start the srcpad loop again */ self->flushing = FALSE; self->downstream_flow_ret = GST_FLOW_OK; gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self), (GstTaskFunction) gst_amc_audio_dec_loop, decoder, NULL); return TRUE; } static void gst_amc_audio_dec_flush (GstAudioDecoder * decoder, gboolean hard) { GstAmcAudioDec *self; GError *err = NULL; self = GST_AMC_AUDIO_DEC (decoder); GST_DEBUG_OBJECT (self, "Resetting decoder"); if (!self->started) { GST_DEBUG_OBJECT (self, "Codec not started yet"); return; } self->flushing = TRUE; /* Wait until the srcpad loop is finished, * unlock GST_AUDIO_DECODER_STREAM_LOCK to prevent deadlocks * caused by using this lock from inside the loop function */ GST_AUDIO_DECODER_STREAM_UNLOCK (self); GST_PAD_STREAM_LOCK (GST_AUDIO_DECODER_SRC_PAD (self)); GST_PAD_STREAM_UNLOCK (GST_AUDIO_DECODER_SRC_PAD (self)); GST_AUDIO_DECODER_STREAM_LOCK (self); gst_amc_codec_flush (self->codec, &err); if (err) GST_ELEMENT_WARNING_FROM_ERROR (self, err); gst_adapter_flush (self->output_adapter, gst_adapter_available (self->output_adapter)); self->flushing = FALSE; /* Start the srcpad loop again */ self->last_upstream_ts = 0; self->drained = TRUE; self->downstream_flow_ret = GST_FLOW_OK; gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self), (GstTaskFunction) gst_amc_audio_dec_loop, decoder, NULL); GST_DEBUG_OBJECT (self, "Reset decoder"); } static GstFlowReturn gst_amc_audio_dec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf) { GstAmcAudioDec *self; gint idx; GstAmcBuffer *buf; GstAmcBufferInfo buffer_info; guint offset = 0; GstClockTime timestamp, duration, timestamp_offset = 0; GstMapInfo minfo; GError *err = NULL; memset (&minfo, 0, sizeof (minfo)); self = GST_AMC_AUDIO_DEC (decoder); GST_DEBUG_OBJECT (self, "Handling frame"); /* Make sure to keep a reference to the input here, * it can be unreffed from the other thread if * finish_frame() is called */ if (inbuf) inbuf = gst_buffer_ref (inbuf); if (!self->started) { GST_ERROR_OBJECT (self, "Codec not started yet"); if (inbuf) gst_buffer_unref (inbuf); return GST_FLOW_NOT_NEGOTIATED; } if (self->flushing) goto flushing; if (self->downstream_flow_ret != GST_FLOW_OK) goto downstream_error; if (!inbuf) return gst_amc_audio_dec_drain (self); timestamp = GST_BUFFER_PTS (inbuf); duration = GST_BUFFER_DURATION (inbuf); gst_buffer_map (inbuf, &minfo, GST_MAP_READ); while (offset < minfo.size) { /* Make sure to release the base class stream lock, otherwise * _loop() can't call _finish_frame() and we might block forever * because no input buffers are released */ GST_AUDIO_DECODER_STREAM_UNLOCK (self); /* Wait at most 100ms here, some codecs don't fail dequeueing if * the codec is flushing, causing deadlocks during shutdown */ idx = gst_amc_codec_dequeue_input_buffer (self->codec, 100000, &err); GST_AUDIO_DECODER_STREAM_LOCK (self); if (idx < 0) { if (self->flushing || self->downstream_flow_ret == GST_FLOW_FLUSHING) { g_clear_error (&err); goto flushing; } switch (idx) { case INFO_TRY_AGAIN_LATER: GST_DEBUG_OBJECT (self, "Dequeueing input buffer timed out"); continue; /* next try */ break; case G_MININT: GST_ERROR_OBJECT (self, "Failed to dequeue input buffer"); goto dequeue_error; default: g_assert_not_reached (); break; } continue; } if (self->flushing) { memset (&buffer_info, 0, sizeof (buffer_info)); gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info, NULL); goto flushing; } if (self->downstream_flow_ret != GST_FLOW_OK) { memset (&buffer_info, 0, sizeof (buffer_info)); gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info, &err); if (err && !self->flushing) GST_ELEMENT_WARNING_FROM_ERROR (self, err); g_clear_error (&err); goto downstream_error; } /* Now handle the frame */ /* Copy the buffer content in chunks of size as requested * by the port */ buf = gst_amc_codec_get_input_buffer (self->codec, idx, &err); if (err) goto failed_to_get_input_buffer; else if (!buf) goto got_null_input_buffer; memset (&buffer_info, 0, sizeof (buffer_info)); buffer_info.offset = 0; buffer_info.size = MIN (minfo.size - offset, buf->size); gst_amc_buffer_set_position_and_limit (buf, NULL, buffer_info.offset, buffer_info.size); orc_memcpy (buf->data, minfo.data + offset, buffer_info.size); gst_amc_buffer_free (buf); buf = NULL; /* Interpolate timestamps if we're passing the buffer * in multiple chunks */ if (offset != 0 && duration != GST_CLOCK_TIME_NONE) { timestamp_offset = gst_util_uint64_scale (offset, duration, minfo.size); } if (timestamp != GST_CLOCK_TIME_NONE) { buffer_info.presentation_time_us = gst_util_uint64_scale (timestamp + timestamp_offset, 1, GST_USECOND); self->last_upstream_ts = timestamp + timestamp_offset; } if (duration != GST_CLOCK_TIME_NONE) self->last_upstream_ts += duration; if (offset == 0) { if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DELTA_UNIT)) buffer_info.flags |= BUFFER_FLAG_SYNC_FRAME; } offset += buffer_info.size; GST_DEBUG_OBJECT (self, "Queueing buffer %d: size %d time %" G_GINT64_FORMAT " flags 0x%08x", idx, buffer_info.size, buffer_info.presentation_time_us, buffer_info.flags); if (!gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info, &err)) { if (self->flushing) { g_clear_error (&err); goto flushing; } goto queue_error; } self->drained = FALSE; } gst_buffer_unmap (inbuf, &minfo); gst_buffer_unref (inbuf); return self->downstream_flow_ret; downstream_error: { GST_ERROR_OBJECT (self, "Downstream returned %s", gst_flow_get_name (self->downstream_flow_ret)); if (minfo.data) gst_buffer_unmap (inbuf, &minfo); if (inbuf) gst_buffer_unref (inbuf); return self->downstream_flow_ret; } failed_to_get_input_buffer: { GST_ELEMENT_ERROR_FROM_ERROR (self, err); if (minfo.data) gst_buffer_unmap (inbuf, &minfo); if (inbuf) gst_buffer_unref (inbuf); return GST_FLOW_ERROR; } got_null_input_buffer: { GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Got no input buffer")); if (minfo.data) gst_buffer_unmap (inbuf, &minfo); if (inbuf) gst_buffer_unref (inbuf); return GST_FLOW_ERROR; } dequeue_error: { GST_ELEMENT_ERROR_FROM_ERROR (self, err); if (minfo.data) gst_buffer_unmap (inbuf, &minfo); if (inbuf) gst_buffer_unref (inbuf); return GST_FLOW_ERROR; } queue_error: { GST_AUDIO_DECODER_ERROR_FROM_ERROR (self, err); if (minfo.data) gst_buffer_unmap (inbuf, &minfo); if (inbuf) gst_buffer_unref (inbuf); return GST_FLOW_ERROR; } flushing: { GST_DEBUG_OBJECT (self, "Flushing -- returning FLUSHING"); if (minfo.data) gst_buffer_unmap (inbuf, &minfo); if (inbuf) gst_buffer_unref (inbuf); return GST_FLOW_FLUSHING; } } static GstFlowReturn gst_amc_audio_dec_drain (GstAmcAudioDec * self) { GstFlowReturn ret; gint idx; GError *err = NULL; GST_DEBUG_OBJECT (self, "Draining codec"); if (!self->started) { GST_DEBUG_OBJECT (self, "Codec not started yet"); return GST_FLOW_OK; } /* Don't send drain buffer twice, this doesn't work */ if (self->drained) { GST_DEBUG_OBJECT (self, "Codec is drained already"); return GST_FLOW_OK; } /* Make sure to release the base class stream lock, otherwise * _loop() can't call _finish_frame() and we might block forever * because no input buffers are released */ GST_AUDIO_DECODER_STREAM_UNLOCK (self); /* Send an EOS buffer to the component and let the base * class drop the EOS event. We will send it later when * the EOS buffer arrives on the output port. * Wait at most 0.5s here. */ idx = gst_amc_codec_dequeue_input_buffer (self->codec, 500000, &err); GST_AUDIO_DECODER_STREAM_LOCK (self); if (idx >= 0) { GstAmcBuffer *buf; GstAmcBufferInfo buffer_info; buf = gst_amc_codec_get_input_buffer (self->codec, idx, &err); if (buf) { GST_AUDIO_DECODER_STREAM_UNLOCK (self); g_mutex_lock (&self->drain_lock); self->draining = TRUE; memset (&buffer_info, 0, sizeof (buffer_info)); buffer_info.size = 0; buffer_info.presentation_time_us = gst_util_uint64_scale (self->last_upstream_ts, 1, GST_USECOND); buffer_info.flags |= BUFFER_FLAG_END_OF_STREAM; gst_amc_buffer_set_position_and_limit (buf, NULL, 0, 0); gst_amc_buffer_free (buf); buf = NULL; if (gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info, &err)) { GST_DEBUG_OBJECT (self, "Waiting until codec is drained"); g_cond_wait (&self->drain_cond, &self->drain_lock); GST_DEBUG_OBJECT (self, "Drained codec"); ret = GST_FLOW_OK; } else { GST_ERROR_OBJECT (self, "Failed to queue input buffer"); if (self->flushing) { g_clear_error (&err); ret = GST_FLOW_FLUSHING; } else { GST_ELEMENT_WARNING_FROM_ERROR (self, err); ret = GST_FLOW_ERROR; } } self->drained = TRUE; self->draining = FALSE; g_mutex_unlock (&self->drain_lock); GST_AUDIO_DECODER_STREAM_LOCK (self); } else { GST_ERROR_OBJECT (self, "Failed to get buffer for EOS: %d", idx); if (err) GST_ELEMENT_WARNING_FROM_ERROR (self, err); ret = GST_FLOW_ERROR; } } else { GST_ERROR_OBJECT (self, "Failed to acquire buffer for EOS: %d", idx); if (err) GST_ELEMENT_WARNING_FROM_ERROR (self, err); ret = GST_FLOW_ERROR; } gst_adapter_flush (self->output_adapter, gst_adapter_available (self->output_adapter)); return ret; }