| /* |
| * 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 (!buf) |
| goto failed_to_get_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; |
| } |
| 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 (!buf) |
| goto failed_to_get_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; |
| } |
| 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; |
| } |