Mercurial > mplayer.hg
view libao2/ao_arts.c @ 33671:a460339acfdf
Fix volume and balance bug.
Changing the volume changed the balance as well,
because the calculation for the balance was wrong.
Additionally, use macro FFMAX() and replace
identical code by a call to existing code.
author | ib |
---|---|
date | Tue, 28 Jun 2011 17:52:30 +0000 |
parents | 32725ca88fed |
children |
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/* * aRts audio output driver for MPlayer * * copyright (c) 2002 Michele Balistreri <brain87@gmx.net> * * This file is part of MPlayer. * * MPlayer is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * MPlayer is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License along * with MPlayer; if not, write to the Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ #include <artsc.h> #include <stdio.h> #include "config.h" #include "audio_out.h" #include "audio_out_internal.h" #include "libaf/af_format.h" #include "mp_msg.h" #include "help_mp.h" #define OBTAIN_BITRATE(a) (((a != AF_FORMAT_U8) && (a != AF_FORMAT_S8)) ? 16 : 8) /* Feel free to experiment with the following values: */ #define ARTS_PACKETS 10 /* Number of audio packets */ #define ARTS_PACKET_SIZE_LOG2 11 /* Log2 of audio packet size */ static arts_stream_t stream; static const ao_info_t info = { "aRts audio output", "arts", "Michele Balistreri <brain87@gmx.net>", "" }; LIBAO_EXTERN(arts) static int control(int cmd, void *arg) { return CONTROL_UNKNOWN; } static int init(int rate_hz, int channels, int format, int flags) { int err; int frag_spec; if( (err=arts_init()) ) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ARTS_CantInit, arts_error_text(err)); return 0; } mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_ServerConnect); /* * arts supports 8bit unsigned and 16bit signed sample formats * (16bit apparently in little endian format, even in the case * when artsd runs on a big endian cpu). * * Unsupported formats are translated to one of these two formats * using mplayer's audio filters. */ switch (format) { case AF_FORMAT_U8: case AF_FORMAT_S8: format = AF_FORMAT_U8; break; default: format = AF_FORMAT_S16_LE; /* artsd always expects little endian?*/ break; } ao_data.format = format; ao_data.channels = channels; ao_data.samplerate = rate_hz; ao_data.bps = (rate_hz*channels); if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8) ao_data.bps*=2; stream=arts_play_stream(rate_hz, OBTAIN_BITRATE(format), channels, "MPlayer"); if(stream == NULL) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ARTS_CantOpenStream); arts_free(); return 0; } /* Set the stream to blocking: it will not block anyway, but it seems */ /* to be working better */ arts_stream_set(stream, ARTS_P_BLOCKING, 1); frag_spec = ARTS_PACKET_SIZE_LOG2 | ARTS_PACKETS << 16; arts_stream_set(stream, ARTS_P_PACKET_SETTINGS, frag_spec); ao_data.buffersize = arts_stream_get(stream, ARTS_P_BUFFER_SIZE); mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_StreamOpen); mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_BufferSize, ao_data.buffersize); mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_BufferSize, arts_stream_get(stream, ARTS_P_PACKET_SIZE)); return 1; } static void uninit(int immed) { arts_close_stream(stream); arts_free(); } static int play(void* data,int len,int flags) { return arts_write(stream, data, len); } static void audio_pause(void) { } static void audio_resume(void) { } static void reset(void) { } static int get_space(void) { return arts_stream_get(stream, ARTS_P_BUFFER_SPACE); } static float get_delay(void) { return ((float) (ao_data.buffersize - arts_stream_get(stream, ARTS_P_BUFFER_SPACE))) / ((float) ao_data.bps); }