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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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audiotestsrc: support more than just one channel
This commit is contained in:
Tim-Philipp Müller
parent
ab75db1653
commit
72c5884f4a
2 changed files with 65 additions and 36 deletions
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@ -101,18 +101,18 @@ static GstStaticPadTemplate gst_audio_test_src_src_template =
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"width = (int) 16, "
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"depth = (int) 16, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) 1; "
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"channels = (int) [ 1, 2 ]; "
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"audio/x-raw-int, "
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"endianness = (int) BYTE_ORDER, "
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"signed = (boolean) true, "
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"width = (int) 32, "
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"depth = (int) 32,"
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"rate = (int) [ 1, MAX ], "
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"channels = (int) 1; "
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"channels = (int) [ 1, 2 ]; "
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"audio/x-raw-float, "
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"endianness = (int) BYTE_ORDER, "
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"width = (int) { 32, 64 }, "
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"rate = (int) [ 1, MAX ], " "channels = (int) 1")
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
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);
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@ -286,6 +286,9 @@ gst_audio_test_src_src_fixate (GstPad * pad, GstCaps * caps)
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gst_structure_fixate_field_nearest_int (structure, "width", 32);
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else if (strcmp (name, "audio/x-raw-float") == 0)
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gst_structure_fixate_field_nearest_int (structure, "width", 64);
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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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}
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static gboolean
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@ -333,6 +336,9 @@ gst_audio_test_src_setcaps (GstBaseSrc * basesrc, GstCaps * caps)
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break;
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}
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ret &= gst_structure_get_int (structure, "channels", &src->channels);
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GST_DEBUG_OBJECT (src, "negotiated to %d channels", src->channels);
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gst_audio_test_src_change_wave (src);
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return ret;
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@ -407,18 +413,21 @@ 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; \
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gint i, c; \
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gdouble step, amp; \
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\
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step = M_PI_M2 * src->freq / src->samplerate; \
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amp = src->volume * scale; \
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\
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for (i = 0; i < src->generate_samples_per_buffer; i++) { \
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i = 0; \
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while (i < (src->generate_samples_per_buffer * src->channels)) { \
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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[i] = (g##type) (sin (src->accumulator) * amp); \
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for (c = 0; c < src->channels; ++c) { \
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samples[i++] = (g##type) (sin (src->accumulator) * amp); \
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} \
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} \
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}
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@ -438,18 +447,21 @@ static 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; \
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gint i, c; \
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gdouble step, amp; \
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\
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step = M_PI_M2 * src->freq / src->samplerate; \
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amp = src->volume * scale; \
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\
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for (i = 0; i < src->generate_samples_per_buffer; i++) { \
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i = 0; \
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while (i < (src->generate_samples_per_buffer * src->channels)) { \
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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[i] = (g##type) ((src->accumulator < M_PI) ? amp : -amp); \
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for (c = 0; c < src->channels; ++c) { \
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samples[i++] = (g##type) ((src->accumulator < M_PI) ? amp : -amp); \
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} \
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} \
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}
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@ -469,21 +481,24 @@ static 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; \
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gint i, c; \
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gdouble step, amp; \
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\
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step = M_PI_M2 * src->freq / src->samplerate; \
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amp = (src->volume * scale) / M_PI; \
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\
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for (i = 0; i < src->generate_samples_per_buffer; i++) { \
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i = 0; \
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while (i < (src->generate_samples_per_buffer * src->channels)) { \
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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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if (src->accumulator < M_PI) { \
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samples[i] = (g##type) (src->accumulator * amp); \
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for (c = 0; c < src->channels; ++c) \
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samples[i++] = (g##type) (src->accumulator * amp); \
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} else { \
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samples[i] = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
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for (c = 0; c < src->channels; ++c) \
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samples[i++] = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
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} \
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} \
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}
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@ -504,23 +519,27 @@ static 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; \
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gint i, c; \
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gdouble step, amp; \
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\
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step = M_PI_M2 * src->freq / src->samplerate; \
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amp = (src->volume * scale) / M_PI_2; \
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\
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for (i = 0; i < src->generate_samples_per_buffer; i++) { \
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i = 0; \
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while (i < (src->generate_samples_per_buffer * src->channels)) { \
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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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if (src->accumulator < (M_PI * 0.5)) { \
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samples[i] = (g##type) (src->accumulator * amp); \
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for (c = 0; c < src->channels; ++c) \
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samples[i++] = (g##type) (src->accumulator * amp); \
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} else if (src->accumulator < (M_PI * 1.5)) { \
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samples[i] = (g##type) ((src->accumulator - M_PI) * -amp); \
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for (c = 0; c < src->channels; ++c) \
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samples[i++] = (g##type) ((src->accumulator - M_PI) * -amp); \
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} else { \
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samples[i] = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
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for (c = 0; c < src->channels; ++c) \
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samples[i++] = (g##type) ((M_PI_M2 - src->accumulator) * -amp); \
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} \
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} \
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}
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@ -541,7 +560,7 @@ static ProcessFunc triangle_funcs[] = {
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static void \
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gst_audio_test_src_create_silence_##type (GstAudioTestSrc * src, g##type * samples) \
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{ \
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memset (samples, 0, src->generate_samples_per_buffer * sizeof (g##type)); \
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memset (samples, 0, src->generate_samples_per_buffer * sizeof (g##type) * src->channels); \
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}
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DEFINE_SILENCE (int16);
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@ -560,11 +579,13 @@ static 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; \
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gint i, c; \
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gdouble amp = (src->volume * scale); \
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\
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for (i = 0; i < src->generate_samples_per_buffer; i++) { \
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samples[i] = (g##type) (amp * g_random_double_range (-1.0, 1.0)); \
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i = 0; \
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while (i < (src->generate_samples_per_buffer * src->channels)) { \
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for (c = 0; c < src->channels; ++c) \
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samples[i++] = (g##type) (amp * g_random_double_range (-1.0, 1.0)); \
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} \
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}
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@ -648,15 +669,18 @@ gst_audio_test_src_generate_pink_noise_value (GstPinkNoise * pink)
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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; \
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gint i, c; \
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gdouble amp; \
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\
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amp = src->volume * scale; \
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\
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for (i = 0; i < src->generate_samples_per_buffer; i++) { \
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samples[i] = \
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i = 0; \
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while (i < (src->generate_samples_per_buffer * src->channels)) { \
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for (c = 0; c < src->channels; ++c) { \
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samples[i++] = \
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(g##type) (gst_audio_test_src_generate_pink_noise_value (&src->pink) * \
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amp); \
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} \
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} \
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}
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@ -690,18 +714,20 @@ 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; \
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gint i, c; \
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gdouble step, scl; \
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\
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step = M_PI_M2 * src->freq / src->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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i = 0; \
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while (i < (src->generate_samples_per_buffer * src->channels)) { \
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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[i] = (g##type) scale * src->wave_table[(gint) (src->accumulator * scl)]; \
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for (c = 0; c < src->channels; ++c) \
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samples[i++] = (g##type) scale * src->wave_table[(gint) (src->accumulator * scl)]; \
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} \
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}
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@ -721,7 +747,7 @@ static 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; \
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gint i, c; \
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gdouble step, scl; \
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\
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step = M_PI_M2 * src->freq / src->samplerate; \
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@ -733,9 +759,11 @@ gst_audio_test_src_create_tick_##type (GstAudioTestSrc * src, g##type * samples)
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src->accumulator -= M_PI_M2; \
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\
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if ((src->next_sample + i)%src->samplerate < 1600) { \
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samples[i] = (g##type) scale * src->wave_table[(gint) (src->accumulator * scl)]; \
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for (c = 0; c < src->channels; ++c) \
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samples[(i * src->channels) + c] = (g##type) scale * src->wave_table[(gint) (src->accumulator * scl)]; \
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} else { \
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samples[i] = 0; \
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for (c = 0; c < src->channels; ++c) \
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samples[(i * src->channels) + c] = 0; \
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} \
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} \
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}
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@ -863,7 +891,7 @@ gst_audio_test_src_stop (GstBaseSrc * basesrc)
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}
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/* seek to time, will be called when we operate in push mode. In pull mode we
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* get the requiested byte offset. */
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* get the requested byte offset. */
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static gboolean
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gst_audio_test_src_do_seek (GstBaseSrc * basesrc, GstSegment * segment)
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{
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@ -876,7 +904,7 @@ gst_audio_test_src_do_seek (GstBaseSrc * basesrc, GstSegment * segment)
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/* now move to the time indicated */
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src->next_sample =
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gst_util_uint64_scale_int (time, src->samplerate, GST_SECOND);
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src->next_byte = src->next_sample * src->sample_size;
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src->next_byte = src->next_sample * src->sample_size * src->channels;
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src->next_time =
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gst_util_uint64_scale_int (src->next_sample, GST_SECOND, src->samplerate);
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@ -949,7 +977,7 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
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if (length == -1)
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samples = src->samples_per_buffer;
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else
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samples = length / src->sample_size;
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samples = length / (src->sample_size * src->channels);
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/* if no offset was given, use our next logical byte */
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if (offset == -1)
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@ -959,7 +987,7 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
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if (offset != src->next_byte) {
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GST_DEBUG_OBJECT (src, "seek to new offset %" G_GUINT64_FORMAT, offset);
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/* we have a discont in the expected sample offset, do a 'seek' */
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src->next_sample = src->next_byte / src->sample_size;
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src->next_sample = src->next_byte / (src->sample_size * src->channels);
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src->next_time =
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gst_util_uint64_scale_int (src->next_sample, GST_SECOND,
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src->samplerate);
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@ -981,7 +1009,7 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
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next_sample = src->next_sample + samples;
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}
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bytes = src->generate_samples_per_buffer * src->sample_size;
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bytes = src->generate_samples_per_buffer * src->sample_size * src->channels;
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if ((res = gst_pad_alloc_buffer (basesrc->srcpad, src->next_sample,
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bytes, GST_PAD_CAPS (basesrc->srcpad), &buf)) != GST_FLOW_OK) {
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@ -105,6 +105,7 @@ struct _GstAudioTestSrc {
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gdouble freq;
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/* audio parameters */
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gint channels;
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gint samplerate;
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gint samples_per_buffer;
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gint sample_size;
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