RepoMirrors/ffmpeg
1
0
Fork 0
mirror of https://git.ffmpeg.org/ffmpeg.git synced 2024-12-11 17:55:21 +00:00
Code Issues Projects Releases Wiki Activity
e22ffb3805
ffmpeg/libavdevice/alsa.c
Diego Biurrun d46cd24986 alsa: Coalesce source files after outdev removal
2017-10-13 23:57:59 +02:00

519 lines
16 KiB
C
Raw Blame History

/*
* ALSA input
* Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* ALSA input
* @author Luca Abeni ( lucabe72 email it )
* @author Benoit Fouet ( benoit fouet free fr )
* @author Nicolas George ( nicolas george normalesup org )
*/
#include <alsa/asoundlib.h>
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "libavformat/avformat.h"
#include "libavformat/internal.h"
/* XXX: we make the assumption that the soundcard accepts this format */
/* XXX: find better solution with "preinit" method, needed also in
other formats */
#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
#define ALSA_BUFFER_SIZE_MAX 32768
typedef struct AlsaData {
AVClass *class;
snd_pcm_t *h;
int frame_size; ///< preferred size for reads and writes
int period_size; ///< bytes per sample * channels
int sample_rate; ///< sample rate set by user
int channels; ///< number of channels set by user
void (*reorder_func)(const void *, void *, int);
void *reorder_buf;
int reorder_buf_size; ///< in frames
} AlsaData;
static av_cold snd_pcm_format_t codec_id_to_pcm_format(int codec_id)
{
switch(codec_id) {
case AV_CODEC_ID_PCM_F64LE: return SND_PCM_FORMAT_FLOAT64_LE;
case AV_CODEC_ID_PCM_F64BE: return SND_PCM_FORMAT_FLOAT64_BE;
case AV_CODEC_ID_PCM_F32LE: return SND_PCM_FORMAT_FLOAT_LE;
case AV_CODEC_ID_PCM_F32BE: return SND_PCM_FORMAT_FLOAT_BE;
case AV_CODEC_ID_PCM_S32LE: return SND_PCM_FORMAT_S32_LE;
case AV_CODEC_ID_PCM_S32BE: return SND_PCM_FORMAT_S32_BE;
case AV_CODEC_ID_PCM_U32LE: return SND_PCM_FORMAT_U32_LE;
case AV_CODEC_ID_PCM_U32BE: return SND_PCM_FORMAT_U32_BE;
case AV_CODEC_ID_PCM_S24LE: return SND_PCM_FORMAT_S24_3LE;
case AV_CODEC_ID_PCM_S24BE: return SND_PCM_FORMAT_S24_3BE;
case AV_CODEC_ID_PCM_U24LE: return SND_PCM_FORMAT_U24_3LE;
case AV_CODEC_ID_PCM_U24BE: return SND_PCM_FORMAT_U24_3BE;
case AV_CODEC_ID_PCM_S16LE: return SND_PCM_FORMAT_S16_LE;
case AV_CODEC_ID_PCM_S16BE: return SND_PCM_FORMAT_S16_BE;
case AV_CODEC_ID_PCM_U16LE: return SND_PCM_FORMAT_U16_LE;
case AV_CODEC_ID_PCM_U16BE: return SND_PCM_FORMAT_U16_BE;
case AV_CODEC_ID_PCM_S8: return SND_PCM_FORMAT_S8;
case AV_CODEC_ID_PCM_U8: return SND_PCM_FORMAT_U8;
case AV_CODEC_ID_PCM_MULAW: return SND_PCM_FORMAT_MU_LAW;
case AV_CODEC_ID_PCM_ALAW: return SND_PCM_FORMAT_A_LAW;
default: return SND_PCM_FORMAT_UNKNOWN;
}
}
#define REORDER_OUT_50(NAME, TYPE) \
static void alsa_reorder_ ## NAME ## _out_50(const void *in_v, void *out_v, int n) \
{ \
const TYPE *in = in_v; \
TYPE *out = out_v; \
\
while (n-- > 0) { \
out[0] = in[0]; \
out[1] = in[1]; \
out[2] = in[3]; \
out[3] = in[4]; \
out[4] = in[2]; \
in += 5; \
out += 5; \
} \
}
#define REORDER_OUT_51(NAME, TYPE) \
static void alsa_reorder_ ## NAME ## _out_51(const void *in_v, void *out_v, int n) \
{ \
const TYPE *in = in_v; \
TYPE *out = out_v; \
\
while (n-- > 0) { \
out[0] = in[0]; \
out[1] = in[1]; \
out[2] = in[4]; \
out[3] = in[5]; \
out[4] = in[2]; \
out[5] = in[3]; \
in += 6; \
out += 6; \
} \
}
#define REORDER_OUT_71(NAME, TYPE) \
static void alsa_reorder_ ## NAME ## _out_71(const void *in_v, void *out_v, int n) \
{ \
const TYPE *in = in_v; \
TYPE *out = out_v; \
\
while (n-- > 0) { \
out[0] = in[0]; \
out[1] = in[1]; \
out[2] = in[4]; \
out[3] = in[5]; \
out[4] = in[2]; \
out[5] = in[3]; \
out[6] = in[6]; \
out[7] = in[7]; \
in += 8; \
out += 8; \
} \
}
REORDER_OUT_50(int8, int8_t)
REORDER_OUT_51(int8, int8_t)
REORDER_OUT_71(int8, int8_t)
REORDER_OUT_50(int16, int16_t)
REORDER_OUT_51(int16, int16_t)
REORDER_OUT_71(int16, int16_t)
REORDER_OUT_50(int32, int32_t)
REORDER_OUT_51(int32, int32_t)
REORDER_OUT_71(int32, int32_t)
REORDER_OUT_50(f32, float)
REORDER_OUT_51(f32, float)
REORDER_OUT_71(f32, float)
#define FORMAT_I8 0
#define FORMAT_I16 1
#define FORMAT_I32 2
#define FORMAT_F32 3
#define PICK_REORDER(layout)\
switch(format) {\
case FORMAT_I8: s->reorder_func = alsa_reorder_int8_out_ ##layout; break;\
case FORMAT_I16: s->reorder_func = alsa_reorder_int16_out_ ##layout; break;\
case FORMAT_I32: s->reorder_func = alsa_reorder_int32_out_ ##layout; break;\
case FORMAT_F32: s->reorder_func = alsa_reorder_f32_out_ ##layout; break;\
}
static av_cold int find_reorder_func(AlsaData *s, int codec_id, uint64_t layout, int out)
{
int format;
/* reordering input is not currently supported */
if (!out)
return AVERROR(ENOSYS);
/* reordering is not needed for QUAD or 2_2 layout */
if (layout == AV_CH_LAYOUT_QUAD || layout == AV_CH_LAYOUT_2_2)
return 0;
switch (codec_id) {
case AV_CODEC_ID_PCM_S8:
case AV_CODEC_ID_PCM_U8:
case AV_CODEC_ID_PCM_ALAW:
case AV_CODEC_ID_PCM_MULAW: format = FORMAT_I8; break;
case AV_CODEC_ID_PCM_S16LE:
case AV_CODEC_ID_PCM_S16BE:
case AV_CODEC_ID_PCM_U16LE:
case AV_CODEC_ID_PCM_U16BE: format = FORMAT_I16; break;
case AV_CODEC_ID_PCM_S32LE:
case AV_CODEC_ID_PCM_S32BE:
case AV_CODEC_ID_PCM_U32LE:
case AV_CODEC_ID_PCM_U32BE: format = FORMAT_I32; break;
case AV_CODEC_ID_PCM_F32LE:
case AV_CODEC_ID_PCM_F32BE: format = FORMAT_F32; break;
default: return AVERROR(ENOSYS);
}
if (layout == AV_CH_LAYOUT_5POINT0_BACK || layout == AV_CH_LAYOUT_5POINT0)
PICK_REORDER(50)
else if (layout == AV_CH_LAYOUT_5POINT1_BACK || layout == AV_CH_LAYOUT_5POINT1)
PICK_REORDER(51)
else if (layout == AV_CH_LAYOUT_7POINT1)
PICK_REORDER(71)
return s->reorder_func ? 0 : AVERROR(ENOSYS);
}
/**
* Open an ALSA PCM.
*
* @param s media file handle
* @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK
* @param sample_rate in: requested sample rate;
* out: actually selected sample rate
* @param channels number of channels
* @param codec_id in: requested AVCodecID or AV_CODEC_ID_NONE;
* out: actually selected AVCodecID, changed only if
* AV_CODEC_ID_NONE was requested
*
* @return 0 if OK, AVERROR_xxx on error
*/
static av_cold int alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
unsigned int *sample_rate,
int channels, enum AVCodecID *codec_id)
{
AlsaData *s = ctx->priv_data;
const char *audio_device;
int res, flags = 0;
snd_pcm_format_t format;
snd_pcm_t *h;
snd_pcm_hw_params_t *hw_params;
snd_pcm_uframes_t buffer_size, period_size;
uint64_t layout = ctx->streams[0]->codecpar->channel_layout;
if (ctx->filename[0] == 0) audio_device = "default";
else audio_device = ctx->filename;
if (*codec_id == AV_CODEC_ID_NONE)
*codec_id = DEFAULT_CODEC_ID;
format = codec_id_to_pcm_format(*codec_id);
if (format == SND_PCM_FORMAT_UNKNOWN) {
av_log(ctx, AV_LOG_ERROR, "sample format 0x%04x is not supported\n", *codec_id);
return AVERROR(ENOSYS);
}
s->frame_size = av_get_bits_per_sample(*codec_id) / 8 * channels;
if (ctx->flags & AVFMT_FLAG_NONBLOCK) {
flags = SND_PCM_NONBLOCK;
}
res = snd_pcm_open(&h, audio_device, mode, flags);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot open audio device %s (%s)\n",
audio_device, snd_strerror(res));
return AVERROR(EIO);
}
res = snd_pcm_hw_params_malloc(&hw_params);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot allocate hardware parameter structure (%s)\n",
snd_strerror(res));
goto fail1;
}
res = snd_pcm_hw_params_any(h, hw_params);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot initialize hardware parameter structure (%s)\n",
snd_strerror(res));
goto fail;
}
res = snd_pcm_hw_params_set_access(h, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot set access type (%s)\n",
snd_strerror(res));
goto fail;
}
res = snd_pcm_hw_params_set_format(h, hw_params, format);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot set sample format 0x%04x %d (%s)\n",
*codec_id, format, snd_strerror(res));
goto fail;
}
res = snd_pcm_hw_params_set_rate_near(h, hw_params, sample_rate, 0);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot set sample rate (%s)\n",
snd_strerror(res));
goto fail;
}
res = snd_pcm_hw_params_set_channels(h, hw_params, channels);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot set channel count to %d (%s)\n",
channels, snd_strerror(res));
goto fail;
}
snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size);
buffer_size = FFMIN(buffer_size, ALSA_BUFFER_SIZE_MAX);
/* TODO: maybe use ctx->max_picture_buffer somehow */
res = snd_pcm_hw_params_set_buffer_size_near(h, hw_params, &buffer_size);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot set ALSA buffer size (%s)\n",
snd_strerror(res));
goto fail;
}
snd_pcm_hw_params_get_period_size_min(hw_params, &period_size, NULL);
if (!period_size)
period_size = buffer_size / 4;
res = snd_pcm_hw_params_set_period_size_near(h, hw_params, &period_size, NULL);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot set ALSA period size (%s)\n",
snd_strerror(res));
goto fail;
}
s->period_size = period_size;
res = snd_pcm_hw_params(h, hw_params);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot set parameters (%s)\n",
snd_strerror(res));
goto fail;
}
snd_pcm_hw_params_free(hw_params);
if (channels > 2 && layout) {
if (find_reorder_func(s, *codec_id, layout, mode == SND_PCM_STREAM_PLAYBACK) < 0) {
char name[128];
av_get_channel_layout_string(name, sizeof(name), channels, layout);
av_log(ctx, AV_LOG_WARNING, "ALSA channel layout unknown or unimplemented for %s %s.\n",
name, mode == SND_PCM_STREAM_PLAYBACK ? "playback" : "capture");
}
if (s->reorder_func) {
s->reorder_buf_size = buffer_size;
s->reorder_buf = av_malloc(s->reorder_buf_size * s->frame_size);
if (!s->reorder_buf)
goto fail1;
}
}
s->h = h;
return 0;
fail:
snd_pcm_hw_params_free(hw_params);
fail1:
snd_pcm_close(h);
return AVERROR(EIO);
}
/**
* Close the ALSA PCM.
*
* @param s1 media file handle
*
* @return 0
*/
static av_cold int alsa_close(AVFormatContext *s1)
{
AlsaData *s = s1->priv_data;
av_freep(&s->reorder_buf);
snd_pcm_close(s->h);
return 0;
}
/**
* Try to recover from ALSA buffer underrun.
*
* @param s1 media file handle
* @param err error code reported by the previous ALSA call
*
* @return 0 if OK, AVERROR_xxx on error
*/
static int alsa_xrun_recover(AVFormatContext *s1, int err)
{
AlsaData *s = s1->priv_data;
snd_pcm_t *handle = s->h;
av_log(s1, AV_LOG_WARNING, "ALSA buffer xrun.\n");
if (err == -EPIPE) {
err = snd_pcm_prepare(handle);
if (err < 0) {
av_log(s1, AV_LOG_ERROR, "cannot recover from underrun (snd_pcm_prepare failed: %s)\n", snd_strerror(err));
return AVERROR(EIO);
}
} else if (err == -ESTRPIPE) {
av_log(s1, AV_LOG_ERROR, "-ESTRPIPE... Unsupported!\n");
return -1;
}
return err;
}
static av_cold int audio_read_header(AVFormatContext *s1)
{
AlsaData *s = s1->priv_data;
AVStream *st;
int ret;
enum AVCodecID codec_id;
snd_pcm_sw_params_t *sw_params;
st = avformat_new_stream(s1, NULL);
if (!st) {
av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
return AVERROR(ENOMEM);
}
codec_id = s1->audio_codec_id;
ret = alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
&codec_id);
if (ret < 0) {
return AVERROR(EIO);
}
if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW)
av_log(s1, AV_LOG_WARNING,
"capture with some ALSA plugins, especially dsnoop, "
"may hang.\n");
ret = snd_pcm_sw_params_malloc(&sw_params);
if (ret < 0) {
av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n",
snd_strerror(ret));
goto fail;
}
snd_pcm_sw_params_current(s->h, sw_params);
snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE);
ret = snd_pcm_sw_params(s->h, sw_params);
snd_pcm_sw_params_free(sw_params);
if (ret < 0) {
av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n",
snd_strerror(ret));
goto fail;
}
/* take real parameters */
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = codec_id;
st->codecpar->sample_rate = s->sample_rate;
st->codecpar->channels = s->channels;
avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
return 0;
fail:
snd_pcm_close(s->h);
return AVERROR(EIO);
}
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
{
AlsaData *s = s1->priv_data;
AVStream *st = s1->streams[0];
int res;
snd_htimestamp_t timestamp;
snd_pcm_uframes_t ts_delay;
if (av_new_packet(pkt, s->period_size) < 0) {
return AVERROR(EIO);
}
while ((res = snd_pcm_readi(s->h, pkt->data, pkt->size / s->frame_size)) < 0) {
if (res == -EAGAIN) {
av_packet_unref(pkt);
return AVERROR(EAGAIN);
}
if (alsa_xrun_recover(s1, res) < 0) {
av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
snd_strerror(res));
av_packet_unref(pkt);
return AVERROR(EIO);
}
}
snd_pcm_htimestamp(s->h, &ts_delay, &timestamp);
ts_delay += res;
pkt->pts = timestamp.tv_sec * 1000000LL
+ (timestamp.tv_nsec * st->codecpar->sample_rate
- (int64_t)ts_delay * 1000000000LL + st->codecpar->sample_rate * 500LL)
/ (st->codecpar->sample_rate * 1000LL);
pkt->size = res * s->frame_size;
return 0;
}
static const AVOption options[] = {
{ "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ NULL },
};
static const AVClass alsa_demuxer_class = {
.class_name = "ALSA demuxer",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
AVInputFormat ff_alsa_demuxer = {
.name = "alsa",
.long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
.priv_data_size = sizeof(AlsaData),
.read_header = audio_read_header,
.read_packet = audio_read_packet,
.read_close = alsa_close,
.flags = AVFMT_NOFILE,
.priv_class = &alsa_demuxer_class,
};
Reference in New Issue View Git Blame Copy Permalink