Пример #1
0
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, 5 * GST_SECOND, TRUE);
  gst_omx_port_set_flushing (self->enc_out_port, 5 * GST_SECOND, 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, 5 * GST_SECOND, FALSE);
  gst_omx_port_set_flushing (self->enc_out_port, 5 * GST_SECOND, FALSE);
  gst_omx_port_populate (self->enc_out_port);

  /* 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_vorbis_enc_output_buffers (GstVorbisEnc * vorbisenc)
{
  GstFlowReturn ret;

  /* vorbis does some data preanalysis, then divides up blocks for
     more involved (potentially parallel) processing.  Get a single
     block for encoding now */
  while (vorbis_analysis_blockout (&vorbisenc->vd, &vorbisenc->vb) == 1) {
    ogg_packet op;

    GST_LOG_OBJECT (vorbisenc, "analysed to a block");

    /* analysis */
    vorbis_analysis (&vorbisenc->vb, NULL);
    vorbis_bitrate_addblock (&vorbisenc->vb);

    while (vorbis_bitrate_flushpacket (&vorbisenc->vd, &op)) {
      GstBuffer *buf;

      if (op.e_o_s) {
        GstAudioEncoder *enc = GST_AUDIO_ENCODER (vorbisenc);
        GstClockTime duration;

        GST_DEBUG_OBJECT (vorbisenc, "Got EOS packet from libvorbis");
        GST_AUDIO_ENCODER_STREAM_LOCK (enc);
        if (!GST_CLOCK_TIME_IS_VALID (enc->output_segment.stop)) {
          GST_DEBUG_OBJECT (vorbisenc,
              "Output segment has no end time, setting");
          duration =
              gst_util_uint64_scale (op.granulepos, GST_SECOND,
              vorbisenc->frequency);
          enc->output_segment.stop = enc->output_segment.start + duration;
          GST_DEBUG_OBJECT (enc, "new output segment %" GST_SEGMENT_FORMAT,
              &enc->output_segment);
          gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
              gst_event_new_segment (&enc->output_segment));
        }
        GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
      }

      GST_LOG_OBJECT (vorbisenc, "pushing out a data packet");
      buf =
          gst_audio_encoder_allocate_output_buffer (GST_AUDIO_ENCODER
          (vorbisenc), op.bytes);
      gst_buffer_fill (buf, 0, op.packet, op.bytes);
      /* tracking granulepos should tell us samples accounted for */
      ret =
          gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER
          (vorbisenc), buf, op.granulepos - vorbisenc->samples_out);
      vorbisenc->samples_out = op.granulepos;

      if (ret != GST_FLOW_OK)
        return ret;
    }
  }

  return GST_FLOW_OK;
}
Пример #3
0
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;
  OMX_ERRORTYPE err;

  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) {
      /* 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);
      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;
    err = gst_omx_port_release_buffer (port, buf);
    if (err != OMX_ErrorNone)
      goto release_error;
  }

  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,
            (guint) buf->omx_buf->nOffset, (guint) 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;
  }
release_error:
  {
    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;
  }
}
Пример #4
0
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);
    gst_omx_port_set_flushing (self->enc_out_port, 5 * GST_SECOND, TRUE);

    /* Wait until the srcpad loop is finished,
     * unlock GST_AUDIO_ENCODER_STREAM_LOCK to prevent deadlocks
     * caused by using this lock from inside the loop function */
    GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
    gst_pad_stop_task (GST_AUDIO_ENCODER_SRC_PAD (encoder));
    GST_AUDIO_ENCODER_STREAM_LOCK (self);

    if (gst_omx_port_set_enabled (self->enc_in_port, FALSE) != OMX_ErrorNone)
      return FALSE;
    if (gst_omx_port_set_enabled (self->enc_out_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_wait_buffers_released (self->enc_out_port,
            1 * GST_SECOND) != OMX_ErrorNone)
      return FALSE;
    if (gst_omx_port_deallocate_buffers (self->enc_in_port) != OMX_ErrorNone)
      return FALSE;
    if (gst_omx_port_deallocate_buffers (self->enc_out_port) != OMX_ErrorNone)
      return FALSE;
    if (gst_omx_port_wait_enabled (self->enc_in_port,
            1 * GST_SECOND) != OMX_ErrorNone)
      return FALSE;
    if (gst_omx_port_wait_enabled (self->enc_out_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) != 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 {
    /* 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_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) != 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, 5 * GST_SECOND, FALSE);
  gst_omx_port_set_flushing (self->enc_out_port, 5 * GST_SECOND, 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;
}
Пример #5
0
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;
  OMX_ERRORTYPE err;

  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;
  } else if (acq_return == GST_OMX_ACQUIRE_BUFFER_EOS) {
    goto eos;
  }

  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;

    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);
      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_ENCODER_STREAM_LOCK (self);
    goto eos;
  }

  GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %" G_GUINT64_FORMAT,
      (guint) buf->omx_buf->nFlags, (guint64) 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);

  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");

  GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));

  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_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;
  }
eos:
  {
    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_ENCODER_SRC_PAD (self));
    } else {
      GST_DEBUG_OBJECT (self, "Component signalled EOS");
      flow_ret = GST_FLOW_EOS;
    }
    g_mutex_unlock (&self->drain_lock);

    GST_AUDIO_ENCODER_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_ENCODER_STREAM_UNLOCK (self);

    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;
  }
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_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;
    GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
    return;
  }
}
Пример #6
0
static GstFlowReturn
gst_omx_audio_enc_drain (GstOMXAudioEnc * self)
{
  GstOMXAudioEncClass *klass;
  GstOMXBuffer *buf;
  GstOMXAcquireBufferReturn acq_ret;
  OMX_ERRORTYPE err;

  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;
  err = gst_omx_port_release_buffer (self->enc_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);
    GST_AUDIO_ENCODER_STREAM_LOCK (self);
    return GST_FLOW_ERROR;
  }
  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;
}
Пример #7
0
static gboolean
gst_omx_audio_enc_sink_event (GstAudioEncoder * encoder, GstEvent * event)
{
  GstOMXAudioEnc *self;
  GstOMXAudioEncClass *klass;
  OMX_ERRORTYPE err;

  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;
      err = gst_omx_port_release_buffer (self->enc_in_port, buf);
      if (err != OMX_ErrorNone) {
        GST_ERROR_OBJECT (self, "Failed to send EOS to component: %s (0x%08x)",
            gst_omx_error_to_string (err), err);
      } else {
        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;
}