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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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voaacenc: some code shuffle cleanup
This commit is contained in:
Mark Nauwelaerts
parent
ea92249ee2
commit
53723f81eb
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@ -94,8 +94,6 @@ GST_DEBUG_CATEGORY_STATIC (gst_voaacenc_debug);
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static gboolean voaacenc_core_init (GstVoAacEnc * voaacenc);
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static gboolean voaacenc_core_set_parameter (GstVoAacEnc * voaacenc);
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static void voaacenc_core_uninit (GstVoAacEnc * voaacenc);
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static GstCaps *gst_voaacenc_create_source_pad_caps (GstVoAacEnc * voaacenc);
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static gint voaacenc_get_rate_index (gint rate);
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static gboolean gst_voaacenc_start (GstAudioEncoder * enc);
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static gboolean gst_voaacenc_stop (GstAudioEncoder * enc);
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@ -105,105 +103,6 @@ static GstFlowReturn gst_voaacenc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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static GstCaps *gst_voaacenc_getcaps (GstAudioEncoder * enc);
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static gpointer
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gst_voaacenc_generate_sink_caps (gpointer data)
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{
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#define VOAAC_ENC_MAX_CHANNELS 6
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/* describe the channels position */
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static const GstAudioChannelPosition
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gst_voaacenc_channel_position[][VOAAC_ENC_MAX_CHANNELS] = {
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{ /* 1 ch: Mono */
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GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
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{ /* 2 ch: front left + front right (front stereo) */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
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{ /* 3 ch: front center + front stereo */
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
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{ /* 4 ch: front center + front stereo + back center */
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_CENTER},
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{ /* 5 ch: front center + front stereo + back stereo */
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
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{ /* 6ch: front center + front stereo + back stereo + LFE */
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_LFE}
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};
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GstCaps *caps = gst_caps_new_empty ();
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gint i, c;
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static const int rates[] = {
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8000, 11025, 12000, 16000, 22050, 24000,
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32000, 44100, 48000, 64000, 88200, 96000
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};
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GValue rates_arr = { 0, };
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GValue tmp = { 0, };
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g_value_init (&rates_arr, GST_TYPE_LIST);
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g_value_init (&tmp, G_TYPE_INT);
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for (i = 0; i < G_N_ELEMENTS (rates); i++) {
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g_value_set_int (&tmp, rates[i]);
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gst_value_list_append_value (&rates_arr, &tmp);
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}
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g_value_unset (&tmp);
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for (i = 0; i < 2 /*VOAAC_ENC_MAX_CHANNELS */ ; i++) {
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GValue chanpos = { 0 };
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GValue pos = { 0 };
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GstStructure *structure;
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g_value_init (&chanpos, GST_TYPE_ARRAY);
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g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
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for (c = 0; c <= i; c++) {
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g_value_set_enum (&pos, gst_voaacenc_channel_position[i][c]);
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gst_value_array_append_value (&chanpos, &pos);
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}
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g_value_unset (&pos);
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structure = gst_structure_new ("audio/x-raw-int",
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"width", G_TYPE_INT, 16,
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"depth", G_TYPE_INT, 16,
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"signed", G_TYPE_BOOLEAN, TRUE,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"channels", G_TYPE_INT, i + 1, NULL);
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gst_structure_set_value (structure, "rate", &rates_arr);
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gst_structure_set_value (structure, "channel-positions", &chanpos);
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g_value_unset (&chanpos);
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gst_caps_append_structure (caps, structure);
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}
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g_value_unset (&rates_arr);
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GST_DEBUG ("generated sink caps: %" GST_PTR_FORMAT, caps);
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return caps;
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}
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static GstCaps *
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gst_voaacenc_get_sink_caps (void)
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{
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static GOnce g_once = G_ONCE_INIT;
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GstCaps *caps;
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g_once (&g_once, gst_voaacenc_generate_sink_caps, NULL);
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caps = g_once.retval;
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return caps;
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}
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GST_BOILERPLATE (GstVoAacEnc, gst_voaacenc, GstAudioEncoder,
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GST_TYPE_AUDIO_ENCODER);
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@ -315,6 +214,111 @@ gst_voaacenc_stop (GstAudioEncoder * enc)
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return TRUE;
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}
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static gpointer
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gst_voaacenc_generate_sink_caps (gpointer data)
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{
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#define VOAAC_ENC_MAX_CHANNELS 6
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/* describe the channels position */
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static const GstAudioChannelPosition
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gst_voaacenc_channel_position[][VOAAC_ENC_MAX_CHANNELS] = {
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{ /* 1 ch: Mono */
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GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
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{ /* 2 ch: front left + front right (front stereo) */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
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{ /* 3 ch: front center + front stereo */
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
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{ /* 4 ch: front center + front stereo + back center */
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_CENTER},
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{ /* 5 ch: front center + front stereo + back stereo */
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
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{ /* 6ch: front center + front stereo + back stereo + LFE */
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_LFE}
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};
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GstCaps *caps = gst_caps_new_empty ();
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gint i, c;
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static const int rates[] = {
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8000, 11025, 12000, 16000, 22050, 24000,
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32000, 44100, 48000, 64000, 88200, 96000
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};
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GValue rates_arr = { 0, };
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GValue tmp = { 0, };
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g_value_init (&rates_arr, GST_TYPE_LIST);
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g_value_init (&tmp, G_TYPE_INT);
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for (i = 0; i < G_N_ELEMENTS (rates); i++) {
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g_value_set_int (&tmp, rates[i]);
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gst_value_list_append_value (&rates_arr, &tmp);
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}
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g_value_unset (&tmp);
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for (i = 0; i < 2 /*VOAAC_ENC_MAX_CHANNELS */ ; i++) {
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GValue chanpos = { 0 };
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GValue pos = { 0 };
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GstStructure *structure;
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g_value_init (&chanpos, GST_TYPE_ARRAY);
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g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
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for (c = 0; c <= i; c++) {
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g_value_set_enum (&pos, gst_voaacenc_channel_position[i][c]);
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gst_value_array_append_value (&chanpos, &pos);
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}
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g_value_unset (&pos);
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structure = gst_structure_new ("audio/x-raw-int",
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"width", G_TYPE_INT, 16,
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"depth", G_TYPE_INT, 16,
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"signed", G_TYPE_BOOLEAN, TRUE,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"channels", G_TYPE_INT, i + 1, NULL);
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gst_structure_set_value (structure, "rate", &rates_arr);
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gst_structure_set_value (structure, "channel-positions", &chanpos);
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g_value_unset (&chanpos);
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gst_caps_append_structure (caps, structure);
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}
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g_value_unset (&rates_arr);
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GST_DEBUG ("generated sink caps: %" GST_PTR_FORMAT, caps);
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return caps;
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}
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static GstCaps *
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gst_voaacenc_get_sink_caps (void)
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{
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static GOnce g_once = G_ONCE_INIT;
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GstCaps *caps;
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g_once (&g_once, gst_voaacenc_generate_sink_caps, NULL);
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caps = g_once.retval;
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return caps;
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}
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static GstCaps *
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gst_voaacenc_getcaps (GstAudioEncoder * benc)
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{
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return gst_audio_encoder_proxy_getcaps (benc, gst_voaacenc_get_sink_caps ());
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}
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/* check downstream caps to configure format */
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static void
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gst_voaacenc_negotiate (GstVoAacEnc * voaacenc)
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@ -347,10 +351,58 @@ gst_voaacenc_negotiate (GstVoAacEnc * voaacenc)
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gst_caps_unref (caps);
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}
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static GstCaps *
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gst_voaacenc_getcaps (GstAudioEncoder * benc)
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static gint
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gst_voaacenc_get_rate_index (gint rate)
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{
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return gst_audio_encoder_proxy_getcaps (benc, gst_voaacenc_get_sink_caps ());
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static const gint rate_table[] = {
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96000, 88200, 64000, 48000, 44100, 32000,
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24000, 22050, 16000, 12000, 11025, 8000
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};
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gint i;
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for (i = 0; i < G_N_ELEMENTS (rate_table); ++i) {
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if (rate == rate_table[i]) {
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return i;
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}
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}
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return -1;
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}
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static GstCaps *
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gst_voaacenc_create_source_pad_caps (GstVoAacEnc * voaacenc)
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{
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GstCaps *caps = NULL;
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GstBuffer *codec_data;
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gint index;
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guint8 data[VOAAC_ENC_CODECDATA_LEN];
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if ((index = gst_voaacenc_get_rate_index (voaacenc->rate)) >= 0) {
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/* LC profile only */
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data[0] = ((0x02 << 3) | (index >> 1));
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data[1] = ((index & 0x01) << 7) | (voaacenc->channels << 3);
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caps = gst_caps_new_simple ("audio/mpeg",
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"mpegversion", G_TYPE_INT, VOAAC_ENC_MPEGVERSION,
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"channels", G_TYPE_INT, voaacenc->channels,
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"rate", G_TYPE_INT, voaacenc->rate,
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"stream-format", G_TYPE_STRING,
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(voaacenc->output_format ? "adts" : "raw")
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, NULL);
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gst_codec_utils_aac_caps_set_level_and_profile (caps, data, sizeof (data));
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if (!voaacenc->output_format) {
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codec_data = gst_buffer_new_and_alloc (VOAAC_ENC_CODECDATA_LEN);
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memcpy (GST_BUFFER_DATA (codec_data), data, sizeof (data));
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gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data,
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NULL);
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gst_buffer_unref (codec_data);
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}
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}
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return caps;
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}
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static gboolean
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@ -471,45 +523,6 @@ encode_failed:
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}
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}
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static GstCaps *
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gst_voaacenc_create_source_pad_caps (GstVoAacEnc * voaacenc)
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{
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GstCaps *caps = NULL;
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GstBuffer *codec_data;
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gint index;
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guint8 data[VOAAC_ENC_CODECDATA_LEN];
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if ((index = voaacenc_get_rate_index (voaacenc->rate)) >= 0) {
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/* LC profile only */
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data[0] = ((0x02 << 3) | (index >> 1));
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data[1] = ((index & 0x01) << 7) | (voaacenc->channels << 3);
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caps = gst_caps_new_simple ("audio/mpeg",
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"mpegversion", G_TYPE_INT, VOAAC_ENC_MPEGVERSION,
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"channels", G_TYPE_INT, voaacenc->channels,
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"rate", G_TYPE_INT, voaacenc->rate,
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"stream-format", G_TYPE_STRING,
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(voaacenc->output_format ? "adts" : "raw")
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, NULL);
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gst_codec_utils_aac_caps_set_level_and_profile (caps, data, sizeof (data));
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if (!voaacenc->output_format) {
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codec_data = gst_buffer_new_and_alloc (VOAAC_ENC_CODECDATA_LEN);
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memcpy (GST_BUFFER_DATA (codec_data), data, sizeof (data));
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gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data,
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NULL);
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gst_buffer_unref (codec_data);
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}
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}
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return caps;
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}
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static VO_U32
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voaacenc_core_mem_alloc (VO_S32 uID, VO_MEM_INFO * pMemInfo)
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{
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@ -602,19 +615,3 @@ voaacenc_core_uninit (GstVoAacEnc * voaacenc)
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voaacenc->handle = NULL;
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}
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}
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static gint
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voaacenc_get_rate_index (gint rate)
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{
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static const gint rate_table[] = {
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96000, 88200, 64000, 48000, 44100, 32000,
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24000, 22050, 16000, 12000, 11025, 8000
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};
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gint i;
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for (i = 0; i < G_N_ELEMENTS (rate_table); ++i) {
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if (rate == rate_table[i]) {
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return i;
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}
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}
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return -1;
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}
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