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/*
* ALSA input and output
* 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 and output: common code
* @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 "libavformat/avformat.h"
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "alsa.h"
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);
}
av_cold int ff_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;
lavf: replace AVStream.codec with AVStream.codecpar Currently, AVStream contains an embedded AVCodecContext instance, which is used by demuxers to export stream parameters to the caller and by muxers to receive stream parameters from the caller. It is also used internally as the codec context that is passed to parsers. In addition, it is also widely used by the callers as the decoding (when demuxer) or encoding (when muxing) context, though this has been officially discouraged since Libav 11. There are multiple important problems with this approach: - the fields in AVCodecContext are in general one of * stream parameters * codec options * codec state However, it's not clear which ones are which. It is consequently unclear which fields are a demuxer allowed to set or a muxer allowed to read. This leads to erratic behaviour depending on whether decoding or encoding is being performed or not (and whether it uses the AVStream embedded codec context). - various synchronization issues arising from the fact that the same context is used by several different APIs (muxers/demuxers, parsers, bitstream filters and encoders/decoders) simultaneously, with there being no clear rules for who can modify what and the different processes being typically delayed with respect to each other. - avformat_find_stream_info() making it necessary to support opening and closing a single codec context multiple times, thus complicating the semantics of freeing various allocated objects in the codec context. Those problems are resolved by replacing the AVStream embedded codec context with a newly added AVCodecParameters instance, which stores only the stream parameters exported by the demuxers or read by the muxers.
11 years ago
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);
}
av_cold int ff_alsa_close(AVFormatContext *s1)
{
AlsaData *s = s1->priv_data;
av_freep(&s->reorder_buf);
snd_pcm_close(s->h);
return 0;
}
int ff_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;
}
int ff_alsa_extend_reorder_buf(AlsaData *s, int min_size)
{
int size = s->reorder_buf_size;
void *r;
av_assert0(size != 0);
while (size < min_size)
size *= 2;
r = av_realloc(s->reorder_buf, size * s->frame_size);
if (!r)
return AVERROR(ENOMEM);
s->reorder_buf = r;
s->reorder_buf_size = size;
return 0;
}