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audiotestsrc: implement producing non-interleaved audio buffers

Browse source 
https://bugzilla.gnome.org/show_bug.cgi?id=796739
This commit is contained in:
George Kiagiadakis 2018-01-31 17:39:05 +02:00
parent 16cba63d43
commit f6f8b979d6
1 changed files with 211 additions and 59 deletions
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270
gst/audiotestsrc/gstaudiotestsrc.c
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@ -93,7 +93,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " FORMAT_STR ", "
"layout = (string) interleaved, "
"layout = (string) { interleaved, non-interleaved }, "
"rate = " GST_AUDIO_RATE_RANGE ", "
"channels = " GST_AUDIO_CHANNELS_RANGE)
);
@ -283,6 +283,8 @@ gst_audio_test_src_fixate (GstBaseSrc * bsrc, GstCaps * caps)
gst_structure_fixate_field_string (structure, "format", DEFAULT_FORMAT_STR);
gst_structure_fixate_field_string (structure, "layout", "interleaved");
/* fixate to mono unless downstream requires stereo, for backwards compat */
gst_structure_fixate_field_nearest_int (structure, "channels", 1);
@ -393,22 +395,32 @@ error:
static void \
gst_audio_test_src_create_sine_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channels; \
gint i, c, channels, channel_step, sample_step; \
gdouble step, amp; \
g##type *ptr; \
\
channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
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; \
} \
step = M_PI_M2 * src->freq / GST_AUDIO_INFO_RATE (&src->info); \
amp = src->volume * scale; \
\
i = 0; \
while (i < (src->generate_samples_per_buffer * channels)) { \
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
src->accumulator += step; \
if (src->accumulator >= M_PI_M2) \
src->accumulator -= M_PI_M2; \
\
ptr = samples; \
for (c = 0; c < channels; ++c) { \
samples[i++] = (g##type) (sin (src->accumulator) * amp); \
*ptr = (g##type) (sin (src->accumulator) * amp); \
ptr += channel_step; \
} \
samples += sample_step; \
} \
}
@ -428,22 +440,32 @@ static const ProcessFunc sine_funcs[] = {
static void \
gst_audio_test_src_create_square_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channels; \
gint i, c, channels, channel_step, sample_step; \
gdouble step, amp; \
g##type *ptr; \
\
channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
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; \
} \
step = M_PI_M2 * src->freq / GST_AUDIO_INFO_RATE (&src->info); \
amp = src->volume * scale; \
\
i = 0; \
while (i < (src->generate_samples_per_buffer * channels)) { \
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
src->accumulator += step; \
if (src->accumulator >= M_PI_M2) \
src->accumulator -= M_PI_M2; \
\
ptr = samples; \
for (c = 0; c < channels; ++c) { \
samples[i++] = (g##type) ((src->accumulator < G_PI) ? amp : -amp); \
*ptr = (g##type) ((src->accumulator < G_PI) ? amp : -amp); \
ptr += channel_step; \
} \
samples += sample_step; \
} \
}
@ -463,26 +485,39 @@ static const ProcessFunc square_funcs[] = {
static void \
gst_audio_test_src_create_saw_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channels; \
gint i, c, channels, channel_step, sample_step; \
gdouble step, amp; \
g##type *ptr; \
\
channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
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; \
} \
step = M_PI_M2 * src->freq / GST_AUDIO_INFO_RATE (&src->info); \
amp = (src->volume * scale) / G_PI; \
\
i = 0; \
while (i < (src->generate_samples_per_buffer * channels)) { \
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
src->accumulator += step; \
if (src->accumulator >= M_PI_M2) \
src->accumulator -= M_PI_M2; \
\
ptr = samples; \
if (src->accumulator < G_PI) { \
for (c = 0; c < channels; ++c) \
samples[i++] = (g##type) (src->accumulator * amp); \
for (c = 0; c < channels; ++c) { \
*ptr = (g##type) (src->accumulator * amp); \
ptr += channel_step; \
} \
} else { \
for (c = 0; c < channels; ++c) \
samples[i++] = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
for (c = 0; c < channels; ++c) { \
*ptr = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
ptr += channel_step; \
} \
} \
samples += sample_step; \
} \
}
@ -502,29 +537,44 @@ static const ProcessFunc saw_funcs[] = {
static void \
gst_audio_test_src_create_triangle_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channels; \
gint i, c, channels, channel_step, sample_step; \
gdouble step, amp; \
g##type *ptr; \
\
channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
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; \
} \
step = M_PI_M2 * src->freq / GST_AUDIO_INFO_RATE (&src->info); \
amp = (src->volume * scale) / G_PI_2; \
\
i = 0; \
while (i < (src->generate_samples_per_buffer * channels)) { \
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
src->accumulator += step; \
if (src->accumulator >= M_PI_M2) \
src->accumulator -= M_PI_M2; \
\
ptr = samples; \
if (src->accumulator < (G_PI_2)) { \
for (c = 0; c < channels; ++c) \
samples[i++] = (g##type) (src->accumulator * amp); \
for (c = 0; c < channels; ++c) { \
*ptr = (g##type) (src->accumulator * amp); \
ptr += channel_step; \
} \
} else if (src->accumulator < (G_PI * 1.5)) { \
for (c = 0; c < channels; ++c) \
samples[i++] = (g##type) ((src->accumulator - G_PI) * -amp); \
for (c = 0; c < channels; ++c) { \
*ptr = (g##type) ((src->accumulator - G_PI) * -amp); \
ptr += channel_step; \
} \
} else { \
for (c = 0; c < channels; ++c) \
samples[i++] = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
for (c = 0; c < channels; ++c) { \
*ptr = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
ptr += channel_step; \
} \
} \
samples += sample_step; \
} \
}
@ -563,14 +613,26 @@ static const ProcessFunc silence_funcs[] = {
static void \
gst_audio_test_src_create_white_noise_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c; \
gint i, c, channel_step, sample_step; \
g##type *ptr; \
gdouble amp = (src->volume * scale); \
gint channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
\
i = 0; \
while (i < (src->generate_samples_per_buffer * channels)) { \
for (c = 0; c < channels; ++c) \
samples[i++] = (g##type) (amp * g_rand_double_range (src->gen, -1.0, 1.0)); \
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) { \
*ptr = (g##type) (amp * g_rand_double_range (src->gen, -1.0, 1.0)); \
ptr += channel_step; \
} \
samples += sample_step; \
} \
}
@ -657,19 +719,27 @@ gst_audio_test_src_generate_pink_noise_value (GstAudioTestSrc * src)
static void \
gst_audio_test_src_create_pink_noise_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channels; \
gint i, c, channels, channel_step, sample_step; \
gdouble amp; \
g##type *ptr; \
\
amp = src->volume * scale; \
channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
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; \
} \
\
i = 0; \
while (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++] = \
(g##type) (gst_audio_test_src_generate_pink_noise_value (src) * \
amp); \
*ptr = (g##type) (gst_audio_test_src_generate_pink_noise_value (src) * amp); \
ptr += channel_step; \
} \
samples += sample_step; \
} \
}
@ -703,21 +773,32 @@ gst_audio_test_src_init_sine_table (GstAudioTestSrc * src)
static void \
gst_audio_test_src_create_sine_table_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channels; \
gint i, c, channels, channel_step, sample_step; \
gdouble step, scl; \
g##type *ptr; \
\
channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
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; \
} \
step = M_PI_M2 * src->freq / GST_AUDIO_INFO_RATE (&src->info); \
scl = 1024.0 / M_PI_M2; \
\
i = 0; \
while (i < (src->generate_samples_per_buffer * channels)) { \
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
src->accumulator += step; \
if (src->accumulator >= M_PI_M2) \
src->accumulator -= M_PI_M2; \
\
for (c = 0; c < channels; ++c) \
samples[i++] = (g##type) scale * src->wave_table[(gint) (src->accumulator * scl)]; \
ptr = samples; \
for (c = 0; c < channels; ++c) { \
*ptr = (g##type) scale * src->wave_table[(gint) (src->accumulator * scl)]; \
ptr += channel_step; \
} \
samples += sample_step; \
} \
}
@ -737,29 +818,46 @@ static const ProcessFunc sine_table_funcs[] = {
static void \
gst_audio_test_src_create_tick_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c, channels, samplerate, samplemod; \
gint i, c, channels, samplerate, samplemod, channel_step, sample_step; \
gdouble step, scl; \
g##type *ptr; \
\
channels = GST_AUDIO_INFO_CHANNELS (&src->info); \
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; \
} \
samplerate = GST_AUDIO_INFO_RATE (&src->info); \
step = M_PI_M2 * src->freq / samplerate; \
scl = 1024.0 / M_PI_M2; \
\
for (i = 0; i < src->generate_samples_per_buffer; i++) { \
samplemod = (src->next_sample + i) % samplerate; \
\
ptr = samples; \
if (samplemod == 0) { \
src->accumulator = 0; \
} else if (samplemod < 1600) { \
for (c = 0; c < channels; ++c) \
samples[(i * channels) + c] = (g##type) scale * src->wave_table[(gint) (src->accumulator * scl)]; \
for (c = 0; c < channels; ++c) { \
*ptr = \
(g##type) scale * src->wave_table[(gint) (src->accumulator * scl)]; \
ptr += channel_step; \
} \
} else { \
for (c = 0; c < channels; ++c) \
samples[(i * channels) + c] = 0; \
for (c = 0; c < channels; ++c) { \
*ptr = 0; \
ptr += channel_step; \
} \
} \
\
src->accumulator += step; \
if (src->accumulator >= M_PI_M2) \
src->accumulator -= M_PI_M2; \
\
samples += sample_step; \
} \
}
@ -785,20 +883,33 @@ static const ProcessFunc tick_funcs[] = {
static void \
gst_audio_test_src_create_gaussian_white_noise_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c; \
gint i, c, channel_step, sample_step; \
g##type *ptr; \
gdouble amp = (src->volume * scale); \
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) { \
gdouble mag = sqrt (-2 * log (1.0 - g_rand_double (src->gen))); \
gdouble phs = g_rand_double_range (src->gen, 0.0, M_PI_M2); \
\
samples[i++] = (g##type) (amp * mag * cos (phs)); \
*ptr = (g##type) (amp * mag * cos (phs)); \
ptr += channel_step; \
if (++c >= channels) \
break; \
samples[i++] = (g##type) (amp * mag * sin (phs)); \
*ptr = (g##type) (amp * mag * sin (phs)); \
ptr += channel_step; \
} \
samples += sample_step; \
} \
}
@ -825,12 +936,22 @@ static const ProcessFunc gaussian_white_noise_funcs[] = {
static void \
gst_audio_test_src_create_red_noise_##type (GstAudioTestSrc * src, g##type * samples) \
{ \
gint i, c; \
gint i, c, channel_step, sample_step; \
g##type *ptr; \
gdouble amp = (src->volume * scale); \
gdouble state = src->red.state; \
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;
}