mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 04:01:08 +00:00
audiotestsrc: implement producing non-interleaved audio buffers
https://bugzilla.gnome.org/show_bug.cgi?id=796739
This commit is contained in:
George Kiagiadakis
parent
16cba63d43
commit
f6f8b979d6
1 changed files with 211 additions and 59 deletions
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@ -93,7 +93,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " FORMAT_STR ", "
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"layout = (string) interleaved, "
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"layout = (string) { interleaved, non-interleaved }, "
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"rate = " GST_AUDIO_RATE_RANGE ", "
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"channels = " GST_AUDIO_CHANNELS_RANGE)
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);
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@ -283,6 +283,8 @@ gst_audio_test_src_fixate (GstBaseSrc * bsrc, GstCaps * caps)
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gst_structure_fixate_field_string (structure, "format", DEFAULT_FORMAT_STR);
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gst_structure_fixate_field_string (structure, "layout", "interleaved");
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/* fixate to mono unless downstream requires stereo, for backwards compat */
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gst_structure_fixate_field_nearest_int (structure, "channels", 1);
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@ -393,22 +395,32 @@ error:
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static void \
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gst_audio_test_src_create_sine_##type (GstAudioTestSrc * src, g##type * samples) \
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{ \
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gint i, c, channels; \
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gint i, c, channels, channel_step, sample_step; \
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gdouble step, amp; \
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g##type *ptr; \
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\
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channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
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if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
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channel_step = 1; \
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sample_step = channels; \
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} else { \
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channel_step = src->generate_samples_per_buffer; \
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sample_step = 1; \
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} \
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step = M_PI_M2 * src->freq / GST_AUDIO_INFO_RATE (&src->info); \
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amp = src->volume * scale; \
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\
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i = 0; \
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while (i < (src->generate_samples_per_buffer * channels)) { \
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for (i = 0; i < src->generate_samples_per_buffer; i++) { \
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src->accumulator += step; \
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if (src->accumulator >= M_PI_M2) \
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src->accumulator -= M_PI_M2; \
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\
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ptr = samples; \
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for (c = 0; c < channels; ++c) { \
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samples[i++] = (g##type) (sin (src->accumulator) * amp); \
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*ptr = (g##type) (sin (src->accumulator) * amp); \
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ptr += channel_step; \
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} \
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samples += sample_step; \
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} \
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}
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@ -428,22 +440,32 @@ static const ProcessFunc sine_funcs[] = {
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static void \
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gst_audio_test_src_create_square_##type (GstAudioTestSrc * src, g##type * samples) \
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{ \
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gint i, c, channels; \
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gint i, c, channels, channel_step, sample_step; \
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gdouble step, amp; \
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g##type *ptr; \
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\
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channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
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if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
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channel_step = 1; \
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sample_step = channels; \
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} else { \
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channel_step = src->generate_samples_per_buffer; \
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sample_step = 1; \
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} \
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step = M_PI_M2 * src->freq / GST_AUDIO_INFO_RATE (&src->info); \
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amp = src->volume * scale; \
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\
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i = 0; \
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while (i < (src->generate_samples_per_buffer * channels)) { \
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for (i = 0; i < src->generate_samples_per_buffer; i++) { \
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src->accumulator += step; \
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if (src->accumulator >= M_PI_M2) \
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src->accumulator -= M_PI_M2; \
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\
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ptr = samples; \
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for (c = 0; c < channels; ++c) { \
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samples[i++] = (g##type) ((src->accumulator < G_PI) ? amp : -amp); \
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*ptr = (g##type) ((src->accumulator < G_PI) ? amp : -amp); \
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ptr += channel_step; \
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} \
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samples += sample_step; \
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} \
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}
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@ -463,26 +485,39 @@ static const ProcessFunc square_funcs[] = {
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static void \
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gst_audio_test_src_create_saw_##type (GstAudioTestSrc * src, g##type * samples) \
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{ \
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gint i, c, channels; \
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gint i, c, channels, channel_step, sample_step; \
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gdouble step, amp; \
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g##type *ptr; \
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\
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channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
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if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
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channel_step = 1; \
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sample_step = channels; \
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} else { \
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channel_step = src->generate_samples_per_buffer; \
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sample_step = 1; \
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} \
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step = M_PI_M2 * src->freq / GST_AUDIO_INFO_RATE (&src->info); \
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amp = (src->volume * scale) / G_PI; \
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\
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i = 0; \
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while (i < (src->generate_samples_per_buffer * channels)) { \
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for (i = 0; i < src->generate_samples_per_buffer; i++) { \
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src->accumulator += step; \
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if (src->accumulator >= M_PI_M2) \
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src->accumulator -= M_PI_M2; \
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\
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ptr = samples; \
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if (src->accumulator < G_PI) { \
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for (c = 0; c < channels; ++c) \
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samples[i++] = (g##type) (src->accumulator * amp); \
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for (c = 0; c < channels; ++c) { \
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*ptr = (g##type) (src->accumulator * amp); \
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ptr += channel_step; \
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} \
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} else { \
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for (c = 0; c < channels; ++c) \
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samples[i++] = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
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for (c = 0; c < channels; ++c) { \
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*ptr = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
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ptr += channel_step; \
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} \
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} \
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samples += sample_step; \
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} \
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}
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@ -502,29 +537,44 @@ static const ProcessFunc saw_funcs[] = {
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static void \
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gst_audio_test_src_create_triangle_##type (GstAudioTestSrc * src, g##type * samples) \
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{ \
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gint i, c, channels; \
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gint i, c, channels, channel_step, sample_step; \
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gdouble step, amp; \
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g##type *ptr; \
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\
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channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
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if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
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channel_step = 1; \
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sample_step = channels; \
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} else { \
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channel_step = src->generate_samples_per_buffer; \
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sample_step = 1; \
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} \
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step = M_PI_M2 * src->freq / GST_AUDIO_INFO_RATE (&src->info); \
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amp = (src->volume * scale) / G_PI_2; \
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\
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i = 0; \
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while (i < (src->generate_samples_per_buffer * channels)) { \
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for (i = 0; i < src->generate_samples_per_buffer; i++) { \
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src->accumulator += step; \
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if (src->accumulator >= M_PI_M2) \
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src->accumulator -= M_PI_M2; \
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\
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ptr = samples; \
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if (src->accumulator < (G_PI_2)) { \
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for (c = 0; c < channels; ++c) \
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samples[i++] = (g##type) (src->accumulator * amp); \
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for (c = 0; c < channels; ++c) { \
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*ptr = (g##type) (src->accumulator * amp); \
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ptr += channel_step; \
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} \
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} else if (src->accumulator < (G_PI * 1.5)) { \
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for (c = 0; c < channels; ++c) \
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samples[i++] = (g##type) ((src->accumulator - G_PI) * -amp); \
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for (c = 0; c < channels; ++c) { \
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*ptr = (g##type) ((src->accumulator - G_PI) * -amp); \
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ptr += channel_step; \
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} \
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} else { \
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for (c = 0; c < channels; ++c) \
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samples[i++] = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
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for (c = 0; c < channels; ++c) { \
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*ptr = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
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ptr += channel_step; \
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} \
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} \
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samples += sample_step; \
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} \
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}
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@ -563,14 +613,26 @@ static const ProcessFunc silence_funcs[] = {
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static void \
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gst_audio_test_src_create_white_noise_##type (GstAudioTestSrc * src, g##type * samples) \
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{ \
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gint i, c; \
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gint i, c, channel_step, sample_step; \
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g##type *ptr; \
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gdouble amp = (src->volume * scale); \
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gint channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
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\
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i = 0; \
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while (i < (src->generate_samples_per_buffer * channels)) { \
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for (c = 0; c < channels; ++c) \
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samples[i++] = (g##type) (amp * g_rand_double_range (src->gen, -1.0, 1.0)); \
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if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
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channel_step = 1; \
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sample_step = channels; \
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} else { \
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channel_step = src->generate_samples_per_buffer; \
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sample_step = 1; \
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} \
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\
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for (i = 0; i < src->generate_samples_per_buffer; i++) { \
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ptr = samples; \
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for (c = 0; c < channels; ++c) { \
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*ptr = (g##type) (amp * g_rand_double_range (src->gen, -1.0, 1.0)); \
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ptr += channel_step; \
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} \
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samples += sample_step; \
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} \
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}
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@ -657,19 +719,27 @@ gst_audio_test_src_generate_pink_noise_value (GstAudioTestSrc * src)
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static void \
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gst_audio_test_src_create_pink_noise_##type (GstAudioTestSrc * src, g##type * samples) \
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{ \
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gint i, c, channels; \
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gint i, c, channels, channel_step, sample_step; \
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gdouble amp; \
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g##type *ptr; \
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\
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amp = src->volume * scale; \
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channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
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if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
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channel_step = 1; \
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sample_step = channels; \
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} else { \
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channel_step = src->generate_samples_per_buffer; \
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sample_step = 1; \
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} \
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\
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i = 0; \
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while (i < (src->generate_samples_per_buffer * channels)) { \
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for (i = 0; i < src->generate_samples_per_buffer; i++) { \
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ptr = samples; \
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for (c = 0; c < channels; ++c) { \
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samples[i++] = \
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(g##type) (gst_audio_test_src_generate_pink_noise_value (src) * \
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amp); \
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*ptr = (g##type) (gst_audio_test_src_generate_pink_noise_value (src) * amp); \
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ptr += channel_step; \
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} \
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samples += sample_step; \
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} \
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}
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@ -703,21 +773,32 @@ gst_audio_test_src_init_sine_table (GstAudioTestSrc * src)
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static void \
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gst_audio_test_src_create_sine_table_##type (GstAudioTestSrc * src, g##type * samples) \
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{ \
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gint i, c, channels; \
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gint i, c, channels, channel_step, sample_step; \
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gdouble step, scl; \
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g##type *ptr; \
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\
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channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
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if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
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channel_step = 1; \
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sample_step = channels; \
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} else { \
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channel_step = src->generate_samples_per_buffer; \
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sample_step = 1; \
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} \
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step = M_PI_M2 * src->freq / GST_AUDIO_INFO_RATE (&src->info); \
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scl = 1024.0 / M_PI_M2; \
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\
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i = 0; \
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while (i < (src->generate_samples_per_buffer * channels)) { \
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for (i = 0; i < src->generate_samples_per_buffer; i++) { \
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src->accumulator += step; \
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if (src->accumulator >= M_PI_M2) \
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src->accumulator -= M_PI_M2; \
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\
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for (c = 0; c < channels; ++c) \
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samples[i++] = (g##type) scale * src->wave_table[(gint) (src->accumulator * scl)]; \
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ptr = samples; \
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for (c = 0; c < channels; ++c) { \
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*ptr = (g##type) scale * src->wave_table[(gint) (src->accumulator * scl)]; \
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ptr += channel_step; \
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} \
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samples += sample_step; \
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} \
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}
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@ -737,29 +818,46 @@ static const ProcessFunc sine_table_funcs[] = {
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static void \
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gst_audio_test_src_create_tick_##type (GstAudioTestSrc * src, g##type * samples) \
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{ \
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gint i, c, channels, samplerate, samplemod; \
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gint i, c, channels, samplerate, samplemod, channel_step, sample_step; \
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gdouble step, scl; \
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g##type *ptr; \
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\
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channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
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if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
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channel_step = 1; \
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sample_step = channels; \
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} else { \
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channel_step = src->generate_samples_per_buffer; \
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sample_step = 1; \
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} \
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samplerate = GST_AUDIO_INFO_RATE (&src->info); \
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step = M_PI_M2 * src->freq / samplerate; \
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scl = 1024.0 / M_PI_M2; \
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\
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for (i = 0; i < src->generate_samples_per_buffer; i++) { \
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samplemod = (src->next_sample + i) % samplerate; \
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\
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ptr = samples; \
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if (samplemod == 0) { \
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src->accumulator = 0; \
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} else if (samplemod < 1600) { \
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for (c = 0; c < channels; ++c) \
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samples[(i * channels) + c] = (g##type) scale * src->wave_table[(gint) (src->accumulator * scl)]; \
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for (c = 0; c < channels; ++c) { \
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*ptr = \
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(g##type) scale * src->wave_table[(gint) (src->accumulator * scl)]; \
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ptr += channel_step; \
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} \
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} else { \
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for (c = 0; c < channels; ++c) \
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samples[(i * channels) + c] = 0; \
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for (c = 0; c < channels; ++c) { \
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*ptr = 0; \
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ptr += channel_step; \
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} \
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} \
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\
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src->accumulator += step; \
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if (src->accumulator >= M_PI_M2) \
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src->accumulator -= M_PI_M2; \
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\
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samples += sample_step; \
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} \
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}
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@ -785,20 +883,33 @@ static const ProcessFunc tick_funcs[] = {
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static void \
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gst_audio_test_src_create_gaussian_white_noise_##type (GstAudioTestSrc * src, g##type * samples) \
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{ \
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gint i, c; \
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gint i, c, channel_step, sample_step; \
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g##type *ptr; \
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gdouble amp = (src->volume * scale); \
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gint channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
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\
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for (i = 0; i < src->generate_samples_per_buffer * channels; ) { \
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if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
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channel_step = 1; \
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sample_step = channels; \
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} else { \
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channel_step = src->generate_samples_per_buffer; \
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sample_step = 1; \
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} \
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\
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for (i = 0; i < src->generate_samples_per_buffer; i++) { \
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ptr = samples; \
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for (c = 0; c < channels; ++c) { \
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gdouble mag = sqrt (-2 * log (1.0 - g_rand_double (src->gen))); \
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gdouble phs = g_rand_double_range (src->gen, 0.0, M_PI_M2); \
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\
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samples[i++] = (g##type) (amp * mag * cos (phs)); \
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*ptr = (g##type) (amp * mag * cos (phs)); \
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ptr += channel_step; \
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if (++c >= channels) \
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break; \
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samples[i++] = (g##type) (amp * mag * sin (phs)); \
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*ptr = (g##type) (amp * mag * sin (phs)); \
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ptr += channel_step; \
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} \
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samples += sample_step; \
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} \
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}
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@ -825,12 +936,22 @@ static const ProcessFunc gaussian_white_noise_funcs[] = {
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static void \
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gst_audio_test_src_create_red_noise_##type (GstAudioTestSrc * src, g##type * samples) \
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{ \
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gint i, c; \
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gint i, c, channel_step, sample_step; \
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g##type *ptr; \
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gdouble amp = (src->volume * scale); \
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gdouble state = src->red.state; \
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gint channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
|
||||
\
|
||||
for (i = 0; i < src->generate_samples_per_buffer * channels; ) { \
|
||||
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
|
||||
channel_step = 1; \
|
||||
sample_step = channels; \
|
||||
} else { \
|
||||
channel_step = src->generate_samples_per_buffer; \
|
||||
sample_step = 1; \
|
||||
} \
|
||||
\
|
||||
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
|
||||
ptr = samples; \
|
||||
for (c = 0; c < channels; ++c) { \
|
||||
while (TRUE) { \
|
||||
gdouble r = g_rand_double_range (src->gen, -1.0, 1.0); \
|
||||
|
|
@ -838,8 +959,10 @@ gst_audio_test_src_create_red_noise_##type (GstAudioTestSrc * src, g##type * sam
|
|||
if (state < -8.0f || state > 8.0f) state -= r; \
|
||||
else break; \
|
||||
} \
|
||||
samples[i++] = (g##type) (amp * state * 0.0625f); /* /16.0 */ \
|
||||
*ptr = (g##type) (amp * state * 0.0625f); /* /16.0 */ \
|
||||
ptr += channel_step; \
|
||||
} \
|
||||
samples += sample_step; \
|
||||
} \
|
||||
src->red.state = state; \
|
||||
}
|
||||
|
|
@ -862,16 +985,28 @@ static const ProcessFunc red_noise_funcs[] = {
|
|||
static void \
|
||||
gst_audio_test_src_create_blue_noise_##type (GstAudioTestSrc * src, g##type * samples) \
|
||||
{ \
|
||||
gint i, c; \
|
||||
gint i, c, channel_step, sample_step; \
|
||||
static gdouble flip=1.0; \
|
||||
gint channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
|
||||
g##type *ptr; \
|
||||
\
|
||||
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
|
||||
channel_step = 1; \
|
||||
sample_step = channels; \
|
||||
} else { \
|
||||
channel_step = src->generate_samples_per_buffer; \
|
||||
sample_step = 1; \
|
||||
} \
|
||||
\
|
||||
gst_audio_test_src_create_pink_noise_##type (src, samples); \
|
||||
for (i = 0; i < src->generate_samples_per_buffer * channels; ) { \
|
||||
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
|
||||
ptr = samples; \
|
||||
for (c = 0; c < channels; ++c) { \
|
||||
samples[i++] *= flip; \
|
||||
*ptr *= flip; \
|
||||
ptr += channel_step; \
|
||||
} \
|
||||
flip *= -1.0; \
|
||||
samples += sample_step; \
|
||||
} \
|
||||
}
|
||||
|
||||
|
|
@ -894,16 +1029,28 @@ static const ProcessFunc blue_noise_funcs[] = {
|
|||
static void \
|
||||
gst_audio_test_src_create_violet_noise_##type (GstAudioTestSrc * src, g##type * samples) \
|
||||
{ \
|
||||
gint i, c; \
|
||||
gint i, c, channel_step, sample_step; \
|
||||
static gdouble flip=1.0; \
|
||||
gint channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
|
||||
g##type *ptr; \
|
||||
\
|
||||
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_INTERLEAVED) { \
|
||||
channel_step = 1; \
|
||||
sample_step = channels; \
|
||||
} else { \
|
||||
channel_step = src->generate_samples_per_buffer; \
|
||||
sample_step = 1; \
|
||||
} \
|
||||
\
|
||||
gst_audio_test_src_create_red_noise_##type (src, samples); \
|
||||
for (i = 0; i < src->generate_samples_per_buffer * channels; ) { \
|
||||
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
|
||||
ptr = samples; \
|
||||
for (c = 0; c < channels; ++c) { \
|
||||
samples[i++] *= flip; \
|
||||
*ptr *= flip; \
|
||||
ptr += channel_step; \
|
||||
} \
|
||||
flip *= -1.0; \
|
||||
samples += sample_step; \
|
||||
} \
|
||||
}
|
||||
|
||||
|
|
@ -1290,6 +1437,11 @@ gst_audio_test_src_fill (GstBaseSrc * basesrc, guint64 offset,
|
|||
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_GAP);
|
||||
}
|
||||
|
||||
if (GST_AUDIO_INFO_LAYOUT (&src->info) == GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
|
||||
gst_buffer_add_audio_meta (buffer, &src->info,
|
||||
src->generate_samples_per_buffer, NULL);
|
||||
}
|
||||
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
|
||||
|
|
|
|||
Loading…
Reference in a new issue