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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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mpg123: Add gapless playback support
Co-authored-by: Sebastian Dröge <sebastian@centricular.com> Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1028>
This commit is contained in:
Carlos Rafael Giani
Sebastian Dröge
parent
abb8d54bb0
commit
671c89c392
5 changed files with 403 additions and 108 deletions
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@ -71,17 +71,32 @@ GST_STATIC_PAD_TEMPLATE ("sink",
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"channels = (int) [ 1, 2 ], " "parsed = (boolean) true ")
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);
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typedef struct
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{
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guint64 clip_start, clip_end;
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} GstMpg123AudioDecClipInfo;
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static void gst_mpg123_audio_dec_dispose (GObject * object);
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static gboolean gst_mpg123_audio_dec_start (GstAudioDecoder * dec);
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static gboolean gst_mpg123_audio_dec_stop (GstAudioDecoder * dec);
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static GstFlowReturn gst_mpg123_audio_dec_push_decoded_bytes (GstMpg123AudioDec
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* mpg123_decoder, unsigned char const *decoded_bytes,
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size_t const num_decoded_bytes);
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size_t num_decoded_bytes, guint64 clip_start, guint64 clip_end);
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static GstFlowReturn gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * input_buffer);
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static gboolean gst_mpg123_audio_dec_set_format (GstAudioDecoder * dec,
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GstCaps * input_caps);
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static void gst_mpg123_audio_dec_flush (GstAudioDecoder * dec, gboolean hard);
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static void gst_mpg123_audio_dec_push_clip_info
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(GstMpg123AudioDec * mpg123_decoder, guint64 clip_start, guint64 clip_end);
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static void gst_mpg123_audio_dec_pop_oldest_clip_info (GstMpg123AudioDec *
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mpg123_decoder, guint64 * clip_start, guint64 * clip_end);
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static void gst_mpg123_audio_dec_clear_clip_info_queue (GstMpg123AudioDec *
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mpg123_decoder);
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static guint gst_mpg123_audio_dec_get_info_queue_size (GstMpg123AudioDec *
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mpg123_decoder);
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G_DEFINE_TYPE (GstMpg123AudioDec, gst_mpg123_audio_dec, GST_TYPE_AUDIO_DECODER);
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GST_ELEMENT_REGISTER_DEFINE (mpg123audiodec, "mpg123audiodec",
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GST_RANK_MARGINAL, GST_TYPE_MPG123_AUDIO_DEC);
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@ -89,6 +104,7 @@ GST_ELEMENT_REGISTER_DEFINE (mpg123audiodec, "mpg123audiodec",
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static void
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gst_mpg123_audio_dec_class_init (GstMpg123AudioDecClass * klass)
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{
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GObjectClass *object_class;
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GstAudioDecoderClass *base_class;
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GstElementClass *element_class;
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GstPadTemplate *src_template, *sink_template;
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@ -96,6 +112,7 @@ gst_mpg123_audio_dec_class_init (GstMpg123AudioDecClass * klass)
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GST_DEBUG_CATEGORY_INIT (mpg123_debug, "mpg123", 0, "mpg123 mp3 decoder");
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object_class = G_OBJECT_CLASS (klass);
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base_class = GST_AUDIO_DECODER_CLASS (klass);
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element_class = GST_ELEMENT_CLASS (klass);
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@ -178,6 +195,7 @@ gst_mpg123_audio_dec_class_init (GstMpg123AudioDecClass * klass)
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gst_element_class_add_pad_template (element_class, sink_template);
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gst_element_class_add_pad_template (element_class, src_template);
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object_class->dispose = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_dispose);
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base_class->start = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_stop);
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base_class->handle_frame =
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@ -198,6 +216,9 @@ void
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gst_mpg123_audio_dec_init (GstMpg123AudioDec * mpg123_decoder)
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{
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mpg123_decoder->handle = NULL;
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mpg123_decoder->audio_clip_info_queue =
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gst_queue_array_new_for_struct (sizeof (GstMpg123AudioDecClipInfo), 16);
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gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (mpg123_decoder), TRUE);
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gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
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(mpg123_decoder), TRUE);
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@ -205,6 +226,20 @@ gst_mpg123_audio_dec_init (GstMpg123AudioDec * mpg123_decoder)
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}
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static void
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gst_mpg123_audio_dec_dispose (GObject * object)
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{
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GstMpg123AudioDec *mpg123_decoder = GST_MPG123_AUDIO_DEC (object);
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if (mpg123_decoder->audio_clip_info_queue != NULL) {
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gst_queue_array_free (mpg123_decoder->audio_clip_info_queue);
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mpg123_decoder->audio_clip_info_queue = NULL;
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}
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G_OBJECT_CLASS (gst_mpg123_audio_dec_parent_class)->dispose (object);
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}
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static gboolean
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gst_mpg123_audio_dec_start (GstAudioDecoder * dec)
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{
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@ -271,6 +306,8 @@ gst_mpg123_audio_dec_stop (GstAudioDecoder * dec)
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mpg123_decoder->handle = NULL;
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}
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gst_mpg123_audio_dec_clear_clip_info_queue (mpg123_decoder);
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GST_INFO_OBJECT (dec, "mpg123 decoder stopped");
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return TRUE;
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@ -279,7 +316,8 @@ gst_mpg123_audio_dec_stop (GstAudioDecoder * dec)
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static GstFlowReturn
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gst_mpg123_audio_dec_push_decoded_bytes (GstMpg123AudioDec * mpg123_decoder,
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unsigned char const *decoded_bytes, size_t const num_decoded_bytes)
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unsigned char const *decoded_bytes, size_t num_decoded_bytes,
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guint64 clip_start, guint64 clip_end)
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{
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GstBuffer *output_buffer;
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GstAudioDecoder *dec;
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@ -287,15 +325,31 @@ gst_mpg123_audio_dec_push_decoded_bytes (GstMpg123AudioDec * mpg123_decoder,
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output_buffer = NULL;
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dec = GST_AUDIO_DECODER (mpg123_decoder);
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if ((num_decoded_bytes == 0) || (decoded_bytes == NULL)) {
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/* This occurs in the first few frames, which do not carry data; once
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* MPG123_AUDIO_DEC_NEW_FORMAT is received, the empty frames stop occurring */
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GST_DEBUG_OBJECT (mpg123_decoder,
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"cannot decode yet, need more data -> no output buffer to push");
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if (G_UNLIKELY ((num_decoded_bytes == 0) || (decoded_bytes == NULL))) {
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/* This occurs in two cases:
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*
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* 1. The first few frames come in. These fill mpg123's buffers, and
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* do not immediately yield decoded output. This stops once the
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* mpg123_decode_frame () returns MPG123_NEW_FORMAT.
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* 2. The decoder is being drained.
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*/
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return GST_FLOW_OK;
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}
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output_buffer = gst_buffer_new_allocate (NULL, num_decoded_bytes, NULL);
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if (G_UNLIKELY (clip_end >= num_decoded_bytes)) {
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/* Fully-clipped frames still need to be finished, since they got
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* decoded properly, they are just made of padding samples. */
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GST_LOG_OBJECT (mpg123_decoder, "frame is fully clipped; "
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"not pushing anything downstream");
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return gst_audio_decoder_finish_frame (dec, NULL, 1);
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}
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/* Apply clipping. */
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decoded_bytes += clip_start;
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num_decoded_bytes -= clip_start + clip_end;
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output_buffer = gst_audio_decoder_allocate_output_buffer (dec,
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num_decoded_bytes);
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if (output_buffer == NULL) {
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/* This is necessary to advance playback in time,
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@ -327,35 +381,98 @@ gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec,
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unsigned char *decoded_bytes;
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size_t num_decoded_bytes;
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GstFlowReturn retval;
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gboolean loop = TRUE;
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mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
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g_assert (mpg123_decoder->handle != NULL);
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/* The actual decoding */
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{
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/* feed input data (if there is any) */
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/* Feed input data (if there is any) into mpg123. */
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if (G_LIKELY (input_buffer != NULL)) {
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GstMapInfo info;
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GstAudioClippingMeta *clipping_meta = NULL;
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if (gst_buffer_map (input_buffer, &info, GST_MAP_READ)) {
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/* Drop any Xing/LAME header as marked from the parser. It's not parsed in
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* this element and would decode to unnecessary silence samples. */
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if (GST_BUFFER_FLAG_IS_SET (input_buffer, GST_BUFFER_FLAG_DECODE_ONLY) &&
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GST_BUFFER_FLAG_IS_SET (input_buffer, GST_BUFFER_FLAG_DROPPABLE)) {
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return gst_audio_decoder_finish_frame (dec, NULL, 1);
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} else if (gst_buffer_map (input_buffer, &info, GST_MAP_READ)) {
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GST_LOG_OBJECT (mpg123_decoder, "got new MPEG audio frame with %"
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G_GSIZE_FORMAT " byte(s); feeding it into mpg123", info.size);
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mpg123_feed (mpg123_decoder->handle, info.data, info.size);
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gst_buffer_unmap (input_buffer, &info);
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} else {
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GST_AUDIO_DECODER_ERROR (mpg123_decoder, 1, RESOURCE, READ, (NULL),
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("gst_memory_map() failed"), retval);
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("gst_memory_map() failed; could not feed MPEG frame into mpg123"),
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retval);
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return retval;
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}
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clipping_meta = gst_buffer_get_audio_clipping_meta (input_buffer);
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if (clipping_meta != NULL) {
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if (clipping_meta->format == GST_FORMAT_DEFAULT) {
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/* Get clipping info and convert it to bytes. */
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gint bpf = GST_AUDIO_INFO_BPF (&(mpg123_decoder->next_audioinfo));
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guint64 clip_start = clipping_meta->start * bpf;
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guint64 clip_end = clipping_meta->end * bpf;
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/* Push the clipping info into the queue. We cannot use clipping info
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* directly since mpg123 might not immediately be able to decode this
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* MPEG frame. In other words, it queues the frames internally. To
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* make sure we apply clipping properly, we therefore also have to
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* queue the clipping info accordingly. */
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gst_mpg123_audio_dec_push_clip_info (mpg123_decoder, clip_start,
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clip_end);
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GST_LOG_OBJECT (dec, "buffer has clipping metadata: start/end %"
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G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT " samples (= %"
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G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT " bytes); pushed it into "
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"audio clip info queue (now has %u item(s))", clipping_meta->start,
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clipping_meta->end, clip_start, clip_end,
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gst_mpg123_audio_dec_get_info_queue_size (mpg123_decoder));
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} else {
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gst_mpg123_audio_dec_push_clip_info (mpg123_decoder, 0, 0);
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GST_WARNING_OBJECT (dec,
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"buffer has clipping metadata in unsupported format %s",
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gst_format_get_name (clipping_meta->format));
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}
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} else {
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gst_mpg123_audio_dec_push_clip_info (mpg123_decoder, 0, 0);
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}
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} else {
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GST_LOG_OBJECT (dec, "got NULL pointer as input; "
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"will drain mpg123 decoder");
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}
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retval = GST_FLOW_OK;
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/* Keep trying to decode with mpg123 until it reports that,
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* it is done, needs more data, or an error occurs. */
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while (loop) {
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guint64 clip_start = 0, clip_end = 0;
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/* Try to decode a frame */
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decoded_bytes = NULL;
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num_decoded_bytes = 0;
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decode_error = mpg123_decode_frame (mpg123_decoder->handle,
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&mpg123_decoder->frame_offset, &decoded_bytes, &num_decoded_bytes);
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if (G_LIKELY (decoded_bytes != NULL)) {
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gst_mpg123_audio_dec_pop_oldest_clip_info (mpg123_decoder, &clip_start,
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&clip_end);
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if ((clip_start + clip_end) > 0) {
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GST_LOG_OBJECT (dec, "retrieved clip info from queue; "
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"will clip %" G_GUINT64_FORMAT " byte(s) at the start and %"
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G_GUINT64_FORMAT " at the end of the decoded frame; queue now "
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"has %u item(s)", clip_start, clip_end,
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gst_mpg123_audio_dec_get_info_queue_size (mpg123_decoder));
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}
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retval = GST_FLOW_OK;
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GST_LOG_OBJECT (dec, "decoded %" G_GSIZE_FORMAT " byte(s)", (gsize)
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num_decoded_bytes);
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}
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switch (decode_error) {
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case MPG123_NEW_FORMAT:
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@ -365,12 +482,12 @@ gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec,
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* with mp3s containing several format headers (for example, first half
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* using 44.1kHz, second half 32 kHz) */
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gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes,
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num_decoded_bytes, clip_start, clip_end);
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GST_LOG_OBJECT (dec,
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"mpg123 reported a new format -> setting next srccaps");
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gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes,
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num_decoded_bytes);
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/* If there is a next audioinfo, use it, then set has_next_audioinfo to
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* FALSE, to make sure gst_audio_decoder_set_output_format() isn't called
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* again until set_format is called by the base class */
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@ -379,6 +496,7 @@ gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec,
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&(mpg123_decoder->next_audioinfo))) {
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GST_WARNING_OBJECT (dec, "Unable to set output format");
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retval = GST_FLOW_NOT_NEGOTIATED;
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loop = FALSE;
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}
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mpg123_decoder->has_next_audioinfo = FALSE;
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}
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@ -386,25 +504,36 @@ gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec,
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break;
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case MPG123_NEED_MORE:
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loop = FALSE;
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GST_LOG_OBJECT (dec, "mpg123 needs more data to continue decoding");
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retval = gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder,
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decoded_bytes, num_decoded_bytes, clip_start, clip_end);
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break;
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case MPG123_OK:
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retval = gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder,
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decoded_bytes, num_decoded_bytes);
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decoded_bytes, num_decoded_bytes, clip_start, clip_end);
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break;
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case MPG123_DONE:
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/* If this happens, then the upstream parser somehow missed the ending
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* of the bitstream */
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GST_LOG_OBJECT (dec, "mpg123 is done decoding");
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gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes,
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num_decoded_bytes);
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num_decoded_bytes, clip_start, clip_end);
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GST_LOG_OBJECT (dec, "mpg123 is done decoding");
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retval = GST_FLOW_EOS;
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loop = FALSE;
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break;
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default:
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{
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/* Anything else is considered an error */
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int errcode;
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retval = GST_FLOW_ERROR; /* use error by default */
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/* use error by default */
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retval = GST_FLOW_ERROR;
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loop = FALSE;
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switch (decode_error) {
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case MPG123_ERR:
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errcode = mpg123_errcode (mpg123_decoder->handle);
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@ -420,7 +549,7 @@ gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec,
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("Output sample format could not be used when trying to decode frame. "
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"This is typically caused when the input caps (often the sample "
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"rate) do not match the actual format of the audio data. "
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"Input caps: %" GST_PTR_FORMAT, input_caps)
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"Input caps: %" GST_PTR_FORMAT, (gpointer) input_caps)
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);
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gst_caps_unref (input_caps);
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break;
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@ -435,6 +564,9 @@ gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec,
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}
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}
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}
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}
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GST_LOG_OBJECT (mpg123_decoder, "done handling frame");
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return retval;
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}
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@ -514,7 +646,7 @@ gst_mpg123_audio_dec_set_format (GstAudioDecoder * dec, GstCaps * input_caps)
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format_str = g_value_get_string (format_value);
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} else {
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GST_ERROR_OBJECT (mpg123_decoder, "unexpected type for 'format' field "
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"in caps structure %" GST_PTR_FORMAT, structure);
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"in caps structure %" GST_PTR_FORMAT, (gpointer) structure);
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gst_caps_unref (allowed_srccaps);
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goto done;
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}
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@ -616,12 +748,55 @@ gst_mpg123_audio_dec_flush (GstAudioDecoder * dec, gboolean hard)
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if (hard)
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mpg123_decoder->has_next_audioinfo = FALSE;
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gst_mpg123_audio_dec_clear_clip_info_queue (mpg123_decoder);
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/* opening/closing feeds do not affect the format defined by the
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* mpg123_format() call that was made in gst_mpg123_audio_dec_set_format(),
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* and since the up/downstream caps are not expected to change here, no
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* mpg123_format() calls are done */
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}
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static void gst_mpg123_audio_dec_push_clip_info
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(GstMpg123AudioDec * mpg123_decoder, guint64 clip_start, guint64 clip_end)
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{
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GstMpg123AudioDecClipInfo clip_info = { clip_start, clip_end };
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gst_queue_array_push_tail_struct (mpg123_decoder->audio_clip_info_queue,
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&clip_info);
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}
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static void
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gst_mpg123_audio_dec_pop_oldest_clip_info (GstMpg123AudioDec *
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mpg123_decoder, guint64 * clip_start, guint64 * clip_end)
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{
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guint queue_length;
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GstMpg123AudioDecClipInfo *clip_info;
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queue_length = gst_mpg123_audio_dec_get_info_queue_size (mpg123_decoder);
|
||||
if (queue_length == 0)
|
||||
return;
|
||||
|
||||
clip_info =
|
||||
gst_queue_array_pop_head_struct (mpg123_decoder->audio_clip_info_queue);
|
||||
|
||||
*clip_start = clip_info->clip_start;
|
||||
*clip_end = clip_info->clip_end;
|
||||
}
|
||||
|
||||
static void
|
||||
gst_mpg123_audio_dec_clear_clip_info_queue (GstMpg123AudioDec * mpg123_decoder)
|
||||
{
|
||||
gst_queue_array_clear (mpg123_decoder->audio_clip_info_queue);
|
||||
}
|
||||
|
||||
|
||||
static guint
|
||||
gst_mpg123_audio_dec_get_info_queue_size (GstMpg123AudioDec * mpg123_decoder)
|
||||
{
|
||||
return gst_queue_array_get_length (mpg123_decoder->audio_clip_info_queue);
|
||||
}
|
||||
|
||||
static gboolean
|
||||
plugin_init (GstPlugin * plugin)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -20,6 +20,7 @@
|
|||
#define __GST_MPG123_AUDIO_DEC_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/base/base.h>
|
||||
#include <gst/audio/gstaudiodecoder.h>
|
||||
#include <mpg123.h>
|
||||
|
||||
|
|
@ -40,6 +41,8 @@ struct _GstMpg123AudioDec
|
|||
gboolean has_next_audioinfo;
|
||||
|
||||
off_t frame_offset;
|
||||
|
||||
GstQueueArray *audio_clip_info_queue;
|
||||
};
|
||||
|
||||
GST_ELEMENT_REGISTER_DECLARE (mpg123audiodec);
|
||||
|
|
|
|||
|
|
@ -98,7 +98,7 @@
|
|||
* backwards compatibility with older hardware MP3 players, but can be safely
|
||||
* dropped.
|
||||
*
|
||||
* For more about Xng header frames, see:
|
||||
* For more about Xing header frames, see:
|
||||
* https://www.codeproject.com/Articles/8295/MPEG-Audio-Frame-Header#XINGHeader
|
||||
* https://www.compuphase.com/mp3/mp3loops.htm#PADDING_DELAYS
|
||||
*
|
||||
|
|
|
|||
|
|
@ -42,6 +42,7 @@ static GstPad *mysrcpad, *mysinkpad;
|
|||
#define MP2_STREAM_FILENAME "stream.mp2"
|
||||
#define MP3_CBR_STREAM_FILENAME "cbr_stream.mp3"
|
||||
#define MP3_VBR_STREAM_FILENAME "vbr_stream.mp3"
|
||||
#define MP3_GAPLESS_STREAM_FILENAME "sine-1009ms-1ch-32000hz-gapless-with-lame-tag.mp3"
|
||||
|
||||
|
||||
/* mpeg 1 layer 2 stream created with:
|
||||
|
|
@ -220,7 +221,7 @@ setup_mpeg1layer2dec (void)
|
|||
}
|
||||
|
||||
static GstElement *
|
||||
setup_mpeg1layer3dec (void)
|
||||
setup_mpeg1layer3dec (gint sample_rate)
|
||||
{
|
||||
GstElement *mpg123audiodec;
|
||||
GstCaps *caps;
|
||||
|
|
@ -237,7 +238,7 @@ setup_mpeg1layer3dec (void)
|
|||
caps = gst_caps_new_simple ("audio/mpeg",
|
||||
"mpegversion", G_TYPE_INT, 1,
|
||||
"layer", G_TYPE_INT, 3,
|
||||
"rate", G_TYPE_INT, 44100,
|
||||
"rate", G_TYPE_INT, sample_rate,
|
||||
"channels", G_TYPE_INT, 1, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
|
||||
gst_check_setup_events (mysrcpad, mpg123audiodec, caps, GST_FORMAT_TIME);
|
||||
gst_caps_unref (caps);
|
||||
|
|
@ -300,7 +301,7 @@ run_decoding_test (GstElement * mpg123audiodec, gchar const *filename)
|
|||
|
||||
/* This is done to be on the safe side - docs say lifetime of the input buffer
|
||||
* depends *solely* on the sample */
|
||||
input_buffer = gst_buffer_copy (input_buffer);
|
||||
input_buffer = gst_buffer_ref (input_buffer);
|
||||
|
||||
fail_unless_equals_int (gst_pad_push (mysrcpad, input_buffer), GST_FLOW_OK);
|
||||
|
||||
|
|
@ -312,7 +313,7 @@ run_decoding_test (GstElement * mpg123audiodec, gchar const *filename)
|
|||
num_decoded_buffers = g_list_length (buffers);
|
||||
|
||||
/* check number of decoded buffers */
|
||||
fail_unless_equals_int (num_decoded_buffers, num_input_buffers - 2);
|
||||
fail_unless_equals_int (num_decoded_buffers, num_input_buffers);
|
||||
|
||||
caps = gst_pad_get_current_caps (mysinkpad);
|
||||
GST_LOG ("output caps %" GST_PTR_FORMAT, caps);
|
||||
|
|
@ -333,6 +334,7 @@ run_decoding_test (GstElement * mpg123audiodec, gchar const *filename)
|
|||
/* here, test if decoded data is a sine tone, and if the sine frequency is at the
|
||||
* right spot in the spectrum */
|
||||
for (i = 0; i < num_decoded_buffers; ++i) {
|
||||
fail_if (buffers == NULL);
|
||||
outbuffer = GST_BUFFER (buffers->data);
|
||||
fail_if (outbuffer == NULL, "Invalid buffer retrieved");
|
||||
|
||||
|
|
@ -342,13 +344,12 @@ run_decoding_test (GstElement * mpg123audiodec, gchar const *filename)
|
|||
|
||||
check_main_frequency_spot_S32 (outbuffer, expected_frequency_spot);
|
||||
|
||||
buffers = g_list_remove (buffers, outbuffer);
|
||||
buffers = g_list_delete_link (buffers, buffers);
|
||||
gst_buffer_unref (outbuffer);
|
||||
outbuffer = NULL;
|
||||
}
|
||||
|
||||
g_list_free (buffers);
|
||||
buffers = NULL;
|
||||
fail_unless (buffers == NULL);
|
||||
|
||||
cleanup_input_pipeline (input_pipeline);
|
||||
gst_bus_set_flushing (bus, TRUE);
|
||||
|
|
@ -372,7 +373,7 @@ GST_END_TEST;
|
|||
GST_START_TEST (test_decode_mpeg1layer3_cbr)
|
||||
{
|
||||
GstElement *mpg123audiodec;
|
||||
mpg123audiodec = setup_mpeg1layer3dec ();
|
||||
mpg123audiodec = setup_mpeg1layer3dec (44100);
|
||||
run_decoding_test (mpg123audiodec, MP3_CBR_STREAM_FILENAME);
|
||||
cleanup_mpg123audiodec (mpg123audiodec);
|
||||
}
|
||||
|
|
@ -383,7 +384,7 @@ GST_END_TEST;
|
|||
GST_START_TEST (test_decode_mpeg1layer3_vbr)
|
||||
{
|
||||
GstElement *mpg123audiodec;
|
||||
mpg123audiodec = setup_mpeg1layer3dec ();
|
||||
mpg123audiodec = setup_mpeg1layer3dec (44100);
|
||||
run_decoding_test (mpg123audiodec, MP3_VBR_STREAM_FILENAME);
|
||||
cleanup_mpg123audiodec (mpg123audiodec);
|
||||
}
|
||||
|
|
@ -391,6 +392,117 @@ GST_START_TEST (test_decode_mpeg1layer3_vbr)
|
|||
GST_END_TEST;
|
||||
|
||||
|
||||
GST_START_TEST (test_decode_mpeg1layer3_gapless)
|
||||
{
|
||||
GstBus *bus;
|
||||
guint num_decoded_buffers;
|
||||
guint num_decoded_pcm_frames;
|
||||
GstCaps *out_caps, *caps;
|
||||
GstAudioInfo audioinfo;
|
||||
GstElement *input_pipeline, *input_appsink;
|
||||
int i;
|
||||
GstBuffer *outbuffer;
|
||||
GstElement *mpg123audiodec;
|
||||
|
||||
/* 440 Hz = frequency of sine wave in audio data
|
||||
* 32000 Hz = sample rate
|
||||
* (32000 / 2) Hz = Nyquist frequency */
|
||||
static double const expected_frequency_spot = 440.0 / (32000.0 / 2.0);
|
||||
|
||||
mpg123audiodec = setup_mpeg1layer3dec (32000);
|
||||
|
||||
fail_unless (gst_element_set_state (mpg123audiodec,
|
||||
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
|
||||
"could not set to playing");
|
||||
bus = gst_bus_new ();
|
||||
|
||||
gst_element_set_bus (mpg123audiodec, bus);
|
||||
|
||||
setup_input_pipeline (MP3_GAPLESS_STREAM_FILENAME, &input_pipeline,
|
||||
&input_appsink);
|
||||
|
||||
while (TRUE) {
|
||||
GstSample *sample;
|
||||
GstBuffer *input_buffer;
|
||||
|
||||
sample = gst_app_sink_pull_sample (GST_APP_SINK (input_appsink));
|
||||
if (sample == NULL)
|
||||
break;
|
||||
|
||||
fail_unless (GST_IS_SAMPLE (sample));
|
||||
|
||||
input_buffer = gst_sample_get_buffer (sample);
|
||||
fail_if (input_buffer == NULL);
|
||||
|
||||
/* This is done to be on the safe side - docs say lifetime of the input buffer
|
||||
* depends *solely* on the sample */
|
||||
input_buffer = gst_buffer_ref (input_buffer);
|
||||
|
||||
fail_unless_equals_int (gst_pad_push (mysrcpad, input_buffer), GST_FLOW_OK);
|
||||
|
||||
gst_sample_unref (sample);
|
||||
}
|
||||
|
||||
num_decoded_buffers = g_list_length (buffers);
|
||||
|
||||
caps = gst_pad_get_current_caps (mysinkpad);
|
||||
GST_LOG ("output caps %" GST_PTR_FORMAT, caps);
|
||||
fail_unless (gst_audio_info_from_caps (&audioinfo, caps),
|
||||
"Getting audio info from caps failed");
|
||||
|
||||
/* check caps */
|
||||
out_caps = gst_caps_new_simple ("audio/x-raw",
|
||||
"format", G_TYPE_STRING, GST_AUDIO_NE (S32),
|
||||
"layout", G_TYPE_STRING, "interleaved",
|
||||
"rate", G_TYPE_INT, 32000, "channels", G_TYPE_INT, 1, NULL);
|
||||
|
||||
fail_unless (gst_caps_is_equal_fixed (caps, out_caps), "Incorrect out caps");
|
||||
|
||||
gst_caps_unref (out_caps);
|
||||
gst_caps_unref (caps);
|
||||
|
||||
/* This is the main check. We see how many PCM frames got decoded
|
||||
* in total. If the amount is not what we expected, then gapless
|
||||
* decoding failed, because padding samples have to be omitted
|
||||
* in order for the playback to be really gapless. */
|
||||
num_decoded_pcm_frames = 0;
|
||||
for (i = 0; i < num_decoded_buffers; ++i) {
|
||||
guint num_frames;
|
||||
|
||||
fail_if (buffers == NULL);
|
||||
outbuffer = GST_BUFFER (buffers->data);
|
||||
fail_if (outbuffer == NULL, "Invalid buffer retrieved");
|
||||
|
||||
num_frames =
|
||||
gst_buffer_get_size (outbuffer) / GST_AUDIO_INFO_BPF (&audioinfo);
|
||||
num_decoded_pcm_frames += num_frames;
|
||||
|
||||
/* Don't check the first frame for a sine wave, because it will
|
||||
* unavoidably have a discontinuity at the beginning, causing the
|
||||
* spectrum to be filled with additional peaks, so the FFT check
|
||||
* will detect false positives. */
|
||||
if (i != 0)
|
||||
check_main_frequency_spot_S32 (outbuffer, expected_frequency_spot);
|
||||
|
||||
buffers = g_list_delete_link (buffers, buffers);
|
||||
gst_buffer_unref (outbuffer);
|
||||
outbuffer = NULL;
|
||||
}
|
||||
|
||||
fail_unless_equals_int (num_decoded_pcm_frames, 32288);
|
||||
fail_unless (buffers == NULL);
|
||||
|
||||
cleanup_input_pipeline (input_pipeline);
|
||||
gst_bus_set_flushing (bus, TRUE);
|
||||
gst_element_set_bus (mpg123audiodec, NULL);
|
||||
gst_object_unref (GST_OBJECT (bus));
|
||||
|
||||
cleanup_mpg123audiodec (mpg123audiodec);
|
||||
}
|
||||
|
||||
GST_END_TEST;
|
||||
|
||||
|
||||
GST_START_TEST (test_decode_garbage_mpeg1layer2)
|
||||
{
|
||||
GstElement *mpg123audiodec;
|
||||
|
|
@ -446,7 +558,7 @@ GST_START_TEST (test_decode_garbage_mpeg1layer3)
|
|||
int i, num_buffers;
|
||||
guint32 *tmpbuf;
|
||||
|
||||
mpg123audiodec = setup_mpeg1layer3dec ();
|
||||
mpg123audiodec = setup_mpeg1layer3dec (44100);
|
||||
|
||||
fail_unless (gst_element_set_state (mpg123audiodec,
|
||||
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
|
||||
|
|
@ -490,14 +602,17 @@ is_test_file_available (gchar const *filename)
|
|||
{
|
||||
gboolean ret;
|
||||
gchar *full_filename;
|
||||
gchar *cwd;
|
||||
|
||||
cwd = g_get_current_dir ();
|
||||
if (g_path_is_absolute (GST_TEST_FILES_PATH)) {
|
||||
full_filename = g_build_filename (GST_TEST_FILES_PATH, filename, NULL);
|
||||
} else {
|
||||
gchar *cwd = g_get_current_dir ();
|
||||
full_filename = g_build_filename (cwd, GST_TEST_FILES_PATH, filename, NULL);
|
||||
g_free (cwd);
|
||||
}
|
||||
ret =
|
||||
g_file_test (full_filename, G_FILE_TEST_IS_REGULAR | G_FILE_TEST_EXISTS);
|
||||
g_free (full_filename);
|
||||
g_free (cwd);
|
||||
return ret;
|
||||
}
|
||||
|
||||
|
|
@ -523,6 +638,8 @@ mpg123audiodec_suite (void)
|
|||
tcase_add_test (tc_chain, test_decode_mpeg1layer3_cbr);
|
||||
if (is_test_file_available (MP3_VBR_STREAM_FILENAME))
|
||||
tcase_add_test (tc_chain, test_decode_mpeg1layer3_vbr);
|
||||
if (is_test_file_available (MP3_GAPLESS_STREAM_FILENAME))
|
||||
tcase_add_test (tc_chain, test_decode_mpeg1layer3_gapless);
|
||||
}
|
||||
tcase_add_test (tc_chain, test_decode_garbage_mpeg1layer2);
|
||||
tcase_add_test (tc_chain, test_decode_garbage_mpeg1layer3);
|
||||
|
|
|
|||
Loading…
Reference in a new issue