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soundtouch: Allow compilation against float and integer version of the library
https://bugzilla.gnome.org/show_bug.cgi?id=707270
This commit is contained in:
Sebastian Dröge
parent
47c35ee52e
commit
576b4826c8
3 changed files with 47 additions and 42 deletions
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@ -31,7 +31,6 @@
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#undef PACKAGE_BUGREPORT
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#undef PACKAGE
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#define FLOAT_SAMPLES 1
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#include <soundtouch/BPMDetect.h>
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#include <gst/audio/audio.h>
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@ -55,11 +54,21 @@ struct _GstBPMDetectPrivate
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#endif
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};
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#define ALLOWED_CAPS \
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"audio/x-raw, " \
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" format = (string) " GST_AUDIO_NE (F32) ", " \
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" rate = (int) [ 8000, MAX ], " \
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" channels = (int) [ 1, 2 ]"
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#if defined(SOUNDTOUCH_FLOAT_SAMPLES)
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#define ALLOWED_CAPS \
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"audio/x-raw, " \
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"format = (string) " GST_AUDIO_NE (F32) ", " \
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"rate = (int) [ 8000, MAX ], " \
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"channels = (int) [ 1, 2 ]"
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#elif defined(SOUNDTOUCH_INTEGER_SAMPLES)
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#define ALLOWED_CAPS \
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"audio/x-raw, " \
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"format = (string) " GST_AUDIO_NE (S16) ", " \
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"rate = (int) [ 8000, MAX ], " \
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"channels = (int) [ 1, 2 ]"
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#else
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#error "Only integer or float samples are supported"
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#endif
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#define gst_bpm_detect_parent_class parent_class
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G_DEFINE_TYPE (GstBPMDetect, gst_bpm_detect, GST_TYPE_AUDIO_FILTER);
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@ -209,13 +218,13 @@ gst_bpm_detect_transform_ip (GstBaseTransform * trans, GstBuffer * in)
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gst_buffer_map (in, &info, GST_MAP_READ);
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nsamples = info.size / (4 * GST_AUDIO_INFO_CHANNELS (&filter->info));
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nsamples = info.size / (GST_AUDIO_INFO_BPF (&filter->info) * GST_AUDIO_INFO_CHANNELS (&filter->info));
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/* For stereo BPMDetect->inputSamples() does downmixing into the input
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* data but our buffer data shouldn't be modified.
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*/
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if (GST_AUDIO_INFO_CHANNELS (&filter->info) == 1) {
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gfloat *inbuf = (gfloat *) info.data;
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soundtouch::SAMPLETYPE *inbuf = (soundtouch::SAMPLETYPE *) info.data;
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while (nsamples > 0) {
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bpm_detect->priv->detect->inputSamples (inbuf, MIN (nsamples, 2048));
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@ -223,13 +232,13 @@ gst_bpm_detect_transform_ip (GstBaseTransform * trans, GstBuffer * in)
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inbuf += 2048;
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}
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} else {
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gfloat *inbuf, *intmp, data[2 * 2048];
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soundtouch::SAMPLETYPE *inbuf, *intmp, data[2 * 2048];
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inbuf = (gfloat *) info.data;
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inbuf = (soundtouch::SAMPLETYPE *) info.data;
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intmp = data;
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while (nsamples > 0) {
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memcpy (intmp, inbuf, sizeof (gfloat) * 2 * MIN (nsamples, 2048));
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memcpy (intmp, inbuf, sizeof (soundtouch::SAMPLETYPE) * 2 * MIN (nsamples, 2048));
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bpm_detect->priv->detect->inputSamples (intmp, MIN (nsamples, 2048));
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nsamples -= 2048;
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inbuf += 2048 * 2;
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@ -31,7 +31,6 @@
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#undef PACKAGE_BUGREPORT
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#undef PACKAGE
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#define FLOAT_SAMPLES 1
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#include <soundtouch/SoundTouch.h>
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#include <gst/gst.h>
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@ -62,11 +61,21 @@ enum
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ARG_PITCH
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};
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#define SUPPORTED_CAPS \
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"audio/x-raw, " \
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"format = (string) " GST_AUDIO_NE (F32) ", " \
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"rate = (int) [ 8000, MAX ], " \
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"channels = (int) [ 1, 2 ]"
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#if defined(SOUNDTOUCH_FLOAT_SAMPLES)
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#define SUPPORTED_CAPS \
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"audio/x-raw, " \
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"format = (string) " GST_AUDIO_NE (F32) ", " \
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"rate = (int) [ 8000, MAX ], " \
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"channels = (int) [ 1, 2 ]"
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#elif defined(SOUNDTOUCH_INTEGER_SAMPLES)
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#define SUPPORTED_CAPS \
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"audio/x-raw, " \
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"format = (string) " GST_AUDIO_NE (S16) ", " \
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"rate = (int) [ 8000, MAX ], " \
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"channels = (int) [ 1, 2 ]"
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#else
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#error "Only integer or float samples are supported"
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#endif
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static GstStaticPadTemplate gst_pitch_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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@ -294,29 +303,17 @@ static gboolean
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gst_pitch_setcaps (GstPitch * pitch, GstCaps * caps)
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{
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GstPitchPrivate *priv;
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GstStructure *structure;
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gint rate, channels;
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priv = GST_PITCH_GET_PRIVATE (pitch);
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structure = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (structure, "rate", &rate) ||
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!gst_structure_get_int (structure, "channels", &channels)) {
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if (gst_audio_info_from_caps (&pitch->info, caps))
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return FALSE;
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}
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GST_OBJECT_LOCK (pitch);
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pitch->samplerate = rate;
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pitch->channels = channels;
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/* notify the soundtouch instance of this change */
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priv->st->setSampleRate (rate);
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priv->st->setChannels (channels);
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/* calculate sample size */
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pitch->sample_size = (sizeof (gfloat) * channels);
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priv->st->setSampleRate (pitch->info.rate);
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priv->st->setChannels (pitch->info.channels);
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GST_OBJECT_UNLOCK (pitch);
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@ -361,10 +358,10 @@ gst_pitch_prepare_buffer (GstPitch * pitch)
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if (samples == 0)
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return NULL;
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buffer = gst_buffer_new_and_alloc (samples * pitch->sample_size);
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buffer = gst_buffer_new_and_alloc (samples * pitch->info.bpf);
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gst_buffer_map (buffer, &info, (GstMapFlags) GST_MAP_READWRITE);
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samples = priv->st->receiveSamples ((gfloat *) info.data, samples);
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samples = priv->st->receiveSamples ((soundtouch::SAMPLETYPE *) info.data, samples);
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gst_buffer_unmap (buffer, &info);
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if (samples <= 0) {
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@ -373,7 +370,7 @@ gst_pitch_prepare_buffer (GstPitch * pitch)
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}
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GST_BUFFER_DURATION (buffer) =
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gst_util_uint64_scale (samples, GST_SECOND, pitch->samplerate);
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gst_util_uint64_scale (samples, GST_SECOND, pitch->info.rate);
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/* temporary store samples here, to avoid having to recalculate this */
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GST_BUFFER_OFFSET (buffer) = (gint64) samples;
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@ -471,8 +468,8 @@ gst_pitch_convert (GstPitch * pitch,
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g_return_val_if_fail (dst_format && dst_value, FALSE);
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GST_OBJECT_LOCK (pitch);
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sample_size = pitch->sample_size;
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samplerate = pitch->samplerate;
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sample_size = pitch->info.bpf;
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samplerate = pitch->info.rate;
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GST_OBJECT_UNLOCK (pitch);
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if (sample_size == 0 || samplerate == 0) {
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@ -847,7 +844,7 @@ gst_pitch_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
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/* push the received samples on the soundtouch buffer */
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GST_LOG_OBJECT (pitch, "incoming buffer (%d samples) %" GST_TIME_FORMAT,
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(gint) (gst_buffer_get_size (buffer) / pitch->sample_size),
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(gint) (gst_buffer_get_size (buffer) / pitch->info.bpf),
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GST_TIME_ARGS (timestamp));
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if (GST_PITCH_GET_PRIVATE (pitch)->pending_segment) {
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@ -872,7 +869,7 @@ gst_pitch_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
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}
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gst_buffer_map (buffer, &info, GST_MAP_READ);
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priv->st->putSamples ((gfloat *) info.data, info.size / pitch->sample_size);
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priv->st->putSamples ((soundtouch::SAMPLETYPE *) info.data, info.size / pitch->info.bpf);
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gst_buffer_unmap (buffer, &info);
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gst_buffer_unref (buffer);
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@ -21,6 +21,7 @@
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#define __GST_PITCH_H__
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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G_BEGIN_DECLS
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@ -71,9 +72,7 @@ struct _GstPitch
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gfloat seg_arate; /* Rate to apply from input segment */
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/* values extracted from caps */
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gint samplerate; /* samplerate */
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gint channels; /* number of audio channels */
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gsize sample_size; /* number of bytes for a single sample */
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GstAudioInfo info;
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/* stream tracking */
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GstClockTime next_buffer_time;
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