/* * Copyright (C) 2011, Hewlett-Packard Development Company, L.P. * Author: Sebastian Dröge , Collabora Ltd. * Copyright (C) 2013, Collabora Ltd. * Author: Sebastian Dröge * Copyright (C) 2014, Sebastian Dröge * * 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 #include #include "gstomxaudiodec.h" GST_DEBUG_CATEGORY_STATIC (gst_omx_audio_dec_debug_category); #define GST_CAT_DEFAULT gst_omx_audio_dec_debug_category /* prototypes */ static void gst_omx_audio_dec_finalize (GObject * object); static GstStateChangeReturn gst_omx_audio_dec_change_state (GstElement * element, GstStateChange transition); static gboolean gst_omx_audio_dec_open (GstAudioDecoder * decoder); static gboolean gst_omx_audio_dec_close (GstAudioDecoder * decoder); static gboolean gst_omx_audio_dec_start (GstAudioDecoder * decoder); static gboolean gst_omx_audio_dec_stop (GstAudioDecoder * decoder); static gboolean gst_omx_audio_dec_set_format (GstAudioDecoder * decoder, GstCaps * caps); static void gst_omx_audio_dec_flush (GstAudioDecoder * decoder, gboolean hard); static GstFlowReturn gst_omx_audio_dec_handle_frame (GstAudioDecoder * decoder, GstBuffer * buffer); static GstFlowReturn gst_omx_audio_dec_drain (GstOMXAudioDec * self); enum { PROP_0 }; /* class initialization */ #define DEBUG_INIT \ GST_DEBUG_CATEGORY_INIT (gst_omx_audio_dec_debug_category, "omxaudiodec", 0, \ "debug category for gst-omx audio decoder base class"); G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstOMXAudioDec, gst_omx_audio_dec, GST_TYPE_AUDIO_DECODER, DEBUG_INIT); static void gst_omx_audio_dec_class_init (GstOMXAudioDecClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstAudioDecoderClass *audio_decoder_class = GST_AUDIO_DECODER_CLASS (klass); gobject_class->finalize = gst_omx_audio_dec_finalize; element_class->change_state = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_change_state); audio_decoder_class->open = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_open); audio_decoder_class->close = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_close); audio_decoder_class->start = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_start); audio_decoder_class->stop = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_stop); audio_decoder_class->flush = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_flush); audio_decoder_class->set_format = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_set_format); audio_decoder_class->handle_frame = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_handle_frame); klass->cdata.type = GST_OMX_COMPONENT_TYPE_FILTER; klass->cdata.default_src_template_caps = "audio/x-raw, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, " G_STRINGIFY (OMX_AUDIO_MAXCHANNELS) " ], " "format = (string) " GST_AUDIO_FORMATS_ALL; } static void gst_omx_audio_dec_init (GstOMXAudioDec * self) { gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (self), TRUE); gst_audio_decoder_set_drainable (GST_AUDIO_DECODER (self), TRUE); gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST (self), TRUE); GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (self)); g_mutex_init (&self->drain_lock); g_cond_init (&self->drain_cond); self->output_adapter = gst_adapter_new (); } static gboolean gst_omx_audio_dec_open (GstAudioDecoder * decoder) { GstOMXAudioDec *self = GST_OMX_AUDIO_DEC (decoder); GstOMXAudioDecClass *klass = GST_OMX_AUDIO_DEC_GET_CLASS (self); gint in_port_index, out_port_index; GST_DEBUG_OBJECT (self, "Opening decoder"); self->dec = gst_omx_component_new (GST_OBJECT_CAST (self), klass->cdata.core_name, klass->cdata.component_name, klass->cdata.component_role, klass->cdata.hacks); self->started = FALSE; if (!self->dec) return FALSE; if (gst_omx_component_get_state (self->dec, GST_CLOCK_TIME_NONE) != OMX_StateLoaded) return FALSE; in_port_index = klass->cdata.in_port_index; out_port_index = klass->cdata.out_port_index; if (in_port_index == -1 || out_port_index == -1) { OMX_PORT_PARAM_TYPE param; OMX_ERRORTYPE err; GST_OMX_INIT_STRUCT (¶m); err = gst_omx_component_get_parameter (self->dec, OMX_IndexParamAudioInit, ¶m); if (err != OMX_ErrorNone) { GST_WARNING_OBJECT (self, "Couldn't get port information: %s (0x%08x)", gst_omx_error_to_string (err), err); /* Fallback */ in_port_index = 0; out_port_index = 1; } else { GST_DEBUG_OBJECT (self, "Detected %u ports, starting at %u", (guint) param.nPorts, (guint) param.nStartPortNumber); in_port_index = param.nStartPortNumber + 0; out_port_index = param.nStartPortNumber + 1; } } self->dec_in_port = gst_omx_component_add_port (self->dec, in_port_index); self->dec_out_port = gst_omx_component_add_port (self->dec, out_port_index); if (!self->dec_in_port || !self->dec_out_port) return FALSE; GST_DEBUG_OBJECT (self, "Opened decoder"); return TRUE; } static gboolean gst_omx_audio_dec_shutdown (GstOMXAudioDec * self) { OMX_STATETYPE state; GST_DEBUG_OBJECT (self, "Shutting down decoder"); state = gst_omx_component_get_state (self->dec, 0); if (state > OMX_StateLoaded || state == OMX_StateInvalid) { if (state > OMX_StateIdle) { gst_omx_component_set_state (self->dec, OMX_StateIdle); gst_omx_component_get_state (self->dec, 5 * GST_SECOND); } gst_omx_component_set_state (self->dec, OMX_StateLoaded); gst_omx_port_deallocate_buffers (self->dec_in_port); gst_omx_port_deallocate_buffers (self->dec_out_port); if (state > OMX_StateLoaded) gst_omx_component_get_state (self->dec, 5 * GST_SECOND); } return TRUE; } static gboolean gst_omx_audio_dec_close (GstAudioDecoder * decoder) { GstOMXAudioDec *self = GST_OMX_AUDIO_DEC (decoder); GST_DEBUG_OBJECT (self, "Closing decoder"); if (!gst_omx_audio_dec_shutdown (self)) return FALSE; self->dec_in_port = NULL; self->dec_out_port = NULL; if (self->dec) gst_omx_component_free (self->dec); self->dec = NULL; self->started = FALSE; GST_DEBUG_OBJECT (self, "Closed decoder"); return TRUE; } static void gst_omx_audio_dec_finalize (GObject * object) { GstOMXAudioDec *self = GST_OMX_AUDIO_DEC (object); g_mutex_clear (&self->drain_lock); g_cond_clear (&self->drain_cond); if (self->output_adapter) gst_object_unref (self->output_adapter); self->output_adapter = NULL; G_OBJECT_CLASS (gst_omx_audio_dec_parent_class)->finalize (object); } static GstStateChangeReturn gst_omx_audio_dec_change_state (GstElement * element, GstStateChange transition) { GstOMXAudioDec *self; GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; g_return_val_if_fail (GST_IS_OMX_AUDIO_DEC (element), GST_STATE_CHANGE_FAILURE); self = GST_OMX_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: if (self->dec_in_port) gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, TRUE); if (self->dec_out_port) gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, TRUE); g_mutex_lock (&self->drain_lock); self->draining = FALSE; g_cond_broadcast (&self->drain_cond); g_mutex_unlock (&self->drain_lock); break; default: break; } ret = GST_ELEMENT_CLASS (gst_omx_audio_dec_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; if (!gst_omx_audio_dec_shutdown (self)) ret = GST_STATE_CHANGE_FAILURE; break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } return ret; } static void gst_omx_audio_dec_loop (GstOMXAudioDec * self) { GstOMXAudioDecClass *klass = GST_OMX_AUDIO_DEC_GET_CLASS (self); GstOMXPort *port = self->dec_out_port; GstOMXBuffer *buf = NULL; GstFlowReturn flow_ret = GST_FLOW_OK; GstOMXAcquireBufferReturn acq_return; OMX_ERRORTYPE err; gint spf; acq_return = gst_omx_port_acquire_buffer (port, &buf); if (acq_return == GST_OMX_ACQUIRE_BUFFER_ERROR) { goto component_error; } else if (acq_return == GST_OMX_ACQUIRE_BUFFER_FLUSHING) { goto flushing; } else if (acq_return == GST_OMX_ACQUIRE_BUFFER_EOS) { goto eos; } if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (self)) || acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) { OMX_PARAM_PORTDEFINITIONTYPE port_def; OMX_AUDIO_PARAM_PCMMODETYPE pcm_param; GstAudioChannelPosition omx_position[OMX_AUDIO_MAXCHANNELS]; GstOMXAudioDecClass *klass = GST_OMX_AUDIO_DEC_GET_CLASS (self); gint i; GST_DEBUG_OBJECT (self, "Port settings have changed, updating caps"); /* Reallocate all buffers */ if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE && gst_omx_port_is_enabled (port)) { err = gst_omx_port_set_enabled (port, FALSE); if (err != OMX_ErrorNone) goto reconfigure_error; err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND); if (err != OMX_ErrorNone) goto reconfigure_error; err = gst_omx_port_deallocate_buffers (port); if (err != OMX_ErrorNone) goto reconfigure_error; err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND); if (err != OMX_ErrorNone) goto reconfigure_error; } /* Just update caps */ GST_AUDIO_DECODER_STREAM_LOCK (self); gst_omx_port_get_port_definition (port, &port_def); g_assert (port_def.format.audio.eEncoding == OMX_AUDIO_CodingPCM); GST_OMX_INIT_STRUCT (&pcm_param); pcm_param.nPortIndex = self->dec_out_port->index; err = gst_omx_component_get_parameter (self->dec, OMX_IndexParamAudioPcm, &pcm_param); if (err != OMX_ErrorNone) { GST_ERROR_OBJECT (self, "Failed to get PCM parameters: %s (0x%08x)", gst_omx_error_to_string (err), err); goto caps_failed; } g_assert (pcm_param.ePCMMode == OMX_AUDIO_PCMModeLinear); g_assert (pcm_param.bInterleaved == OMX_TRUE); gst_audio_info_init (&self->info); for (i = 0; i < pcm_param.nChannels; i++) { switch (pcm_param.eChannelMapping[i]) { case OMX_AUDIO_ChannelLF: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; break; case OMX_AUDIO_ChannelRF: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; break; case OMX_AUDIO_ChannelCF: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER; break; case OMX_AUDIO_ChannelLS: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT; break; case OMX_AUDIO_ChannelRS: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT; break; case OMX_AUDIO_ChannelLFE: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_LFE1; break; case OMX_AUDIO_ChannelCS: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER; break; case OMX_AUDIO_ChannelLR: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT; break; case OMX_AUDIO_ChannelRR: omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT; break; case OMX_AUDIO_ChannelNone: default: /* This will break the outer loop too as the * i == pcm_param.nChannels afterwards */ for (i = 0; i < pcm_param.nChannels; i++) omx_position[i] = GST_AUDIO_CHANNEL_POSITION_NONE; break; } } if (pcm_param.nChannels == 1 && omx_position[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER) omx_position[0] = GST_AUDIO_CHANNEL_POSITION_MONO; if (omx_position[0] == GST_AUDIO_CHANNEL_POSITION_NONE && klass->get_channel_positions) { GST_WARNING_OBJECT (self, "Failed to get a valid channel layout, trying fallback"); klass->get_channel_positions (self, self->dec_out_port, omx_position); } memcpy (self->position, omx_position, sizeof (omx_position)); gst_audio_channel_positions_to_valid_order (self->position, pcm_param.nChannels); self->needs_reorder = (memcmp (self->position, omx_position, sizeof (GstAudioChannelPosition) * pcm_param.nChannels) != 0); if (self->needs_reorder) gst_audio_get_channel_reorder_map (pcm_param.nChannels, self->position, omx_position, self->reorder_map); gst_audio_info_set_format (&self->info, gst_audio_format_build_integer (pcm_param.eNumData == OMX_NumericalDataSigned, pcm_param.eEndian == OMX_EndianLittle ? G_LITTLE_ENDIAN : G_BIG_ENDIAN, pcm_param.nBitPerSample, pcm_param.nBitPerSample), pcm_param.nSamplingRate, pcm_param.nChannels, self->position); GST_DEBUG_OBJECT (self, "Setting output state: format %s, rate %u, channels %u", gst_audio_format_to_string (self->info.finfo->format), (guint) pcm_param.nSamplingRate, (guint) pcm_param.nChannels); if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (self), &self->info) || !gst_audio_decoder_negotiate (GST_AUDIO_DECODER (self))) { if (buf) gst_omx_port_release_buffer (port, buf); goto caps_failed; } GST_AUDIO_DECODER_STREAM_UNLOCK (self); if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) { err = gst_omx_port_set_enabled (port, TRUE); if (err != OMX_ErrorNone) goto reconfigure_error; err = gst_omx_port_allocate_buffers (port); if (err != OMX_ErrorNone) goto reconfigure_error; err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND); if (err != OMX_ErrorNone) goto reconfigure_error; err = gst_omx_port_populate (port); if (err != OMX_ErrorNone) goto reconfigure_error; err = gst_omx_port_mark_reconfigured (port); if (err != OMX_ErrorNone) goto reconfigure_error; } /* Now get a buffer */ if (acq_return != GST_OMX_ACQUIRE_BUFFER_OK) { return; } } g_assert (acq_return == GST_OMX_ACQUIRE_BUFFER_OK); if (!buf) { g_assert ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER)); GST_AUDIO_DECODER_STREAM_LOCK (self); goto eos; } /* This prevents a deadlock between the srcpad stream * lock and the audiocodec stream lock, if ::reset() * is called at the wrong time */ if (gst_omx_port_is_flushing (port)) { GST_DEBUG_OBJECT (self, "Flushing"); gst_omx_port_release_buffer (port, buf); goto flushing; } GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %" G_GUINT64_FORMAT, (guint) buf->omx_buf->nFlags, (guint64) GST_OMX_GET_TICKS (buf->omx_buf->nTimeStamp)); GST_AUDIO_DECODER_STREAM_LOCK (self); spf = klass->get_samples_per_frame (self, self->dec_out_port); if (buf->omx_buf->nFilledLen > 0) { GstBuffer *outbuf; GstMapInfo minfo; GST_DEBUG_OBJECT (self, "Handling output data"); if (buf->omx_buf->nFilledLen % self->info.bpf != 0) { gst_omx_port_release_buffer (port, buf); goto invalid_buffer; } outbuf = gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (self), buf->omx_buf->nFilledLen); 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->omx_buf->pBuffer + buf->omx_buf->nOffset); n_samples = buf->omx_buf->nFilledLen / 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 { memcpy (minfo.data, buf->omx_buf->pBuffer + buf->omx_buf->nOffset, buf->omx_buf->nFilledLen); } gst_buffer_unmap (outbuf, &minfo); if (spf != -1) { gst_adapter_push (self->output_adapter, outbuf); } else { flow_ret = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf, 1); } } GST_DEBUG_OBJECT (self, "Read frame from component"); if (spf != -1) { GstBuffer *outbuf; guint avail = gst_adapter_available (self->output_adapter); guint nframes; /* We take a multiple of codec frames and push * them downstream */ avail /= self->info.bpf; nframes = avail / spf; avail = nframes * 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); } } GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret)); if (buf) { err = gst_omx_port_release_buffer (port, buf); if (err != OMX_ErrorNone) goto release_error; } self->downstream_flow_ret = flow_ret; if (flow_ret != GST_FLOW_OK) goto flow_error; GST_AUDIO_DECODER_STREAM_UNLOCK (self); return; component_error: { GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL), ("OpenMAX component in error state %s (0x%08x)", gst_omx_component_get_last_error_string (self->dec), gst_omx_component_get_last_error (self->dec))); 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; self->started = FALSE; return; } flushing: { GST_DEBUG_OBJECT (self, "Flushing -- stopping task"); g_mutex_lock (&self->drain_lock); if (self->draining) { self->draining = FALSE; g_cond_broadcast (&self->drain_cond); } gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_FLUSHING; self->started = FALSE; g_mutex_unlock (&self->drain_lock); return; } eos: { spf = klass->get_samples_per_frame (self, self->dec_out_port); if (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. */ avail /= self->info.bpf; nframes = (avail + spf - 1) / 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); } } g_mutex_lock (&self->drain_lock); if (self->draining) { GST_DEBUG_OBJECT (self, "Drained"); self->draining = FALSE; g_cond_broadcast (&self->drain_cond); flow_ret = GST_FLOW_OK; gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); } else { GST_DEBUG_OBJECT (self, "Component signalled EOS"); flow_ret = GST_FLOW_EOS; } g_mutex_unlock (&self->drain_lock); GST_AUDIO_DECODER_STREAM_LOCK (self); self->downstream_flow_ret = flow_ret; /* Here we fallback and pause the task for the EOS case */ if (flow_ret != GST_FLOW_OK) goto flow_error; 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)); self->started = FALSE; } 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)); self->started = FALSE; } else if (flow_ret == GST_FLOW_FLUSHING) { GST_DEBUG_OBJECT (self, "Flushing -- stopping task"); g_mutex_lock (&self->drain_lock); if (self->draining) { self->draining = FALSE; g_cond_broadcast (&self->drain_cond); } gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); self->started = FALSE; g_mutex_unlock (&self->drain_lock); } GST_AUDIO_DECODER_STREAM_UNLOCK (self); return; } reconfigure_error: { GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Unable to reconfigure output port")); 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; self->started = FALSE; return; } invalid_buffer: { GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Invalid sized input 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_NOT_NEGOTIATED; self->started = FALSE; GST_AUDIO_DECODER_STREAM_UNLOCK (self); return; } caps_failed: { GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to set caps")); gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self)); GST_AUDIO_DECODER_STREAM_UNLOCK (self); self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED; self->started = FALSE; return; } release_error: { GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to relase output buffer to component: %s (0x%08x)", gst_omx_error_to_string (err), 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; self->started = FALSE; GST_AUDIO_DECODER_STREAM_UNLOCK (self); return; } } static gboolean gst_omx_audio_dec_start (GstAudioDecoder * decoder) { GstOMXAudioDec *self; self = GST_OMX_AUDIO_DEC (decoder); self->last_upstream_ts = 0; self->downstream_flow_ret = GST_FLOW_OK; return TRUE; } static gboolean gst_omx_audio_dec_stop (GstAudioDecoder * decoder) { GstOMXAudioDec *self; self = GST_OMX_AUDIO_DEC (decoder); GST_DEBUG_OBJECT (self, "Stopping decoder"); gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, TRUE); gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, TRUE); gst_pad_stop_task (GST_AUDIO_DECODER_SRC_PAD (decoder)); if (gst_omx_component_get_state (self->dec, 0) > OMX_StateIdle) gst_omx_component_set_state (self->dec, OMX_StateIdle); self->downstream_flow_ret = GST_FLOW_FLUSHING; self->started = FALSE; g_mutex_lock (&self->drain_lock); self->draining = FALSE; g_cond_broadcast (&self->drain_cond); g_mutex_unlock (&self->drain_lock); gst_adapter_flush (self->output_adapter, gst_adapter_available (self->output_adapter)); gst_omx_component_get_state (self->dec, 5 * GST_SECOND); gst_buffer_replace (&self->codec_data, NULL); GST_DEBUG_OBJECT (self, "Stopped decoder"); return TRUE; } static gboolean gst_omx_audio_dec_set_format (GstAudioDecoder * decoder, GstCaps * caps) { GstOMXAudioDec *self; GstOMXAudioDecClass *klass; GstStructure *s; const GValue *codec_data; gboolean is_format_change = FALSE; gboolean needs_disable = FALSE; self = GST_OMX_AUDIO_DEC (decoder); klass = GST_OMX_AUDIO_DEC_GET_CLASS (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. */ if (klass->is_format_change) is_format_change = klass->is_format_change (self, self->dec_in_port, caps); needs_disable = gst_omx_component_get_state (self->dec, GST_CLOCK_TIME_NONE) != OMX_StateLoaded; /* If the component is not in Loaded state and a real format change happens * we have to disable the port and re-allocate all buffers. If no real * format change happened we can just exit here. */ if (needs_disable && !is_format_change) { GST_DEBUG_OBJECT (self, "Already running and caps did not change the format"); return TRUE; } if (needs_disable && is_format_change) { GstOMXPort *out_port = self->dec_out_port; GST_DEBUG_OBJECT (self, "Need to disable and drain decoder"); gst_omx_audio_dec_drain (self); gst_omx_audio_dec_flush (decoder, FALSE); gst_omx_port_set_flushing (out_port, 5 * GST_SECOND, TRUE); if (klass->cdata.hacks & GST_OMX_HACK_NO_COMPONENT_RECONFIGURE) { GST_AUDIO_DECODER_STREAM_UNLOCK (self); gst_omx_audio_dec_stop (GST_AUDIO_DECODER (self)); gst_omx_audio_dec_close (GST_AUDIO_DECODER (self)); GST_AUDIO_DECODER_STREAM_LOCK (self); if (!gst_omx_audio_dec_open (GST_AUDIO_DECODER (self))) return FALSE; needs_disable = FALSE; } else { /* Disabling at the same time input port and output port is only * required when a buffer is shared between the ports. This cannot * be the case for a decoder because its input and output buffers * are of different nature. So let's disable ports sequencially. * Starting from IL 1.2.0, this point has been clarified. * OMX_SendCommand will return an error if the IL client attempts to * call it when there is already an on-going command being processed. * The exception is for buffer sharing above and the event * OMX_EventPortNeedsDisable will be sent to request disabling the * other port at the same time. */ if (gst_omx_port_set_enabled (self->dec_in_port, FALSE) != OMX_ErrorNone) return FALSE; if (gst_omx_port_wait_buffers_released (self->dec_in_port, 5 * GST_SECOND) != OMX_ErrorNone) return FALSE; if (gst_omx_port_deallocate_buffers (self->dec_in_port) != OMX_ErrorNone) return FALSE; if (gst_omx_port_wait_enabled (self->dec_in_port, 1 * GST_SECOND) != OMX_ErrorNone) return FALSE; if (gst_omx_port_set_enabled (out_port, FALSE) != OMX_ErrorNone) return FALSE; if (gst_omx_port_wait_buffers_released (out_port, 1 * GST_SECOND) != OMX_ErrorNone) return FALSE; if (gst_omx_port_deallocate_buffers (out_port) != OMX_ErrorNone) return FALSE; if (gst_omx_port_wait_enabled (out_port, 1 * GST_SECOND) != OMX_ErrorNone) return FALSE; } GST_DEBUG_OBJECT (self, "Decoder drained and disabled"); } if (klass->set_format) { if (!klass->set_format (self, self->dec_in_port, caps)) { GST_ERROR_OBJECT (self, "Subclass failed to set the new format"); return FALSE; } } GST_DEBUG_OBJECT (self, "Updating outport port definition"); if (gst_omx_port_update_port_definition (self->dec_out_port, NULL) != OMX_ErrorNone) return FALSE; /* Get codec data from caps */ gst_buffer_replace (&self->codec_data, NULL); s = gst_caps_get_structure (caps, 0); codec_data = gst_structure_get_value (s, "codec_data"); if (codec_data) { /* Vorbis and some other codecs have multiple buffers in * the stream-header field */ self->codec_data = gst_value_get_buffer (codec_data); if (self->codec_data) gst_buffer_ref (self->codec_data); } GST_DEBUG_OBJECT (self, "Enabling component"); if (needs_disable) { if (gst_omx_port_set_enabled (self->dec_in_port, TRUE) != OMX_ErrorNone) return FALSE; if (gst_omx_port_allocate_buffers (self->dec_in_port) != OMX_ErrorNone) return FALSE; if ((klass->cdata.hacks & GST_OMX_HACK_NO_DISABLE_OUTPORT)) { if (gst_omx_port_set_enabled (self->dec_out_port, TRUE) != OMX_ErrorNone) return FALSE; if (gst_omx_port_allocate_buffers (self->dec_out_port) != OMX_ErrorNone) return FALSE; if (gst_omx_port_wait_enabled (self->dec_out_port, 5 * GST_SECOND) != OMX_ErrorNone) return FALSE; } if (gst_omx_port_wait_enabled (self->dec_in_port, 5 * GST_SECOND) != OMX_ErrorNone) return FALSE; if (gst_omx_port_mark_reconfigured (self->dec_in_port) != OMX_ErrorNone) return FALSE; } else { if (!(klass->cdata.hacks & GST_OMX_HACK_NO_DISABLE_OUTPORT)) { /* Disable output port */ if (gst_omx_port_set_enabled (self->dec_out_port, FALSE) != OMX_ErrorNone) return FALSE; if (gst_omx_port_wait_enabled (self->dec_out_port, 1 * GST_SECOND) != OMX_ErrorNone) return FALSE; if (gst_omx_component_set_state (self->dec, OMX_StateIdle) != OMX_ErrorNone) return FALSE; /* Need to allocate buffers to reach Idle state */ if (gst_omx_port_allocate_buffers (self->dec_in_port) != OMX_ErrorNone) return FALSE; } else { if (gst_omx_component_set_state (self->dec, OMX_StateIdle) != OMX_ErrorNone) return FALSE; /* Need to allocate buffers to reach Idle state */ if (gst_omx_port_allocate_buffers (self->dec_in_port) != OMX_ErrorNone) return FALSE; if (gst_omx_port_allocate_buffers (self->dec_out_port) != OMX_ErrorNone) return FALSE; } if (gst_omx_component_get_state (self->dec, GST_CLOCK_TIME_NONE) != OMX_StateIdle) return FALSE; if (gst_omx_component_set_state (self->dec, OMX_StateExecuting) != OMX_ErrorNone) return FALSE; if (gst_omx_component_get_state (self->dec, GST_CLOCK_TIME_NONE) != OMX_StateExecuting) return FALSE; } /* Unset flushing to allow ports to accept data again */ gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, FALSE); gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, FALSE); if (gst_omx_component_get_last_error (self->dec) != OMX_ErrorNone) { GST_ERROR_OBJECT (self, "Component in error state: %s (0x%08x)", gst_omx_component_get_last_error_string (self->dec), gst_omx_component_get_last_error (self->dec)); return FALSE; } self->downstream_flow_ret = GST_FLOW_OK; return TRUE; } static void gst_omx_audio_dec_flush (GstAudioDecoder * decoder, gboolean hard) { GstOMXAudioDec *self = GST_OMX_AUDIO_DEC (decoder); OMX_ERRORTYPE err = OMX_ErrorNone; GST_DEBUG_OBJECT (self, "Flushing decoder"); if (gst_omx_component_get_state (self->dec, 0) == OMX_StateLoaded) return; /* 0) Pause the components */ if (gst_omx_component_get_state (self->dec, 0) == OMX_StateExecuting) { gst_omx_component_set_state (self->dec, OMX_StatePause); gst_omx_component_get_state (self->dec, GST_CLOCK_TIME_NONE); } /* 1) Flush the ports */ GST_DEBUG_OBJECT (self, "flushing ports"); gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, TRUE); gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, TRUE); /* 2) Wait until the srcpad loop is stopped, * 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_stop_task (GST_AUDIO_DECODER_SRC_PAD (decoder)); GST_DEBUG_OBJECT (self, "Flushing -- task stopped"); GST_AUDIO_DECODER_STREAM_LOCK (self); /* 3) Resume components */ gst_omx_component_set_state (self->dec, OMX_StateExecuting); gst_omx_component_get_state (self->dec, GST_CLOCK_TIME_NONE); /* 4) Unset flushing to allow ports to accept data again */ gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, FALSE); gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, FALSE); err = gst_omx_port_populate (self->dec_out_port); if (err != OMX_ErrorNone) { GST_WARNING_OBJECT (self, "Failed to populate output port: %s (0x%08x)", gst_omx_error_to_string (err), err); } /* Reset our state */ gst_adapter_flush (self->output_adapter, gst_adapter_available (self->output_adapter)); self->last_upstream_ts = 0; self->downstream_flow_ret = GST_FLOW_OK; self->started = FALSE; GST_DEBUG_OBJECT (self, "Flush finished"); } static GstFlowReturn gst_omx_audio_dec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf) { GstOMXAcquireBufferReturn acq_ret = GST_OMX_ACQUIRE_BUFFER_ERROR; GstOMXAudioDec *self; GstOMXPort *port; GstOMXBuffer *buf; GstBuffer *codec_data = NULL; guint offset = 0; GstClockTime timestamp, duration; OMX_ERRORTYPE err; GstMapInfo minfo; self = GST_OMX_AUDIO_DEC (decoder); GST_DEBUG_OBJECT (self, "Handling frame"); if (self->downstream_flow_ret != GST_FLOW_OK) { return self->downstream_flow_ret; } if (!self->started) { GST_DEBUG_OBJECT (self, "Starting task"); gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self), (GstTaskFunction) gst_omx_audio_dec_loop, decoder, NULL); } if (inbuf == NULL) return gst_omx_audio_dec_drain (self); /* Make sure to keep a reference to the input here, * it can be unreffed from the other thread if * finish_frame() is called */ gst_buffer_ref (inbuf); timestamp = GST_BUFFER_TIMESTAMP (inbuf); duration = GST_BUFFER_DURATION (inbuf); port = self->dec_in_port; 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); acq_ret = gst_omx_port_acquire_buffer (port, &buf); if (acq_ret == GST_OMX_ACQUIRE_BUFFER_ERROR) { GST_AUDIO_DECODER_STREAM_LOCK (self); goto component_error; } else if (acq_ret == GST_OMX_ACQUIRE_BUFFER_FLUSHING) { GST_AUDIO_DECODER_STREAM_LOCK (self); goto flushing; } else if (acq_ret == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) { /* Reallocate all buffers */ err = gst_omx_port_set_enabled (port, FALSE); if (err != OMX_ErrorNone) { GST_AUDIO_DECODER_STREAM_LOCK (self); goto reconfigure_error; } err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND); if (err != OMX_ErrorNone) { GST_AUDIO_DECODER_STREAM_LOCK (self); goto reconfigure_error; } err = gst_omx_port_deallocate_buffers (port); if (err != OMX_ErrorNone) { GST_AUDIO_DECODER_STREAM_LOCK (self); goto reconfigure_error; } err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND); if (err != OMX_ErrorNone) { GST_AUDIO_DECODER_STREAM_LOCK (self); goto reconfigure_error; } err = gst_omx_port_set_enabled (port, TRUE); if (err != OMX_ErrorNone) { GST_AUDIO_DECODER_STREAM_LOCK (self); goto reconfigure_error; } err = gst_omx_port_allocate_buffers (port); if (err != OMX_ErrorNone) { GST_AUDIO_DECODER_STREAM_LOCK (self); goto reconfigure_error; } err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND); if (err != OMX_ErrorNone) { GST_AUDIO_DECODER_STREAM_LOCK (self); goto reconfigure_error; } err = gst_omx_port_mark_reconfigured (port); if (err != OMX_ErrorNone) { GST_AUDIO_DECODER_STREAM_LOCK (self); goto reconfigure_error; } /* Now get a new buffer and fill it */ GST_AUDIO_DECODER_STREAM_LOCK (self); continue; } GST_AUDIO_DECODER_STREAM_LOCK (self); g_assert (acq_ret == GST_OMX_ACQUIRE_BUFFER_OK && buf != NULL); if (buf->omx_buf->nAllocLen - buf->omx_buf->nOffset <= 0) { gst_omx_port_release_buffer (port, buf); goto full_buffer; } if (self->downstream_flow_ret != GST_FLOW_OK) { gst_omx_port_release_buffer (port, buf); goto flow_error; } if (self->codec_data) { GST_DEBUG_OBJECT (self, "Passing codec data to the component"); codec_data = self->codec_data; if (buf->omx_buf->nAllocLen - buf->omx_buf->nOffset < gst_buffer_get_size (codec_data)) { gst_omx_port_release_buffer (port, buf); goto too_large_codec_data; } buf->omx_buf->nFlags |= OMX_BUFFERFLAG_CODECCONFIG; buf->omx_buf->nFlags |= OMX_BUFFERFLAG_ENDOFFRAME; buf->omx_buf->nFilledLen = gst_buffer_get_size (codec_data); gst_buffer_extract (codec_data, 0, buf->omx_buf->pBuffer + buf->omx_buf->nOffset, buf->omx_buf->nFilledLen); if (GST_CLOCK_TIME_IS_VALID (timestamp)) GST_OMX_SET_TICKS (buf->omx_buf->nTimeStamp, gst_util_uint64_scale (timestamp, OMX_TICKS_PER_SECOND, GST_SECOND)); else GST_OMX_SET_TICKS (buf->omx_buf->nTimeStamp, G_GUINT64_CONSTANT (0)); buf->omx_buf->nTickCount = 0; self->started = TRUE; err = gst_omx_port_release_buffer (port, buf); gst_buffer_replace (&self->codec_data, NULL); if (err != OMX_ErrorNone) goto release_error; /* Acquire new buffer for the actual frame */ continue; } /* Now handle the frame */ GST_DEBUG_OBJECT (self, "Passing frame offset %d to the component", offset); /* Copy the buffer content in chunks of size as requested * by the port */ buf->omx_buf->nFilledLen = MIN (minfo.size - offset, buf->omx_buf->nAllocLen - buf->omx_buf->nOffset); gst_buffer_extract (inbuf, offset, buf->omx_buf->pBuffer + buf->omx_buf->nOffset, buf->omx_buf->nFilledLen); if (timestamp != GST_CLOCK_TIME_NONE) { GST_OMX_SET_TICKS (buf->omx_buf->nTimeStamp, gst_util_uint64_scale (timestamp, OMX_TICKS_PER_SECOND, GST_SECOND)); self->last_upstream_ts = timestamp; } else { GST_OMX_SET_TICKS (buf->omx_buf->nTimeStamp, G_GUINT64_CONSTANT (0)); } if (duration != GST_CLOCK_TIME_NONE && offset == 0) { buf->omx_buf->nTickCount = gst_util_uint64_scale (duration, OMX_TICKS_PER_SECOND, GST_SECOND); self->last_upstream_ts += duration; } else { buf->omx_buf->nTickCount = 0; } if (offset == 0) buf->omx_buf->nFlags |= OMX_BUFFERFLAG_SYNCFRAME; /* TODO: Set flags * - OMX_BUFFERFLAG_DECODEONLY for buffers that are outside * the segment */ offset += buf->omx_buf->nFilledLen; if (offset == minfo.size) buf->omx_buf->nFlags |= OMX_BUFFERFLAG_ENDOFFRAME; self->started = TRUE; err = gst_omx_port_release_buffer (port, buf); if (err != OMX_ErrorNone) goto release_error; } gst_buffer_unmap (inbuf, &minfo); gst_buffer_unref (inbuf); GST_DEBUG_OBJECT (self, "Passed frame to component"); return self->downstream_flow_ret; full_buffer: { gst_buffer_unmap (inbuf, &minfo); gst_buffer_unref (inbuf); GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL), ("Got OpenMAX buffer with no free space (%p, %u/%u)", buf, (guint) buf->omx_buf->nOffset, (guint) buf->omx_buf->nAllocLen)); return GST_FLOW_ERROR; } flow_error: { gst_buffer_unmap (inbuf, &minfo); gst_buffer_unref (inbuf); return self->downstream_flow_ret; } too_large_codec_data: { gst_buffer_unmap (inbuf, &minfo); gst_buffer_unref (inbuf); GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL), ("codec_data larger than supported by OpenMAX port " "(%" G_GSIZE_FORMAT " > %u)", gst_buffer_get_size (codec_data), (guint) self->dec_in_port->port_def.nBufferSize)); return GST_FLOW_ERROR; } component_error: { gst_buffer_unmap (inbuf, &minfo); gst_buffer_unref (inbuf); GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL), ("OpenMAX component in error state %s (0x%08x)", gst_omx_component_get_last_error_string (self->dec), gst_omx_component_get_last_error (self->dec))); return GST_FLOW_ERROR; } flushing: { gst_buffer_unmap (inbuf, &minfo); gst_buffer_unref (inbuf); GST_DEBUG_OBJECT (self, "Flushing -- returning FLUSHING"); return GST_FLOW_FLUSHING; } reconfigure_error: { gst_buffer_unmap (inbuf, &minfo); gst_buffer_unref (inbuf); GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Unable to reconfigure input port")); return GST_FLOW_ERROR; } release_error: { gst_buffer_unmap (inbuf, &minfo); gst_buffer_unref (inbuf); GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to relase input buffer to component: %s (0x%08x)", gst_omx_error_to_string (err), err)); return GST_FLOW_ERROR; } } static GstFlowReturn gst_omx_audio_dec_drain (GstOMXAudioDec * self) { GstOMXAudioDecClass *klass; GstOMXBuffer *buf; GstOMXAcquireBufferReturn acq_ret; OMX_ERRORTYPE err; GST_DEBUG_OBJECT (self, "Draining component"); klass = GST_OMX_AUDIO_DEC_GET_CLASS (self); if (!self->started) { GST_DEBUG_OBJECT (self, "Component not started yet"); return GST_FLOW_OK; } self->started = FALSE; if ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER)) { GST_WARNING_OBJECT (self, "Component does not support empty EOS buffers"); 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. */ acq_ret = gst_omx_port_acquire_buffer (self->dec_in_port, &buf); if (acq_ret != GST_OMX_ACQUIRE_BUFFER_OK) { GST_AUDIO_DECODER_STREAM_LOCK (self); GST_ERROR_OBJECT (self, "Failed to acquire buffer for draining: %d", acq_ret); return GST_FLOW_ERROR; } g_mutex_lock (&self->drain_lock); self->draining = TRUE; buf->omx_buf->nFilledLen = 0; GST_OMX_SET_TICKS (buf->omx_buf->nTimeStamp, gst_util_uint64_scale (self->last_upstream_ts, OMX_TICKS_PER_SECOND, GST_SECOND)); buf->omx_buf->nTickCount = 0; buf->omx_buf->nFlags |= OMX_BUFFERFLAG_EOS; err = gst_omx_port_release_buffer (self->dec_in_port, buf); if (err != OMX_ErrorNone) { GST_ERROR_OBJECT (self, "Failed to drain component: %s (0x%08x)", gst_omx_error_to_string (err), err); g_mutex_unlock (&self->drain_lock); GST_AUDIO_DECODER_STREAM_LOCK (self); return GST_FLOW_ERROR; } GST_DEBUG_OBJECT (self, "Waiting until component is drained"); if (G_UNLIKELY (self->dec->hacks & GST_OMX_HACK_DRAIN_MAY_NOT_RETURN)) { gint64 wait_until = g_get_monotonic_time () + G_TIME_SPAN_SECOND / 2; if (!g_cond_wait_until (&self->drain_cond, &self->drain_lock, wait_until)) GST_WARNING_OBJECT (self, "Drain timed out"); else GST_DEBUG_OBJECT (self, "Drained component"); } else { g_cond_wait (&self->drain_cond, &self->drain_lock); GST_DEBUG_OBJECT (self, "Drained component"); } g_mutex_unlock (&self->drain_lock); GST_AUDIO_DECODER_STREAM_LOCK (self); gst_adapter_flush (self->output_adapter, gst_adapter_available (self->output_adapter)); self->started = FALSE; return GST_FLOW_OK; }