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Update chan_console to natively use a 16 kHz sample rate. If it is talking

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to an 8 kHz endpoint, then codec_resample will automatically be used to properly
resample the audio before sending it to/from chan_console.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
2007-12-31 21:33:45 +00:00
parent 21cb767db7
commit 067fcc2a03
1 changed files with 12 additions and 20 deletions
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32
channels/chan_console.c
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@@ -67,28 +67,22 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
/*! /*!
* \brief The sample rate to request from PortAudio * \brief The sample rate to request from PortAudio
* *
* \note This should be changed to 16000 once there is a translator for going * \todo Make this optional. If this is only going to talk to 8 kHz endpoints,
* between SLINEAR and SLINEAR16. Making it a configuration parameter * then it makes sense to use 8 kHz natively.
* would be even better, but 16 kHz should be the default.
*
* \note If this changes, NUM_SAMPLES will need to change, as well.
*/ */
#define SAMPLE_RATE 8000 #define SAMPLE_RATE 16000
/*! /*!
* \brief The number of samples to configure the portaudio stream for * \brief The number of samples to configure the portaudio stream for
* *
* 160 samples (20 ms) is the most common frame size in Asterisk. So, the code * 320 samples (20 ms) is the most common frame size in Asterisk. So, the code
* in this module reads 160 sample frames from the portaudio stream and queues * in this module reads 320 sample frames from the portaudio stream and queues
* them up on the Asterisk channel. Frames of any sizes can be written to a * them up on the Asterisk channel. Frames of any size can be written to a
* portaudio stream, but the portaudio documentation does say that for high * portaudio stream, but the portaudio documentation does say that for high
* performance applications, the data should be written to Pa_WriteStream in * performance applications, the data should be written to Pa_WriteStream in
* the same size as what is used to initialize the stream. * the same size as what is used to initialize the stream.
*
* \note This will need to be dynamic once the sample rate can be something
* other than 8 kHz.
*/ */
#define NUM_SAMPLES 160 #define NUM_SAMPLES 320
/*! \brief Mono Input */ /*! \brief Mono Input */
#define INPUT_CHANNELS 1 #define INPUT_CHANNELS 1
@@ -198,10 +192,8 @@ static int console_fixup(struct ast_channel *oldchan, struct ast_channel *newcha
/*! /*!
* \brief Formats natively supported by this module. * \brief Formats natively supported by this module.
*
* \note Once 16 kHz is supported, AST_FORMAT_SLINEAR16 needs to be added.
*/ */
#define SUPPORTED_FORMATS ( AST_FORMAT_SLINEAR ) #define SUPPORTED_FORMATS ( AST_FORMAT_SLINEAR16 )
static const struct ast_channel_tech console_tech = { static const struct ast_channel_tech console_tech = {
.type = "Console", .type = "Console",
@@ -243,7 +235,7 @@ static void *stream_monitor(void *data)
PaError res; PaError res;
struct ast_frame f = { struct ast_frame f = {
.frametype = AST_FRAME_VOICE, .frametype = AST_FRAME_VOICE,
.subclass = AST_FORMAT_SLINEAR, .subclass = AST_FORMAT_SLINEAR16,
.src = "console_stream_monitor", .src = "console_stream_monitor",
.data = buf, .data = buf,
.datalen = sizeof(buf), .datalen = sizeof(buf),
@@ -335,9 +327,9 @@ static struct ast_channel *console_new(struct console_pvt *pvt, const char *ext,
} }
chan->tech = &console_tech; chan->tech = &console_tech;
chan->nativeformats = AST_FORMAT_SLINEAR; chan->nativeformats = AST_FORMAT_SLINEAR16;
chan->readformat = AST_FORMAT_SLINEAR; chan->readformat = AST_FORMAT_SLINEAR16;
chan->writeformat = AST_FORMAT_SLINEAR; chan->writeformat = AST_FORMAT_SLINEAR16;
chan->tech_pvt = pvt; chan->tech_pvt = pvt;
pvt->owner = chan; pvt->owner = chan;