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/**
* @file reporting.h - Definition of functions whose represents an interface
* to report errors.
* <p>
* Copyright (C) 2006 Nokia Corporation
* <p>
* Contact: Eduardo Bezerra Valentin <eduardo.valentin@indt.org.br>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
* */
#ifndef _REPORTING_H
#define _REPORTING_H
#include "dsp-protocol.h"
#ifdef DEBUG
const char *dsp_commands[] = {
"DSP_CMD_NONE", "No command",
"DSP_CMD_INIT", "Informs the DSP that the following data is"
"about initialisation",
"DSP_CMD_SET_PARAMS", "Informs the DSP that the following "
"data is parameters",
"DSP_CMD_DATA_WRITE", "Informs the DSP that the following "
"data is general data (compressed " "or raw audio or video)",
"DSP_CMD_PLAY", "Starts audio or video playback or recording",
"DSP_CMD_PAUSE", "Pauses playback",
"DSP_CMD_STOP", "Stops playback",
"DSP_CMD_SET_VOLUME", "Informs the DSP that the following "
"data is volume",
"DSP_CMD_STATE", "Requests from the DSP to send information"
" about current task node state",
"DSP_CMD_SET_TIME", "Informs the DSP that the following data"
" is about setting the current" " presentation time",
"DSP_CMD_GET_TIME", "Informs the DSP that the ARM queries the"
" current presentation time",
"ERROR", "This is unused!!!!!",
"DSP_CMD_SET_POSTPROC", "Informs the DSP that the following data"
" is about setting video post-processing " "parameters",
"DSP_CMD_SET_PANNING", "Informs the DSP that the following data "
"is about setting the panning",
"DSP_CMD_DISCONT", "Informs the DSP about discontinuity in the "
"audio stream",
"DSP_CMD_MUTE", "Mutes the audio playback",
"DSP_CMD_UNMUTE", "Unmutes the audio playback",
"ERROR", "This is unused!!!!!",
"ERROR", "This is unused!!!!!",
"ERROR", "This is unused!!!!!",
"DSP_CMD_CLOSE", "Closes the task node"
};
const char *dsp_return_values[] = {
"DSP_NONE", "Error. This isn't a valid return value",
"DSP_OK", "Operation successful",
"DSP_ERROR_CMD", "Unrecognised or unsupported command value",
"DSP_ERROR_FMT", "Unrecognised or unsupported audio format value",
"DSP_ERROR_RATE", "Unrecognised or unsupported sampling rate value",
"DSP_ERROR_CHANNELS", "Unrecognised or unsupported number of channels",
"DSP_ERROR_DS_id", "Destination/source stream id out of range",
"DSP_ERROR_MEMORY", "Insufficient memory to perform requested action",
"DSP_ERROR_GENERAL", "Unspecified error",
"DSP_ERROR_STREAM", "Error in stream (audio or video)",
"DSP_ERROR_STATE", "Unexpected task node state",
"DSP_ERROR_SYNC", "Error in synchronisation:"
"For MP3 – synchronisation marker not found"
};
const char *dsp_states[] = {
"STATE_INITIALISED", "Initialised",
"STATE_PLAYING", "Playing/recording",
"STATE_STOPPED", "Stopped",
"STATE_PAUSED", "Paused",
"STATE_UNINITIALISED", "Not initialised",
"STATE_RESET", "Reseted",
"STATE_MUTED", "Muted"
};
const char *dsp_rates[] = {
"SAMPLE_RATE_96KHZ", "96KHz sampling rate",
"SAMPLE_RATE_88_2KHZ", "88.2KHz sampling rate",
"SAMPLE_RATE_64KHZ", "64KHz sampling rate",
"SAMPLE_RATE_48KHZ", "48KHz sampling rate",
"SAMPLE_RATE_44_1KHZ", "44.1KHz sampling rate",
"SAMPLE_RATE_32KHZ", "32KHz sampling rate",
"SAMPLE_RATE_24KHZ", "24KHz sampling rate",
"SAMPLE_RATE_22_05KHZ", "22.05KHz sampling rate",
"SAMPLE_RATE_16KHZ", "16KHz sampling rate",
"SAMPLE_RATE_12KHZ", "12KHz sampling rate",
"SAMPLE_RATE_11_025KHZ", "11.025KHz sampling rate",
"SAMPLE_RATE_8KHZ", "8KHz sampling rate",
"SAMPLE_RATE_5_5125KHZ", "5.5125Khz sampling rate"
};
const char *dsp_channels[] = {
"0--", "Error - No channel!",
"CHANNELS_1", "One channel (mono)",
"CHANNELS_2", "Two channels (stereo)"
};
const char *dsp_audio_fmt[] = {
"0", "Error No format!!!",
"DSP_AFMT_U8", "Unsigned 8 bits per sample PCM",
"DSP_AFMT_S16_LE", "Signed 16 bits per sample PCM, little endian",
"DSP_AFMT_S16_BE", "Signed 16 bits per sample PCM, big endian",
"DSP_AFMT_S8", "Signed 8 bits per sample PCM",
"DSP_AFMT_U16_LE", "Unsigned 16 bits per sample PCM, little endian",
"DSP_AFMT_U16_BE", "Unsigned 16 bits per sample PCM, big endian",
"DSP_AFMT_ALAW", "A-law encoded PCM",
"DSP_AFMT_ULAW", "μ-Law encoded PCM",
"DSP_AFMT_MP3", "MP3 stream",
"DSP_AFMT_AAC", "AAC stream",
"DSP_AFMT_AMR", "AMR stream",
"DSP_AFMT_MP2", "MP2 stream",
"DSP_AFMT_ILBC", "iLBC stream",
"DSP_AFMT_G729", "G.729 stream"
};
#define ARRAY_SIZE(ary) (sizeof(ary)/sizeof(ary[0]))
#define report_table(mens,name,value,table)\
do{\
if ((unsigned)value >= ARRAY_SIZE(table))\
DPRINT("%s: %d isnt a valid %s value\n",mens,\
value,name);\
else\
DPRINT("%s: [%d|%s] - %s\n", mens, value, \
table[value * 2], \
table[value * 2 + 1]);\
}while(0)
#define report_command(m,v) report_table(m,"command",v,\
/*20,*/dsp_commands)
#define report_return_value(m,v) report_table(m,"return",v,\
/*11,*/dsp_return_values)
#define report_state(m,v) report_table(m,"state",v,\
/*6,*/dsp_states)
#define report_sample_rate(m,v) report_table(m,"sample rate",v,\
/*12,*/dsp_rates)
#define report_number_channels(m,v) report_table(m,"number of channels",v,\
/*2,*/dsp_channels)
#define report_audio_fmt(m,v) report_table(m,"audio format",v,\
/*14,*/dsp_audio_fmt)
#define report_dsp_protocol(m,dp)\
do{\
DPRINT("%s:\n"\
"fd: %d\n"\
"stream_id: %d\n"\
"bridge_buffer_size: %d\n"\
"mmap_buffer_size: %d\n"\
"mmap_buffer: %p\n",\
m,\
dp->fd,\
dp->stream_id,\
dp->bridge_buffer_size,\
dp->mmap_buffer_size,\
dp->mmap_buffer);\
report_state("state", dp->state);\
}while(0)
#define report_audio_status_info(m, asi)\
do{\
DPRINT("%s\n", m);\
DPRINT("***** Audio status info *****\n");\
report_command("\tdsp_cmd", asi.dsp_cmd);\
DPRINT("\tstream_id: %d\n", asi.stream_id);\
DPRINT("\tds_stream_id: %d\n", asi.ds_stream_id);\
DPRINT("\tbridge_buffer_size: %d\n", asi.bridge_buffer_size);\
DPRINT("\tmmap_buffer_size: %d\n", asi.mmap_buffer_size);\
report_state("\tstatus", asi.status);\
DPRINT("\tnum_frames: %d\n", asi.num_frames);\
report_sample_rate("\tsample_rate", asi.sample_rate);\
report_number_channels("\tnumber_channels", \
asi.number_channels);\
DPRINT("\tvol_scale: %d\n", asi.vol_scale);\
DPRINT("\tvol_power2: %d\n", asi.vol_power2);\
DPRINT("\tleft_gain: %d\n", asi.left_gain);\
DPRINT("\tright_gain: %d\n", asi.right_gain);\
report_audio_fmt("\tdsp_audio_fmt", asi.dsp_audio_fmt);\
}while(0)
#define report_audio_init_status(m, ais)\
do{\
DPRINT("%s\n", m);\
DPRINT("***** Audio init status *****\n");\
report_command("\tdsp_cmd", ais.dsp_cmd);\
DPRINT("\tstream_id: %d\n", ais.stream_id);\
DPRINT("\tbridge_buffer_size: %d\n", ais.bridge_buffer_size);\
DPRINT("\tmmap_buffer_size: %d\n", ais.mmap_buffer_size);\
report_return_value("\tinit_status", ais.init_status);\
}while(0)
#define report_audio_params(m,ap)\
do{\
DPRINT("%s\n",m);\
DPRINT("**** Audio parameters *****\n");\
report_command("\tdsp_cmd",ap.dsp_cmd);\
report_audio_fmt("\taudio_format", ap.dsp_audio_fmt);\
report_sample_rate("\tsample_rate", ap.sample_rate);\
DPRINT("Number of channels %d\n", ap.number_channels);\
DPRINT("ds_stream_id: %d\n", ap.ds_stream_id);\
DPRINT("stream_priority: %d\n", ap.stream_priority);\
}while(0)
#define report_speech_params(m,sp)\
do{\
DPRINT("%s\n",m);\
DPRINT("**** Speech parameters *****\n");\
DPRINT("\tdsp_cmd 0x%x\n",sp.dsp_cmd);\
report_audio_fmt("\taudio_format", sp.audio_fmt);\
report_sample_rate("\tsample_rate", sp.sample_rate);\
DPRINT("ds_stream_id: %d\n", sp.ds_stream_id);\
DPRINT("stream_priority: %d\n", sp.stream_priority);\
DPRINT("frame_size: %d\n", sp.frame_size);\
}while(0)
#else
#define report_command(m,c)
#define report_return_value(m,c)
#define report_state(m,c)
#define report_sample_rate(m,sr)
#define report_number_channel(m,nc)
#define report_audio_fmt(m,af)
#define report_dsp_protocol(m,dp)
#define report_audio_status_info(m,asi)
#define report_audio_init_status(m,ais)
#define report_audio_params(m,ap)
#define report_speech_params(m,ap)
#endif /* _DEBUG */
#endif /* _REPORTING_H */
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