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applemedia: atdec: 1st round of fixes
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parent
06717150bf
commit
d71aaec486
2 changed files with 33 additions and 26 deletions
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@ -66,17 +66,16 @@ static GstStaticPadTemplate gst_atdec_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE ("S16LE") ";"
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GST_AUDIO_CAPS_MAKE ("F32LE")
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)
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GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE ("S16LE") ", layout=interleaved;"
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GST_AUDIO_CAPS_MAKE ("F32LE") ", layout=interleaved")
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);
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static GstStaticPadTemplate gst_atdec_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/mpeg, mpegversion=4, framed=true,"
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"channels=[1,max]")
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GST_STATIC_CAPS ("audio/mpeg, mpegversion=4, framed=true, channels=[1,max];"
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"audio/mpeg, mpegversion=1, layer=3")
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);
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G_DEFINE_TYPE_WITH_CODE (GstATDec, gst_atdec, GST_TYPE_AUDIO_DECODER,
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@ -279,12 +278,14 @@ gst_atdec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
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// negotiate output caps
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output_caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (atdec));
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output_caps = gst_caps_fixate (output_caps);
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if (!output_caps)
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goto negotiation_error;
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output_caps =
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gst_pad_get_pad_template_caps (GST_AUDIO_DECODER_SRC_PAD (atdec));
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output_caps = gst_caps_fixate (output_caps);
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gst_caps_set_simple (output_caps,
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"rate", G_TYPE_INT, (int) input_format.mSampleRate, NULL);
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"rate", G_TYPE_INT, (int) input_format.mSampleRate,
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"channels", G_TYPE_INT, input_format.mChannelsPerFrame, NULL);
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// configure output_format from caps
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gst_caps_to_at_format (output_caps, &output_format);
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@ -320,11 +321,6 @@ gst_atdec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
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return TRUE;
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negotiation_error:
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GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
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("no compatible downstream caps"));
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return FALSE;
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create_queue_error:
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GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
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("AudioQueueNewOutput returned error: %d", status));
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@ -358,8 +354,14 @@ gst_atdec_handle_frame (GstAudioDecoder * decoder, GstBuffer * buffer)
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GstMapInfo info;
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GstAudioInfo *audio_info;
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int size, out_frames;
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GstFlowReturn flow_ret = GST_FLOW_OK;
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GstATDec *atdec = GST_ATDEC (decoder);
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if (buffer == NULL)
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return GST_FLOW_OK;
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audio_info = gst_audio_decoder_get_audio_info (decoder);
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// copy the input buffer into an AudioQueueBuffer
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size = gst_buffer_get_size (buffer);
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AudioQueueAllocateBuffer (atdec->queue, size, &input_buffer);
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@ -376,24 +378,29 @@ gst_atdec_handle_frame (GstAudioDecoder * decoder, GstBuffer * buffer)
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AudioQueueEnqueueBuffer (atdec->queue, input_buffer, 1, &packet);
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// figure out how many frames we need to pull out of the queue
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audio_info = gst_audio_decoder_get_audio_info (decoder);
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out_frames =
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GST_CLOCK_TIME_TO_FRAMES (GST_BUFFER_DURATION (buffer), audio_info->rate);
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out_frames = GST_CLOCK_TIME_TO_FRAMES (GST_BUFFER_DURATION (buffer),
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audio_info->rate);
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size = out_frames * audio_info->bpf;
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AudioQueueAllocateBuffer (atdec->queue, size, &output_buffer);
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// pull the frames
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AudioQueueOfflineRender (atdec->queue, ×tamp, output_buffer, out_frames);
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if (output_buffer->mAudioDataByteSize) {
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out =
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gst_audio_decoder_allocate_output_buffer (decoder,
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output_buffer->mAudioDataByteSize);
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gst_buffer_map (out, &info, GST_MAP_WRITE);
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memcpy (info.data, output_buffer->mAudioData,
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output_buffer->mAudioDataByteSize);
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gst_buffer_unmap (out, &info);
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flow_ret = gst_audio_decoder_finish_frame (decoder, out, 1);
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}
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AudioQueueFreeBuffer (atdec->queue, output_buffer);
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return gst_audio_decoder_finish_frame (decoder, out, 1);
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return flow_ret;
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}
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static void
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@ -1,5 +1,5 @@
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/* GStreamer
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* Copyright (C) 2013 FIXME <alessandro.d@gmail.com>
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* Copyright (C) 2013 Alessandro Decina <alessandro.d@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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