mirror of https://github.com/mpv-player/mpv
137 lines
2.9 KiB
C
137 lines
2.9 KiB
C
/*
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* ao_arts - aRts audio output driver for MPlayer
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*
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* Michele Balistreri <brain87@gmx.net>
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*
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* This driver is distribuited under terms of GPL
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*
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*/
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#include <artsc.h>
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#include <stdio.h>
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#include "audio_out.h"
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#include "audio_out_internal.h"
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#include "afmt.h"
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#include "../config.h"
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#include "../mp_msg.h"
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#define OBTAIN_BITRATE(a) (((a != AFMT_U8) && (a != AFMT_S8)) ? 16 : 8)
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/* Feel free to experiment with the following values: */
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#define ARTS_PACKETS 10 /* Number of audio packets */
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#define ARTS_PACKET_SIZE_LOG2 11 /* Log2 of audio packet size */
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static arts_stream_t stream;
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static ao_info_t info =
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{
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"aRts audio output",
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"arts",
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"Michele Balistreri <brain87@gmx.net>",
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""
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};
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LIBAO_EXTERN(arts)
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static int control(int cmd, int arg)
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{
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return(CONTROL_UNKNOWN);
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}
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static int init(int rate_hz, int channels, int format, int flags)
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{
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int err;
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int frag_spec;
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if( (err=arts_init()) ) {
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mp_msg(MSGT_AO, MSGL_ERR, "AO: [arts] %s\n", arts_error_text(err));
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return 0;
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}
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mp_msg(MSGT_AO, MSGL_INFO, "AO: [arts] Connected to sound server\n");
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/*
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* arts supports 8bit unsigned and 16bit signed sample formats
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* (16bit apparently in little endian format, even in the case
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* when artsd runs on a big endian cpu).
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*
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* Unsupported formats are translated to one of these two formats
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* using mplayer's audio filters.
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*/
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switch (format) {
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case AFMT_U8:
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case AFMT_S8:
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format = AFMT_U8;
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break;
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default:
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format = AFMT_S16_LE; /* artsd always expects little endian?*/
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break;
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}
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ao_data.format = format;
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ao_data.channels = channels;
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ao_data.samplerate = rate_hz;
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ao_data.bps = (rate_hz*channels);
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if(format != AFMT_U8 && format != AFMT_S8)
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ao_data.bps*=2;
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stream=arts_play_stream(rate_hz, OBTAIN_BITRATE(format), channels, "Mplayer");
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if(stream == NULL) {
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mp_msg(MSGT_AO, MSGL_ERR, "AO: [arts] Unable to open a stream\n");
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arts_free();
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return 0;
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}
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/* Set the stream to blocking: it will not block anyway, but it seems */
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/* to be working better */
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arts_stream_set(stream, ARTS_P_BLOCKING, 1);
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frag_spec = ARTS_PACKET_SIZE_LOG2 | ARTS_PACKETS << 16;
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arts_stream_set(stream, ARTS_P_PACKET_SETTINGS, frag_spec);
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ao_data.buffersize = arts_stream_get(stream, ARTS_P_BUFFER_SIZE);
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mp_msg(MSGT_AO, MSGL_INFO, "AO: [arts] Stream opened\n");
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mp_msg(MSGT_AO, MSGL_INFO,"AO: [arts] buffer size: %d\n",
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ao_data.buffersize);
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mp_msg(MSGT_AO, MSGL_INFO,"AO: [arts] packet size: %d\n",
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arts_stream_get(stream, ARTS_P_PACKET_SIZE));
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return 1;
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}
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static void uninit()
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{
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arts_close_stream(stream);
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arts_free();
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}
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static int play(void* data,int len,int flags)
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{
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return arts_write(stream, data, len);
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}
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static void audio_pause()
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{
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}
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static void audio_resume()
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{
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}
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static void reset()
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{
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}
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static int get_space()
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{
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return arts_stream_get(stream, ARTS_P_BUFFER_SPACE);
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}
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static float get_delay()
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{
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return ((float) (ao_data.buffersize - arts_stream_get(stream,
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ARTS_P_BUFFER_SPACE))) / ((float) ao_data.bps);
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}
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