/* * Copyright (C) 2011, Hewlett-Packard Development Company, L.P. * Author: Sebastian Dröge , Collabora Ltd. * * 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 "gstomxaudioenc.h" GST_DEBUG_CATEGORY_STATIC (gst_omx_audio_enc_debug_category); #define GST_CAT_DEFAULT gst_omx_audio_enc_debug_category /* prototypes */ static void gst_omx_audio_enc_finalize (GObject * object); static GstStateChangeReturn gst_omx_audio_enc_change_state (GstElement * element, GstStateChange transition); static gboolean gst_omx_audio_enc_start (GstAudioEncoder * encoder); static gboolean gst_omx_audio_enc_stop (GstAudioEncoder * encoder); static gboolean gst_omx_audio_enc_set_format (GstAudioEncoder * encoder, GstAudioInfo * info); static gboolean gst_omx_audio_enc_sink_event (GstAudioEncoder * encoder, GstEvent * event); static GstFlowReturn gst_omx_audio_enc_handle_frame (GstAudioEncoder * encoder, GstBuffer * buffer); static void gst_omx_audio_enc_flush (GstAudioEncoder * encoder); static GstFlowReturn gst_omx_audio_enc_drain (GstOMXAudioEnc * self); enum { PROP_0 }; /* class initialization */ #define DEBUG_INIT \ GST_DEBUG_CATEGORY_INIT (gst_omx_audio_enc_debug_category, "omxaudioenc", 0, \ "debug category for gst-omx audio encoder base class"); G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstOMXAudioEnc, gst_omx_audio_enc, GST_TYPE_AUDIO_ENCODER, DEBUG_INIT); static void gst_omx_audio_enc_class_init (GstOMXAudioEncClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstAudioEncoderClass *audio_encoder_class = GST_AUDIO_ENCODER_CLASS (klass); gobject_class->finalize = gst_omx_audio_enc_finalize; element_class->change_state = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_change_state); audio_encoder_class->start = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_start); audio_encoder_class->stop = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_stop); audio_encoder_class->flush = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_flush); audio_encoder_class->set_format = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_set_format); audio_encoder_class->handle_frame = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_handle_frame); audio_encoder_class->sink_event = GST_DEBUG_FUNCPTR (gst_omx_audio_enc_sink_event); klass->cdata.default_sink_template_caps = "audio/x-raw, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, " G_STRINGIFY (OMX_AUDIO_MAXCHANNELS) " ], " "format = (string) { S8, U8, S16LE, S16BE, U16LE, U16BE, " "S24LE, S24BE, U24LE, U24BE, S32LE, S32BE, U32LE, U32BE }"; } static void gst_omx_audio_enc_init (GstOMXAudioEnc * self) { g_mutex_init (&self->drain_lock); g_cond_init (&self->drain_cond); } static gboolean gst_omx_audio_enc_open (GstOMXAudioEnc * self) { GstOMXAudioEncClass *klass = GST_OMX_AUDIO_ENC_GET_CLASS (self); gint in_port_index, out_port_index; self->enc = 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->enc) return FALSE; if (gst_omx_component_get_state (self->enc, 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->enc, 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", param.nPorts, param.nStartPortNumber); in_port_index = param.nStartPortNumber + 0; out_port_index = param.nStartPortNumber + 1; } } self->enc_in_port = gst_omx_component_add_port (self->enc, in_port_index); self->enc_out_port = gst_omx_component_add_port (self->enc, out_port_index); if (!self->enc_in_port || !self->enc_out_port) return FALSE; return TRUE; } static gboolean gst_omx_audio_enc_shutdown (GstOMXAudioEnc * self) { OMX_STATETYPE state; GST_DEBUG_OBJECT (self, "Shutting down encoder"); state = gst_omx_component_get_state (self->enc, 0); if (state > OMX_StateLoaded || state == OMX_StateInvalid) { if (state > OMX_StateIdle) { gst_omx_component_set_state (self->enc, OMX_StateIdle); gst_omx_component_get_state (self->enc, 5 * GST_SECOND); } gst_omx_component_set_state (self->enc, OMX_StateLoaded); gst_omx_port_deallocate_buffers (self->enc_in_port); gst_omx_port_deallocate_buffers (self->enc_out_port); if (state > OMX_StateLoaded) gst_omx_component_get_state (self->enc, 5 * GST_SECOND); } return TRUE; } static gboolean gst_omx_audio_enc_close (GstOMXAudioEnc * self) { GST_DEBUG_OBJECT (self, "Closing encoder"); if (!gst_omx_audio_enc_shutdown (self)) return FALSE; self->enc_in_port = NULL; self->enc_out_port = NULL; if (self->enc) gst_omx_component_free (self->enc); self->enc = NULL; return TRUE; } static void gst_omx_audio_enc_finalize (GObject * object) { GstOMXAudioEnc *self = GST_OMX_AUDIO_ENC (object); g_mutex_clear (&self->drain_lock); g_cond_clear (&self->drain_cond); G_OBJECT_CLASS (gst_omx_audio_enc_parent_class)->finalize (object); } static GstStateChangeReturn gst_omx_audio_enc_change_state (GstElement * element, GstStateChange transition) { GstOMXAudioEnc *self; GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; g_return_val_if_fail (GST_IS_OMX_AUDIO_ENC (element), GST_STATE_CHANGE_FAILURE); self = GST_OMX_AUDIO_ENC (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: if (!gst_omx_audio_enc_open (self)) ret = GST_STATE_CHANGE_FAILURE; break; case GST_STATE_CHANGE_READY_TO_PAUSED: if (self->enc_in_port) gst_omx_port_set_flushing (self->enc_in_port, FALSE); if (self->enc_out_port) gst_omx_port_set_flushing (self->enc_out_port, FALSE); 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->enc_in_port) gst_omx_port_set_flushing (self->enc_in_port, TRUE); if (self->enc_out_port) gst_omx_port_set_flushing (self->enc_out_port, 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; } if (ret == GST_STATE_CHANGE_FAILURE) return ret; ret = GST_ELEMENT_CLASS (gst_omx_audio_enc_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_enc_shutdown (self)) ret = GST_STATE_CHANGE_FAILURE; break; case GST_STATE_CHANGE_READY_TO_NULL: if (!gst_omx_audio_enc_close (self)) ret = GST_STATE_CHANGE_FAILURE; break; default: break; } return ret; } static void gst_omx_audio_enc_loop (GstOMXAudioEnc * self) { GstOMXAudioEncClass *klass; GstOMXPort *port = self->enc_out_port; GstOMXBuffer *buf = NULL; GstFlowReturn flow_ret = GST_FLOW_OK; GstOMXAcquireBufferReturn acq_return; gboolean is_eos; klass = GST_OMX_AUDIO_ENC_GET_CLASS (self); 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; } if (!gst_pad_has_current_caps (GST_AUDIO_ENCODER_SRC_PAD (self)) || acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) { GstAudioInfo *info = gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self)); GstCaps *caps; OMX_ERRORTYPE err; GST_DEBUG_OBJECT (self, "Port settings have changed, updating caps"); /* Reallocate all buffers */ if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) { 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; } GST_AUDIO_ENCODER_STREAM_LOCK (self); caps = klass->get_caps (self, self->enc_out_port, info); if (!caps) { if (buf) gst_omx_port_release_buffer (self->enc_out_port, buf); GST_AUDIO_ENCODER_STREAM_UNLOCK (self); goto caps_failed; } GST_DEBUG_OBJECT (self, "Setting output caps: %" GST_PTR_FORMAT, caps); if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) { gst_caps_unref (caps); if (buf) gst_omx_port_release_buffer (self->enc_out_port, buf); GST_AUDIO_ENCODER_STREAM_UNLOCK (self); goto caps_failed; } gst_caps_unref (caps); GST_AUDIO_ENCODER_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, -1); 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_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) { GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %lu", buf->omx_buf->nFlags, buf->omx_buf->nTimeStamp); /* This prevents a deadlock between the srcpad stream * lock and the videocodec stream lock, if ::reset() * is called at the wrong time */ if (gst_omx_port_is_flushing (self->enc_out_port)) { GST_DEBUG_OBJECT (self, "Flushing"); gst_omx_port_release_buffer (self->enc_out_port, buf); goto flushing; } GST_AUDIO_ENCODER_STREAM_LOCK (self); is_eos = ! !(buf->omx_buf->nFlags & OMX_BUFFERFLAG_EOS); if ((buf->omx_buf->nFlags & OMX_BUFFERFLAG_CODECCONFIG) && buf->omx_buf->nFilledLen > 0) { GstCaps *caps; GstBuffer *codec_data; GstMapInfo map = GST_MAP_INFO_INIT; GST_DEBUG_OBJECT (self, "Handling codec data"); caps = gst_caps_copy (gst_pad_get_current_caps (GST_AUDIO_ENCODER_SRC_PAD (self))); codec_data = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen); gst_buffer_map (codec_data, &map, GST_MAP_WRITE); memcpy (map.data, buf->omx_buf->pBuffer + buf->omx_buf->nOffset, buf->omx_buf->nFilledLen); gst_buffer_unmap (codec_data, &map); gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data, NULL); if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) { gst_caps_unref (caps); if (buf) gst_omx_port_release_buffer (self->enc_out_port, buf); GST_AUDIO_ENCODER_STREAM_UNLOCK (self); goto caps_failed; } gst_caps_unref (caps); flow_ret = GST_FLOW_OK; } else if (buf->omx_buf->nFilledLen > 0) { GstBuffer *outbuf; guint n_samples; GST_DEBUG_OBJECT (self, "Handling output data"); n_samples = klass->get_num_samples (self, self->enc_out_port, gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self)), buf); if (buf->omx_buf->nFilledLen > 0) { GstMapInfo map = GST_MAP_INFO_INIT; outbuf = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen); gst_buffer_map (outbuf, &map, GST_MAP_WRITE); memcpy (map.data, buf->omx_buf->pBuffer + buf->omx_buf->nOffset, buf->omx_buf->nFilledLen); gst_buffer_unmap (outbuf, &map); } else { outbuf = gst_buffer_new (); } GST_BUFFER_TIMESTAMP (outbuf) = gst_util_uint64_scale (buf->omx_buf->nTimeStamp, GST_SECOND, OMX_TICKS_PER_SECOND); if (buf->omx_buf->nTickCount != 0) GST_BUFFER_DURATION (outbuf) = gst_util_uint64_scale (buf->omx_buf->nTickCount, GST_SECOND, OMX_TICKS_PER_SECOND); flow_ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (self), outbuf, n_samples); } GST_DEBUG_OBJECT (self, "Handled output data"); if (is_eos || flow_ret == GST_FLOW_EOS) { 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); } else { GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret)); } gst_omx_port_release_buffer (port, buf); self->downstream_flow_ret = flow_ret; } else { g_assert ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER)); GST_AUDIO_ENCODER_STREAM_LOCK (self); flow_ret = GST_FLOW_EOS; } if (flow_ret != GST_FLOW_OK) goto flow_error; GST_AUDIO_ENCODER_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->enc), gst_omx_component_get_last_error (self->enc))); gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_ERROR; self->started = FALSE; return; } flushing: { GST_DEBUG_OBJECT (self, "Flushing -- stopping task"); gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_FLUSHING; self->started = FALSE; return; } flow_error: { if (flow_ret == GST_FLOW_EOS) { GST_DEBUG_OBJECT (self, "EOS"); gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self)); } else if (flow_ret == GST_FLOW_NOT_LINKED || 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_ENCODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self)); } self->started = FALSE; GST_AUDIO_ENCODER_STREAM_UNLOCK (self); return; } reconfigure_error: { GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Unable to reconfigure output port")); gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED; self->started = FALSE; return; } caps_failed: { GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to set caps")); gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ()); gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self)); self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED; self->started = FALSE; return; } } static gboolean gst_omx_audio_enc_start (GstAudioEncoder * encoder) { GstOMXAudioEnc *self; self = GST_OMX_AUDIO_ENC (encoder); self->last_upstream_ts = 0; self->eos = FALSE; self->downstream_flow_ret = GST_FLOW_OK; return TRUE; } static gboolean gst_omx_audio_enc_stop (GstAudioEncoder * encoder) { GstOMXAudioEnc *self; self = GST_OMX_AUDIO_ENC (encoder); GST_DEBUG_OBJECT (self, "Stopping encoder"); gst_omx_port_set_flushing (self->enc_in_port, TRUE); gst_omx_port_set_flushing (self->enc_out_port, TRUE); gst_pad_stop_task (GST_AUDIO_ENCODER_SRC_PAD (encoder)); if (gst_omx_component_get_state (self->enc, 0) > OMX_StateIdle) gst_omx_component_set_state (self->enc, OMX_StateIdle); self->downstream_flow_ret = GST_FLOW_FLUSHING; self->started = FALSE; self->eos = FALSE; g_mutex_lock (&self->drain_lock); self->draining = FALSE; g_cond_broadcast (&self->drain_cond); g_mutex_unlock (&self->drain_lock); gst_omx_component_get_state (self->enc, 5 * GST_SECOND); return TRUE; } static gboolean gst_omx_audio_enc_set_format (GstAudioEncoder * encoder, GstAudioInfo * info) { GstOMXAudioEnc *self; GstOMXAudioEncClass *klass; gboolean needs_disable = FALSE; OMX_PARAM_PORTDEFINITIONTYPE port_def; OMX_AUDIO_PARAM_PCMMODETYPE pcm_param; gint i; OMX_ERRORTYPE err; self = GST_OMX_AUDIO_ENC (encoder); klass = GST_OMX_AUDIO_ENC_GET_CLASS (encoder); GST_DEBUG_OBJECT (self, "Setting new caps"); /* Set audio encoder base class properties */ gst_audio_encoder_set_frame_samples_min (encoder, gst_util_uint64_scale_ceil (OMX_MIN_PCMPAYLOAD_MSEC, GST_MSECOND * info->rate, GST_SECOND)); gst_audio_encoder_set_frame_samples_max (encoder, 0); gst_omx_port_get_port_definition (self->enc_in_port, &port_def); needs_disable = gst_omx_component_get_state (self->enc, 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) { GST_DEBUG_OBJECT (self, "Need to disable and drain encoder"); gst_omx_audio_enc_drain (self); if (gst_omx_port_set_enabled (self->enc_in_port, FALSE) != OMX_ErrorNone) return FALSE; if (gst_omx_port_wait_buffers_released (self->enc_in_port, 5 * GST_SECOND) != OMX_ErrorNone) return FALSE; if (gst_omx_port_deallocate_buffers (self->enc_in_port) != OMX_ErrorNone) return FALSE; if (gst_omx_port_wait_enabled (self->enc_in_port, 1 * GST_SECOND) != OMX_ErrorNone) return FALSE; GST_DEBUG_OBJECT (self, "Encoder drained and disabled"); } port_def.format.audio.eEncoding = OMX_AUDIO_CodingPCM; GST_DEBUG_OBJECT (self, "Setting inport port definition"); if (gst_omx_port_update_port_definition (self->enc_in_port, &port_def) != OMX_ErrorNone) return FALSE; GST_OMX_INIT_STRUCT (&pcm_param); pcm_param.nPortIndex = self->enc_in_port->index; pcm_param.nChannels = info->channels; pcm_param.eNumData = ((info->finfo->flags & GST_AUDIO_FORMAT_FLAG_SIGNED) ? OMX_NumericalDataSigned : OMX_NumericalDataUnsigned); pcm_param.eEndian = ((info->finfo->endianness == G_LITTLE_ENDIAN) ? OMX_EndianLittle : OMX_EndianBig); pcm_param.bInterleaved = OMX_TRUE; pcm_param.nBitPerSample = info->finfo->width; pcm_param.nSamplingRate = info->rate; pcm_param.ePCMMode = OMX_AUDIO_PCMModeLinear; for (i = 0; i < pcm_param.nChannels; i++) { OMX_AUDIO_CHANNELTYPE pos; switch (info->position[i]) { case GST_AUDIO_CHANNEL_POSITION_MONO: case GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER: pos = OMX_AUDIO_ChannelCF; break; case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT: pos = OMX_AUDIO_ChannelLF; break; case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT: pos = OMX_AUDIO_ChannelRF; break; case GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT: pos = OMX_AUDIO_ChannelLS; break; case GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT: pos = OMX_AUDIO_ChannelRS; break; case GST_AUDIO_CHANNEL_POSITION_LFE1: pos = OMX_AUDIO_ChannelLFE; break; case GST_AUDIO_CHANNEL_POSITION_REAR_CENTER: pos = OMX_AUDIO_ChannelCS; break; case GST_AUDIO_CHANNEL_POSITION_REAR_LEFT: pos = OMX_AUDIO_ChannelLR; break; case GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT: pos = OMX_AUDIO_ChannelRR; break; default: pos = OMX_AUDIO_ChannelNone; break; } pcm_param.eChannelMapping[i] = pos; } GST_DEBUG_OBJECT (self, "Setting PCM parameters"); err = gst_omx_component_set_parameter (self->enc, OMX_IndexParamAudioPcm, &pcm_param); if (err != OMX_ErrorNone) { GST_ERROR_OBJECT (self, "Failed to set PCM parameters: %s (0x%08x)", gst_omx_error_to_string (err), err); return FALSE; } if (klass->set_format) { if (!klass->set_format (self, self->enc_in_port, info)) { 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->enc_out_port, NULL) != OMX_ErrorNone) return FALSE; GST_DEBUG_OBJECT (self, "Enabling component"); if (needs_disable) { if (gst_omx_port_set_enabled (self->enc_in_port, TRUE) != OMX_ErrorNone) return FALSE; if (gst_omx_port_allocate_buffers (self->enc_in_port, -1) != OMX_ErrorNone) return FALSE; if (gst_omx_port_wait_enabled (self->enc_in_port, 5 * GST_SECOND) != OMX_ErrorNone) return FALSE; if (gst_omx_port_mark_reconfigured (self->enc_in_port) != OMX_ErrorNone) return FALSE; } else { if (gst_omx_component_set_state (self->enc, OMX_StateIdle) != OMX_ErrorNone) return FALSE; /* Need to allocate buffers to reach Idle state */ if (gst_omx_port_allocate_buffers (self->enc_in_port, -1) != OMX_ErrorNone) return FALSE; /* And disable output port */ if (gst_omx_port_set_enabled (self->enc_out_port, FALSE) != OMX_ErrorNone) return FALSE; if (gst_omx_port_wait_enabled (self->enc_out_port, 1 * GST_SECOND) != OMX_ErrorNone) return FALSE; if (gst_omx_component_get_state (self->enc, GST_CLOCK_TIME_NONE) != OMX_StateIdle) return FALSE; if (gst_omx_component_set_state (self->enc, OMX_StateExecuting) != OMX_ErrorNone) return FALSE; if (gst_omx_component_get_state (self->enc, GST_CLOCK_TIME_NONE) != OMX_StateExecuting) return FALSE; } /* Unset flushing to allow ports to accept data again */ gst_omx_port_set_flushing (self->enc_in_port, FALSE); gst_omx_port_set_flushing (self->enc_out_port, FALSE); if (gst_omx_component_get_last_error (self->enc) != OMX_ErrorNone) { GST_ERROR_OBJECT (self, "Component in error state: %s (0x%08x)", gst_omx_component_get_last_error_string (self->enc), gst_omx_component_get_last_error (self->enc)); return FALSE; } /* Start the srcpad loop again */ GST_DEBUG_OBJECT (self, "Starting task again"); self->downstream_flow_ret = GST_FLOW_OK; gst_pad_start_task (GST_AUDIO_ENCODER_SRC_PAD (self), (GstTaskFunction) gst_omx_audio_enc_loop, encoder, NULL); return TRUE; } static void gst_omx_audio_enc_flush (GstAudioEncoder * encoder) { GstOMXAudioEnc *self; self = GST_OMX_AUDIO_ENC (encoder); GST_DEBUG_OBJECT (self, "Resetting encoder"); gst_omx_audio_enc_drain (self); gst_omx_port_set_flushing (self->enc_in_port, TRUE); gst_omx_port_set_flushing (self->enc_out_port, TRUE); /* Wait until the srcpad loop is finished */ GST_AUDIO_ENCODER_STREAM_UNLOCK (self); GST_PAD_STREAM_LOCK (GST_AUDIO_ENCODER_SRC_PAD (self)); GST_PAD_STREAM_UNLOCK (GST_AUDIO_ENCODER_SRC_PAD (self)); GST_AUDIO_ENCODER_STREAM_LOCK (self); gst_omx_port_set_flushing (self->enc_in_port, FALSE); gst_omx_port_set_flushing (self->enc_out_port, FALSE); /* Start the srcpad loop again */ self->last_upstream_ts = 0; self->downstream_flow_ret = GST_FLOW_OK; self->eos = FALSE; gst_pad_start_task (GST_AUDIO_ENCODER_SRC_PAD (self), (GstTaskFunction) gst_omx_audio_enc_loop, encoder, NULL); } static GstFlowReturn gst_omx_audio_enc_handle_frame (GstAudioEncoder * encoder, GstBuffer * inbuf) { GstOMXAcquireBufferReturn acq_ret = GST_OMX_ACQUIRE_BUFFER_ERROR; GstOMXAudioEnc *self; GstOMXPort *port; GstOMXBuffer *buf; gsize size; guint offset = 0; GstClockTime timestamp, duration, timestamp_offset = 0; self = GST_OMX_AUDIO_ENC (encoder); if (self->eos) { GST_WARNING_OBJECT (self, "Got frame after EOS"); return GST_FLOW_EOS; } if (self->downstream_flow_ret != GST_FLOW_OK) { return self->downstream_flow_ret; } if (inbuf == NULL) return GST_FLOW_OK; GST_DEBUG_OBJECT (self, "Handling frame"); timestamp = GST_BUFFER_TIMESTAMP (inbuf); duration = GST_BUFFER_DURATION (inbuf); port = self->enc_in_port; size = gst_buffer_get_size (inbuf); while (offset < 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_ENCODER_STREAM_UNLOCK (self); acq_ret = gst_omx_port_acquire_buffer (port, &buf); if (acq_ret == GST_OMX_ACQUIRE_BUFFER_ERROR) { GST_AUDIO_ENCODER_STREAM_LOCK (self); goto component_error; } else if (acq_ret == GST_OMX_ACQUIRE_BUFFER_FLUSHING) { GST_AUDIO_ENCODER_STREAM_LOCK (self); goto flushing; } else if (acq_ret == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) { OMX_ERRORTYPE err; /* Reallocate all buffers */ err = gst_omx_port_set_enabled (port, FALSE); if (err != OMX_ErrorNone) { GST_AUDIO_ENCODER_STREAM_LOCK (self); goto reconfigure_error; } err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND); if (err != OMX_ErrorNone) { GST_AUDIO_ENCODER_STREAM_LOCK (self); goto reconfigure_error; } err = gst_omx_port_deallocate_buffers (port); if (err != OMX_ErrorNone) { GST_AUDIO_ENCODER_STREAM_LOCK (self); goto reconfigure_error; } err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND); if (err != OMX_ErrorNone) { GST_AUDIO_ENCODER_STREAM_LOCK (self); goto reconfigure_error; } err = gst_omx_port_set_enabled (port, TRUE); if (err != OMX_ErrorNone) { GST_AUDIO_ENCODER_STREAM_LOCK (self); goto reconfigure_error; } err = gst_omx_port_allocate_buffers (port, -1); if (err != OMX_ErrorNone) { GST_AUDIO_ENCODER_STREAM_LOCK (self); goto reconfigure_error; } err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND); if (err != OMX_ErrorNone) { GST_AUDIO_ENCODER_STREAM_LOCK (self); goto reconfigure_error; } err = gst_omx_port_mark_reconfigured (port); if (err != OMX_ErrorNone) { GST_AUDIO_ENCODER_STREAM_LOCK (self); goto reconfigure_error; } /* Now get a new buffer and fill it */ GST_AUDIO_ENCODER_STREAM_LOCK (self); continue; } GST_AUDIO_ENCODER_STREAM_LOCK (self); g_assert (acq_ret == GST_OMX_ACQUIRE_BUFFER_OK && buf != NULL); if (self->downstream_flow_ret != GST_FLOW_OK) { gst_omx_port_release_buffer (port, buf); return self->downstream_flow_ret; } if (buf->omx_buf->nAllocLen - buf->omx_buf->nOffset <= 0) { gst_omx_port_release_buffer (port, buf); goto full_buffer; } GST_DEBUG_OBJECT (self, "Handling frame at offset %d", offset); /* Copy the buffer content in chunks of size as requested * by the port */ buf->omx_buf->nFilledLen = MIN (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); /* 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, size); } if (timestamp != GST_CLOCK_TIME_NONE) { buf->omx_buf->nTimeStamp = gst_util_uint64_scale (timestamp + timestamp_offset, OMX_TICKS_PER_SECOND, GST_SECOND); self->last_upstream_ts = timestamp + timestamp_offset; } if (duration != GST_CLOCK_TIME_NONE) { buf->omx_buf->nTickCount = gst_util_uint64_scale (buf->omx_buf->nFilledLen, duration, size); self->last_upstream_ts += duration; } offset += buf->omx_buf->nFilledLen; self->started = TRUE; gst_omx_port_release_buffer (port, buf); } GST_DEBUG_OBJECT (self, "Passed frame to component"); return self->downstream_flow_ret; full_buffer: { GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL), ("Got OpenMAX buffer with no free space (%p, %u/%u)", buf, buf->omx_buf->nOffset, buf->omx_buf->nAllocLen)); return GST_FLOW_ERROR; } 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->enc), gst_omx_component_get_last_error (self->enc))); return GST_FLOW_ERROR; } flushing: { GST_DEBUG_OBJECT (self, "Flushing -- returning FLUSHING"); return GST_FLOW_FLUSHING; } reconfigure_error: { GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Unable to reconfigure input port")); return GST_FLOW_ERROR; } } static gboolean gst_omx_audio_enc_sink_event (GstAudioEncoder * encoder, GstEvent * event) { GstOMXAudioEnc *self; GstOMXAudioEncClass *klass; self = GST_OMX_AUDIO_ENC (encoder); klass = GST_OMX_AUDIO_ENC_GET_CLASS (self); if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) { GstOMXBuffer *buf; GstOMXAcquireBufferReturn acq_ret; GST_DEBUG_OBJECT (self, "Sending EOS to the component"); /* Don't send EOS buffer twice, this doesn't work */ if (self->eos) { GST_DEBUG_OBJECT (self, "Component is already EOS"); return TRUE; } self->eos = TRUE; if ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER)) { GST_WARNING_OBJECT (self, "Component does not support empty EOS buffers"); /* Insert a NULL into the queue to signal EOS */ g_mutex_lock (&self->enc->lock); g_queue_push_tail (&self->enc_out_port->pending_buffers, NULL); g_mutex_unlock (&self->enc->lock); g_mutex_lock (&self->enc->messages_lock); g_cond_broadcast (&self->enc->messages_cond); g_mutex_unlock (&self->enc->messages_lock); return TRUE; } /* 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_ENCODER_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->enc_in_port, &buf); if (acq_ret == GST_OMX_ACQUIRE_BUFFER_OK) { buf->omx_buf->nFilledLen = 0; 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; gst_omx_port_release_buffer (self->enc_in_port, buf); GST_DEBUG_OBJECT (self, "Sent EOS to the component"); } else { GST_ERROR_OBJECT (self, "Failed to acquire buffer for EOS: %d", acq_ret); } GST_AUDIO_ENCODER_STREAM_LOCK (self); return TRUE; } return FALSE; } static GstFlowReturn gst_omx_audio_enc_drain (GstOMXAudioEnc * self) { GstOMXAudioEncClass *klass; GstOMXBuffer *buf; GstOMXAcquireBufferReturn acq_ret; GST_DEBUG_OBJECT (self, "Draining component"); klass = GST_OMX_AUDIO_ENC_GET_CLASS (self); if (!self->started) { GST_DEBUG_OBJECT (self, "Component not started yet"); return GST_FLOW_OK; } self->started = FALSE; /* Don't send EOS buffer twice, this doesn't work */ if (self->eos) { GST_DEBUG_OBJECT (self, "Component is EOS already"); return GST_FLOW_OK; } 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_ENCODER_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->enc_in_port, &buf); if (acq_ret != GST_OMX_ACQUIRE_BUFFER_OK) { GST_AUDIO_ENCODER_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; 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; gst_omx_port_release_buffer (self->enc_in_port, buf); GST_DEBUG_OBJECT (self, "Waiting until component is drained"); g_cond_wait (&self->drain_cond, &self->drain_lock); GST_DEBUG_OBJECT (self, "Drained component"); g_mutex_unlock (&self->drain_lock); GST_AUDIO_ENCODER_STREAM_LOCK (self); self->started = FALSE; return GST_FLOW_OK; }