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mirror of https://git.ffmpeg.org/ffmpeg.git synced 2024-12-24 00:02:52 +00:00
ffmpeg/libavdevice/openal-dec.c
Michael Niedermayer e37f161e66 Merge remote-tracking branch 'qatar/master'
* qatar/master: (71 commits)
  movenc: Allow writing to a non-seekable output if using empty moov
  movenc: Support adding isml (smooth streaming live) metadata
  libavcodec: Don't crash in avcodec_encode_audio if time_base isn't set
  sunrast: Document the different Sun Raster file format types.
  sunrast: Add a check for experimental type.
  libspeexenc: use AVSampleFormat instead of deprecated/removed SampleFormat
  lavf: remove disabled FF_API_SET_PTS_INFO cruft
  lavf: remove disabled FF_API_OLD_INTERRUPT_CB cruft
  lavf: remove disabled FF_API_REORDER_PRIVATE cruft
  lavf: remove disabled FF_API_SEEK_PUBLIC cruft
  lavf: remove disabled FF_API_STREAM_COPY cruft
  lavf: remove disabled FF_API_PRELOAD cruft
  lavf: remove disabled FF_API_NEW_STREAM cruft
  lavf: remove disabled FF_API_RTSP_URL_OPTIONS cruft
  lavf: remove disabled FF_API_MUXRATE cruft
  lavf: remove disabled FF_API_FILESIZE cruft
  lavf: remove disabled FF_API_TIMESTAMP cruft
  lavf: remove disabled FF_API_LOOP_OUTPUT cruft
  lavf: remove disabled FF_API_LOOP_INPUT cruft
  lavf: remove disabled FF_API_AVSTREAM_QUALITY cruft
  ...

Conflicts:
	doc/APIchanges
	libavcodec/8bps.c
	libavcodec/avcodec.h
	libavcodec/libx264.c
	libavcodec/mjpegbdec.c
	libavcodec/options.c
	libavcodec/sunrast.c
	libavcodec/utils.c
	libavcodec/version.h
	libavcodec/x86/h264_deblock.asm
	libavdevice/libdc1394.c
	libavdevice/v4l2.c
	libavformat/avformat.h
	libavformat/avio.c
	libavformat/avio.h
	libavformat/aviobuf.c
	libavformat/dv.c
	libavformat/mov.c
	libavformat/utils.c
	libavformat/version.h
	libavformat/wtv.c
	libavutil/Makefile
	libavutil/file.c
	libswscale/x86/input.asm
	libswscale/x86/swscale_mmx.c
	libswscale/x86/swscale_template.c
	tests/ref/lavf/ffm

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-28 07:53:34 +01:00

252 lines
8.2 KiB
C

/*
* Copyright (c) 2011 Jonathan Baldwin
*
* This file is part of FFmpeg.
*
* Permission to use, copy, modify, and/or distribute this software for any
* purpose with or without fee is hereby granted, provided that the above
* copyright notice and this permission notice appear in all copies.
*
* THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES WITH
* REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF MERCHANTABILITY
* AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY SPECIAL, DIRECT,
* INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES WHATSOEVER RESULTING FROM
* LOSS OF USE, DATA OR PROFITS, WHETHER IN AN ACTION OF CONTRACT, NEGLIGENCE OR
* OTHER TORTIOUS ACTION, ARISING OUT OF OR IN CONNECTION WITH THE USE OR
* PERFORMANCE OF THIS SOFTWARE.
*/
/**
* @file
* OpenAL 1.1 capture device for libavdevice
**/
#include <AL/al.h>
#include <AL/alc.h>
#include "libavutil/opt.h"
#include "libavformat/internal.h"
#include "avdevice.h"
typedef struct {
AVClass *class;
/** OpenAL capture device context. **/
ALCdevice *device;
/** The number of channels in the captured audio. **/
int channels;
/** The sample rate (in Hz) of the captured audio. **/
int sample_rate;
/** The sample size (in bits) of the captured audio. **/
int sample_size;
/** The OpenAL sample format of the captured audio. **/
ALCenum sample_format;
/** The number of bytes between two consecutive samples of the same channel/component. **/
ALCint sample_step;
/** If true, print a list of capture devices on this system and exit. **/
int list_devices;
} al_data;
typedef struct {
ALCenum al_fmt;
enum CodecID codec_id;
int channels;
} al_format_info;
#define LOWEST_AL_FORMAT FFMIN(FFMIN(AL_FORMAT_MONO8,AL_FORMAT_MONO16),FFMIN(AL_FORMAT_STEREO8,AL_FORMAT_STEREO16))
/**
* Get information about an AL_FORMAT value.
* @param al_fmt the AL_FORMAT value to find information about.
* @return A pointer to a structure containing information about the AL_FORMAT value.
*/
static inline al_format_info* get_al_format_info(ALCenum al_fmt)
{
static al_format_info info_table[] = {
[AL_FORMAT_MONO8-LOWEST_AL_FORMAT] = {AL_FORMAT_MONO8, CODEC_ID_PCM_U8, 1},
[AL_FORMAT_MONO16-LOWEST_AL_FORMAT] = {AL_FORMAT_MONO16, AV_NE (CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE), 1},
[AL_FORMAT_STEREO8-LOWEST_AL_FORMAT] = {AL_FORMAT_STEREO8, CODEC_ID_PCM_U8, 2},
[AL_FORMAT_STEREO16-LOWEST_AL_FORMAT] = {AL_FORMAT_STEREO16, AV_NE (CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE), 2},
};
return &info_table[al_fmt-LOWEST_AL_FORMAT];
}
/**
* Get the OpenAL error code, translated into an av/errno error code.
* @param device The ALC device to check for errors.
* @param error_msg_ret A pointer to a char* in which to return the error message, or NULL if desired.
* @return The error code, or 0 if there is no error.
*/
static inline int al_get_error(ALCdevice *device, const char** error_msg_ret)
{
ALCenum error = alcGetError(device);
if (error_msg_ret)
*error_msg_ret = (const char*) alcGetString(device, error);
switch (error) {
case ALC_NO_ERROR:
return 0;
case ALC_INVALID_DEVICE:
return AVERROR(ENODEV);
break;
case ALC_INVALID_CONTEXT:
case ALC_INVALID_ENUM:
case ALC_INVALID_VALUE:
return AVERROR(EINVAL);
break;
case ALC_OUT_OF_MEMORY:
return AVERROR(ENOMEM);
break;
default:
return AVERROR(EIO);
}
}
/**
* Print out a list of OpenAL capture devices on this system.
*/
static inline void print_al_capture_devices(void *log_ctx)
{
const char *devices;
if (!(devices = alcGetString(NULL, ALC_CAPTURE_DEVICE_SPECIFIER)))
return;
av_log(log_ctx, AV_LOG_INFO, "List of OpenAL capture devices on this system:\n");
for (; *devices != '\0'; devices += strlen(devices) + 1)
av_log(log_ctx, AV_LOG_INFO, " %s\n", devices);
}
static int read_header(AVFormatContext *ctx)
{
al_data *ad = ctx->priv_data;
static const ALCenum sample_formats[2][2] = {
{ AL_FORMAT_MONO8, AL_FORMAT_STEREO8 },
{ AL_FORMAT_MONO16, AL_FORMAT_STEREO16 }
};
int error = 0;
const char *error_msg;
AVStream *st = NULL;
AVCodecContext *codec = NULL;
if (ad->list_devices) {
print_al_capture_devices(ctx);
return AVERROR_EXIT;
}
ad->sample_format = sample_formats[ad->sample_size/8-1][ad->channels-1];
/* Open device for capture */
ad->device =
alcCaptureOpenDevice(ctx->filename[0] ? ctx->filename : NULL,
ad->sample_rate,
ad->sample_format,
ad->sample_rate); /* Maximum 1 second of sample data to be read at once */
if (error = al_get_error(ad->device, &error_msg)) goto fail;
/* Create stream */
if (!(st = avformat_new_stream(ctx, NULL))) {
error = AVERROR(ENOMEM);
goto fail;
}
/* We work in microseconds */
avpriv_set_pts_info(st, 64, 1, 1000000);
/* Set codec parameters */
codec = st->codec;
codec->codec_type = AVMEDIA_TYPE_AUDIO;
codec->sample_rate = ad->sample_rate;
codec->channels = get_al_format_info(ad->sample_format)->channels;
codec->codec_id = get_al_format_info(ad->sample_format)->codec_id;
/* This is needed to read the audio data */
ad->sample_step = (av_get_bits_per_sample(get_al_format_info(ad->sample_format)->codec_id) *
get_al_format_info(ad->sample_format)->channels) / 8;
/* Finally, start the capture process */
alcCaptureStart(ad->device);
return 0;
fail:
/* Handle failure */
if (ad->device)
alcCaptureCloseDevice(ad->device);
if (error_msg)
av_log(ctx, AV_LOG_ERROR, "Cannot open device: %s\n", error_msg);
return error;
}
static int read_packet(AVFormatContext* ctx, AVPacket *pkt)
{
al_data *ad = ctx->priv_data;
int error=0;
const char *error_msg;
ALCint nb_samples;
/* Get number of samples available */
alcGetIntegerv(ad->device, ALC_CAPTURE_SAMPLES, (ALCsizei) sizeof(ALCint), &nb_samples);
if (error = al_get_error(ad->device, &error_msg)) goto fail;
/* Create a packet of appropriate size */
av_new_packet(pkt, nb_samples*ad->sample_step);
pkt->pts = av_gettime();
/* Fill the packet with the available samples */
alcCaptureSamples(ad->device, pkt->data, nb_samples);
if (error = al_get_error(ad->device, &error_msg)) goto fail;
return pkt->size;
fail:
/* Handle failure */
if (pkt->data)
av_destruct_packet(pkt);
if (error_msg)
av_log(ctx, AV_LOG_ERROR, "Error: %s\n", error_msg);
return error;
}
static int read_close(AVFormatContext* ctx)
{
al_data *ad = ctx->priv_data;
if (ad->device) {
alcCaptureStop(ad->device);
alcCaptureCloseDevice(ad->device);
}
return 0;
}
#define OFFSET(x) offsetof(al_data, x)
static const AVOption options[] = {
{"channels", "set number of channels", OFFSET(channels), AV_OPT_TYPE_INT, {.dbl=2}, 1, 2, AV_OPT_FLAG_DECODING_PARAM },
{"sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.dbl=44100}, 1, 192000, AV_OPT_FLAG_DECODING_PARAM },
{"sample_size", "set sample size", OFFSET(sample_size), AV_OPT_TYPE_INT, {.dbl=16}, 8, 16, AV_OPT_FLAG_DECODING_PARAM },
{"list_devices", "list available devices", OFFSET(list_devices), AV_OPT_TYPE_INT, {.dbl=0}, 0, 1, AV_OPT_FLAG_DECODING_PARAM, "list_devices" },
{"true", "", 0, AV_OPT_TYPE_CONST, {.dbl=1}, 0, 0, AV_OPT_FLAG_DECODING_PARAM, "list_devices" },
{"false", "", 0, AV_OPT_TYPE_CONST, {.dbl=0}, 0, 0, AV_OPT_FLAG_DECODING_PARAM, "list_devices" },
{NULL},
};
static const AVClass class = {
.class_name = "openal",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT
};
AVInputFormat ff_openal_demuxer = {
.name = "openal",
.long_name = NULL_IF_CONFIG_SMALL("OpenAL audio capture device"),
.priv_data_size = sizeof(al_data),
.read_probe = NULL,
.read_header = read_header,
.read_packet = read_packet,
.read_close = read_close,
.flags = AVFMT_NOFILE,
.priv_class = &class
};