shout: moved code to audioOutput_shout_ogg.c
Begin dividing audioOutput_shout.c: move everything OGG Vorbis related to audioOutput_shout_ogg.c. The header audioOutput_shout.h has to keep its dependency on vorbis/vorbisenc.h, because it needs the vorbis encoder types. For this patch, we have to export several internal functions with generic names to the ABI; these will be removed later when the encoder plugin patches are merged.
This commit is contained in:
Max Kellermann
parent
7d3ca9c1ff
commit
fcac05a207
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@ -3,6 +3,7 @@ SUBDIRS = $(MP4FF_SUBDIR)
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mpd_audioOutputs = \
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audioOutputs/audioOutput_shout.c \
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audioOutputs/audioOutput_shout_ogg.c \
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audioOutputs/audioOutput_null.c \
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audioOutputs/audioOutput_fifo.c \
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audioOutputs/audioOutput_alsa.c \
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@ -248,7 +248,7 @@ static int handle_shout_error(struct shout_data *sd, int err)
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return 0;
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}
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static int write_page(struct shout_data *sd)
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int write_page(struct shout_data *sd)
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{
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int err;
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@ -263,39 +263,6 @@ static int write_page(struct shout_data *sd)
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return 0;
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}
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static void finish_encoder(struct shout_data *sd)
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{
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vorbis_analysis_wrote(&sd->vd, 0);
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while (vorbis_analysis_blockout(&sd->vd, &sd->vb) == 1) {
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vorbis_analysis(&sd->vb, NULL);
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vorbis_bitrate_addblock(&sd->vb);
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while (vorbis_bitrate_flushpacket(&sd->vd, &sd->op)) {
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ogg_stream_packetin(&sd->os, &sd->op);
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}
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}
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}
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static int flush_encoder(struct shout_data *sd)
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{
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return (ogg_stream_pageout(&sd->os, &sd->og) > 0);
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}
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static void shout_ogg_encoder_clear_encoder(struct shout_data *sd)
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{
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finish_encoder(sd);
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while (1 == flush_encoder(sd)) {
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if (!sd->shout_error)
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write_page(sd);
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}
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vorbis_comment_clear(&sd->vc);
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ogg_stream_clear(&sd->os);
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vorbis_block_clear(&sd->vb);
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vorbis_dsp_clear(&sd->vd);
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vorbis_info_clear(&sd->vi);
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}
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static void close_shout_conn(struct shout_data *sd)
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{
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if (sd->opened)
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@ -346,72 +313,6 @@ static void my_shout_close_device(struct audio_output *audio_output)
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audio_output->open = 0;
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}
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static void add_tag(struct shout_data *sd, const char *name, char *value)
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{
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if (value) {
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union const_hack u;
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u.in = name;
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vorbis_comment_add_tag(&(sd->vc), u.out, value);
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}
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}
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static void copy_tag_to_vorbis_comment(struct shout_data *sd)
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{
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if (sd->tag) {
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int i;
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for (i = 0; i < sd->tag->numOfItems; i++) {
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switch (sd->tag->items[i]->type) {
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case TAG_ITEM_ARTIST:
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add_tag(sd, "ARTIST", sd->tag->items[i]->value);
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break;
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case TAG_ITEM_ALBUM:
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add_tag(sd, "ALBUM", sd->tag->items[i]->value);
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break;
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case TAG_ITEM_TITLE:
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add_tag(sd, "TITLE", sd->tag->items[i]->value);
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break;
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default:
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break;
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}
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}
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}
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}
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static int init_encoder(struct shout_data *sd)
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{
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vorbis_info_init(&(sd->vi));
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if (sd->quality >= -1.0) {
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if (0 != vorbis_encode_init_vbr(&(sd->vi),
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sd->audio_format.channels,
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sd->audio_format.sampleRate,
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sd->quality * 0.1)) {
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ERROR("problem setting up vorbis encoder for shout\n");
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vorbis_info_clear(&(sd->vi));
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return -1;
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}
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} else {
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if (0 != vorbis_encode_init(&(sd->vi),
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sd->audio_format.channels,
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sd->audio_format.sampleRate, -1.0,
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sd->bitrate * 1000, -1.0)) {
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ERROR("problem setting up vorbis encoder for shout\n");
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vorbis_info_clear(&(sd->vi));
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return -1;
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}
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}
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vorbis_analysis_init(&(sd->vd), &(sd->vi));
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vorbis_block_init(&(sd->vd), &(sd->vb));
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ogg_stream_init(&(sd->os), rand());
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vorbis_comment_init(&(sd->vc));
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return 0;
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}
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static int shout_connect(struct shout_data *sd)
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{
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time_t t = time(NULL);
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@ -486,13 +387,7 @@ static int open_shout_conn(struct audio_output *audio_output)
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copy_tag_to_vorbis_comment(sd);
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vorbis_analysis_headerout(&(sd->vd), &(sd->vc), &(sd->header_main),
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&(sd->header_comments),
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&(sd->header_codebooks));
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ogg_stream_packetin(&(sd->os), &(sd->header_main));
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ogg_stream_packetin(&(sd->os), &(sd->header_comments));
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ogg_stream_packetin(&(sd->os), &(sd->header_codebooks));
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send_ogg_vorbis_header(sd);
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sd->opened = 1;
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sd->tag_to_send = 0;
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@ -531,25 +426,9 @@ static void send_metadata(struct shout_data * sd)
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if (!sd->opened || !sd->tag)
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return;
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shout_ogg_encoder_clear_encoder(sd);
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if (init_encoder(sd) < 0)
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if (shout_ogg_encoder_send_metadata(sd)) {
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close_shout_conn(sd);
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return;
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copy_tag_to_vorbis_comment(sd);
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vorbis_analysis_headerout(&(sd->vd), &(sd->vc), &(sd->header_main),
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&(sd->header_comments),
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&(sd->header_codebooks));
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ogg_stream_packetin(&(sd->os), &(sd->header_main));
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ogg_stream_packetin(&(sd->os), &(sd->header_comments));
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ogg_stream_packetin(&(sd->os), &(sd->header_codebooks));
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while (ogg_stream_flush(&(sd->os), &(sd->og))) {
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if (write_page(sd) < 0) {
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close_shout_conn(sd);
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return;
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}
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}
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sd->tag_to_send = 0;
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@ -558,12 +437,7 @@ static void send_metadata(struct shout_data * sd)
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static int my_shout_play(struct audio_output *audio_output,
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const char *chunk, size_t size)
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{
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unsigned int i;
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int j;
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struct shout_data *sd = (struct shout_data *) audio_output->data;
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float **vorbbuf;
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unsigned int samples;
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int bytes = sd->audio_format.bits / 8;
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int status;
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if (!sd->timer->started)
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@ -585,29 +459,7 @@ static int my_shout_play(struct audio_output *audio_output,
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}
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}
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samples = size / (bytes * sd->audio_format.channels);
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/* this is for only 16-bit audio */
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vorbbuf = vorbis_analysis_buffer(&(sd->vd), samples);
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for (i = 0; i < samples; i++) {
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for (j = 0; j < sd->audio_format.channels; j++) {
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vorbbuf[j][i] = (*((const mpd_sint16 *) chunk)) / 32768.0;
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chunk += bytes;
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}
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}
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vorbis_analysis_wrote(&(sd->vd), samples);
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while (1 == vorbis_analysis_blockout(&(sd->vd), &(sd->vb))) {
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vorbis_analysis(&(sd->vb), NULL);
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vorbis_bitrate_addblock(&(sd->vb));
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while (vorbis_bitrate_flushpacket(&(sd->vd), &(sd->op))) {
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ogg_stream_packetin(&(sd->os), &(sd->op));
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}
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}
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shout_ogg_encoder_encode(sd, chunk, size);
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while (ogg_stream_pageout(&(sd->os), &(sd->og)) != 0) {
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if (write_page(sd) < 0) {
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@ -63,6 +63,21 @@ struct shout_data {
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struct audio_format audio_format;
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};
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int write_page(struct shout_data *sd);
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void copy_tag_to_vorbis_comment(struct shout_data *sd);
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void send_ogg_vorbis_header(struct shout_data *sd);
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void shout_ogg_encoder_clear_encoder(struct shout_data *sd);
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int init_encoder(struct shout_data *sd);
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int shout_ogg_encoder_send_metadata(struct shout_data * sd);
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void shout_ogg_encoder_encode(struct shout_data *sd,
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const char *chunk, size_t len);
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#endif
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#endif
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@ -0,0 +1,190 @@
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/* the Music Player Daemon (MPD)
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* Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
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* This project's homepage is: http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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#include "audioOutput_shout.h"
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#ifdef HAVE_SHOUT
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#include "../utils.h"
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static void add_tag(struct shout_data *sd, const char *name, char *value)
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{
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if (value) {
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union const_hack u;
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u.in = name;
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vorbis_comment_add_tag(&(sd->vc), u.out, value);
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}
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}
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void copy_tag_to_vorbis_comment(struct shout_data *sd)
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{
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if (sd->tag) {
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int i;
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for (i = 0; i < sd->tag->numOfItems; i++) {
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switch (sd->tag->items[i]->type) {
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case TAG_ITEM_ARTIST:
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add_tag(sd, "ARTIST", sd->tag->items[i]->value);
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break;
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case TAG_ITEM_ALBUM:
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add_tag(sd, "ALBUM", sd->tag->items[i]->value);
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break;
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case TAG_ITEM_TITLE:
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add_tag(sd, "TITLE", sd->tag->items[i]->value);
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break;
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default:
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break;
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}
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}
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}
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}
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void send_ogg_vorbis_header(struct shout_data *sd)
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{
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vorbis_analysis_headerout(&(sd->vd), &(sd->vc), &(sd->header_main),
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&(sd->header_comments),
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&(sd->header_codebooks));
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ogg_stream_packetin(&(sd->os), &(sd->header_main));
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ogg_stream_packetin(&(sd->os), &(sd->header_comments));
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ogg_stream_packetin(&(sd->os), &(sd->header_codebooks));
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}
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static void finish_encoder(struct shout_data *sd)
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{
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vorbis_analysis_wrote(&sd->vd, 0);
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while (vorbis_analysis_blockout(&sd->vd, &sd->vb) == 1) {
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vorbis_analysis(&sd->vb, NULL);
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vorbis_bitrate_addblock(&sd->vb);
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while (vorbis_bitrate_flushpacket(&sd->vd, &sd->op)) {
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ogg_stream_packetin(&sd->os, &sd->op);
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}
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}
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}
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static int flush_encoder(struct shout_data *sd)
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{
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return (ogg_stream_pageout(&sd->os, &sd->og) > 0);
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}
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void shout_ogg_encoder_clear_encoder(struct shout_data *sd)
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{
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finish_encoder(sd);
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while (1 == flush_encoder(sd)) {
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if (!sd->shout_error)
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write_page(sd);
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}
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vorbis_comment_clear(&sd->vc);
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ogg_stream_clear(&sd->os);
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vorbis_block_clear(&sd->vb);
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vorbis_dsp_clear(&sd->vd);
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vorbis_info_clear(&sd->vi);
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}
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int init_encoder(struct shout_data *sd)
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{
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vorbis_info_init(&(sd->vi));
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if (sd->quality >= -1.0) {
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if (0 != vorbis_encode_init_vbr(&(sd->vi),
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sd->audio_format.channels,
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sd->audio_format.sampleRate,
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sd->quality * 0.1)) {
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ERROR("problem setting up vorbis encoder for shout\n");
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vorbis_info_clear(&(sd->vi));
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return -1;
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}
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} else {
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if (0 != vorbis_encode_init(&(sd->vi),
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sd->audio_format.channels,
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sd->audio_format.sampleRate, -1.0,
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sd->bitrate * 1000, -1.0)) {
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ERROR("problem setting up vorbis encoder for shout\n");
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vorbis_info_clear(&(sd->vi));
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return -1;
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}
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}
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vorbis_analysis_init(&(sd->vd), &(sd->vi));
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vorbis_block_init(&(sd->vd), &(sd->vb));
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ogg_stream_init(&(sd->os), rand());
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vorbis_comment_init(&(sd->vc));
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return 0;
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}
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int shout_ogg_encoder_send_metadata(struct shout_data * sd)
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{
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shout_ogg_encoder_clear_encoder(sd);
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if (init_encoder(sd) < 0)
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return 0;
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copy_tag_to_vorbis_comment(sd);
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vorbis_analysis_headerout(&(sd->vd), &(sd->vc), &(sd->header_main),
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&(sd->header_comments),
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&(sd->header_codebooks));
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ogg_stream_packetin(&(sd->os), &(sd->header_main));
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ogg_stream_packetin(&(sd->os), &(sd->header_comments));
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ogg_stream_packetin(&(sd->os), &(sd->header_codebooks));
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while (ogg_stream_flush(&sd->os, &sd->og)) {
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if (write_page(sd) < 0)
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return -1;
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}
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return 0;
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}
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void shout_ogg_encoder_encode(struct shout_data *sd,
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const char *chunk, size_t size)
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{
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unsigned int i;
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int j;
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float **vorbbuf;
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unsigned int samples;
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int bytes = sd->audio_format.bits / 8;
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samples = size / (bytes * sd->audio_format.channels);
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vorbbuf = vorbis_analysis_buffer(&(sd->vd), samples);
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/* this is for only 16-bit audio */
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for (i = 0; i < samples; i++) {
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for (j = 0; j < sd->audio_format.channels; j++) {
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vorbbuf[j][i] = (*((const mpd_sint16 *) chunk)) / 32768.0;
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chunk += bytes;
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}
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}
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vorbis_analysis_wrote(&(sd->vd), samples);
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while (1 == vorbis_analysis_blockout(&(sd->vd), &(sd->vb))) {
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vorbis_analysis(&(sd->vb), NULL);
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vorbis_bitrate_addblock(&(sd->vb));
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while (vorbis_bitrate_flushpacket(&(sd->vd), &(sd->op))) {
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ogg_stream_packetin(&(sd->os), &(sd->op));
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}
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}
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}
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#endif
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