ffmpeg/libavdevice/alsa-audio-common.c
Michael Niedermayer ec7f0b527c Merge remote-tracking branch 'khirnov/release/0.7' into release/0.8
* khirnov/release/0.7: (64 commits)
  rv34: Check for invalid slice offsets
  rv34: Fix potential overreads
  rv34: Avoid NULL dereference on corrupted bitstream
  rv10: Reject slices that does not have the same type as the first one
  lavf: Fix context pointer in av_open_input_stream when avformat_open_input fails
  oggdec: fix out of bound write in the ogg demuxer
  Fixed size given to init_get_bits().
  smacker: fix a few off by 1 errors
  Check for invalid VLC value in smacker decoder.
  Check and propagate errors when VLC trees cannot be built in smacker decoder.
  Fixed off by one packet size allocation in the smacker demuxer.
  Check for invalid packet size in the smacker demuxer.
  ape demuxer: fix segfault on memory allocation failure.
  xan: Add some buffer checks (cherry picked from commit 0872bb23b4bd2d94a8ba91070f706d1bc1c3ced8)
  Fixed size given to init_get_bits() in xan decoder. (cherry picked from commit 393d5031c6aaaf8c2dda4eb5d676974c349fae85)
  smacker demuxer: handle possible av_realloc() failure.
  Fixed segfault with wavpack decoder on corrupted decorrelation terms sub-blocks.
  cljr: init_get_bits size in bits instead of bytes (cherry picked from commit 0c1f5b93d9b97c4cc3684ba91a040e90bfc760d2)
  indeo2: fail if input buffer too small (cherry picked from commit b7ce4f1d1c3add86ece7ca595ea6c4a10b471055)
  indeo2: init_get_bits size in bits instead of bytes (cherry picked from commit 68ca330cbd479111db9cb7649d7530ad59f04cc8)
  ...

Conflicts:
	ffmpeg.c
	libavdevice/alsa-audio.h
	libavformat/gxf.c
	libswscale/x86/swscale_template.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-22 01:10:24 +02:00

351 lines
12 KiB
C

/*
* 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 FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; 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 "avdevice.h"
#include "alsa-audio.h"
static av_cold snd_pcm_format_t codec_id_to_pcm_format(int codec_id)
{
switch(codec_id) {
case CODEC_ID_PCM_F64LE: return SND_PCM_FORMAT_FLOAT64_LE;
case CODEC_ID_PCM_F64BE: return SND_PCM_FORMAT_FLOAT64_BE;
case CODEC_ID_PCM_F32LE: return SND_PCM_FORMAT_FLOAT_LE;
case CODEC_ID_PCM_F32BE: return SND_PCM_FORMAT_FLOAT_BE;
case CODEC_ID_PCM_S32LE: return SND_PCM_FORMAT_S32_LE;
case CODEC_ID_PCM_S32BE: return SND_PCM_FORMAT_S32_BE;
case CODEC_ID_PCM_U32LE: return SND_PCM_FORMAT_U32_LE;
case CODEC_ID_PCM_U32BE: return SND_PCM_FORMAT_U32_BE;
case CODEC_ID_PCM_S24LE: return SND_PCM_FORMAT_S24_3LE;
case CODEC_ID_PCM_S24BE: return SND_PCM_FORMAT_S24_3BE;
case CODEC_ID_PCM_U24LE: return SND_PCM_FORMAT_U24_3LE;
case CODEC_ID_PCM_U24BE: return SND_PCM_FORMAT_U24_3BE;
case CODEC_ID_PCM_S16LE: return SND_PCM_FORMAT_S16_LE;
case CODEC_ID_PCM_S16BE: return SND_PCM_FORMAT_S16_BE;
case CODEC_ID_PCM_U16LE: return SND_PCM_FORMAT_U16_LE;
case CODEC_ID_PCM_U16BE: return SND_PCM_FORMAT_U16_BE;
case CODEC_ID_PCM_S8: return SND_PCM_FORMAT_S8;
case CODEC_ID_PCM_U8: return SND_PCM_FORMAT_U8;
case CODEC_ID_PCM_MULAW: return SND_PCM_FORMAT_MU_LAW;
case 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(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 REORDER_DUMMY ((void *)1)
static av_cold ff_reorder_func find_reorder_func(int codec_id,
int64_t layout,
int out)
{
return
codec_id == CODEC_ID_PCM_U16LE || codec_id == CODEC_ID_PCM_U16BE ||
codec_id == CODEC_ID_PCM_S16LE || codec_id == CODEC_ID_PCM_S16BE ?
layout == AV_CH_LAYOUT_QUAD || layout == AV_CH_LAYOUT_2_2 ?
REORDER_DUMMY :
layout == AV_CH_LAYOUT_5POINT0_BACK || layout == AV_CH_LAYOUT_5POINT0 ?
out ? alsa_reorder_int16_out_50 : NULL :
layout == AV_CH_LAYOUT_5POINT1_BACK || layout == AV_CH_LAYOUT_5POINT1 ?
out ? alsa_reorder_int16_out_51 : NULL :
layout == AV_CH_LAYOUT_7POINT1 ?
out ? alsa_reorder_int16_out_71 : NULL :
NULL :
codec_id == CODEC_ID_PCM_U32LE || codec_id == CODEC_ID_PCM_U32BE ||
codec_id == CODEC_ID_PCM_S32LE || codec_id == CODEC_ID_PCM_S32BE ?
layout == AV_CH_LAYOUT_QUAD || layout == AV_CH_LAYOUT_2_2 ?
REORDER_DUMMY :
layout == AV_CH_LAYOUT_5POINT0_BACK || layout == AV_CH_LAYOUT_5POINT0 ?
out ? alsa_reorder_int32_out_50 : NULL :
layout == AV_CH_LAYOUT_5POINT1_BACK || layout == AV_CH_LAYOUT_5POINT1 ?
out ? alsa_reorder_int32_out_51 : NULL :
layout == AV_CH_LAYOUT_7POINT1 ?
out ? alsa_reorder_int32_out_71 : NULL :
NULL :
codec_id == CODEC_ID_PCM_F32LE || codec_id == CODEC_ID_PCM_F32BE ?
layout == AV_CH_LAYOUT_QUAD || layout == AV_CH_LAYOUT_2_2 ?
REORDER_DUMMY :
layout == AV_CH_LAYOUT_5POINT0_BACK || layout == AV_CH_LAYOUT_5POINT0 ?
out ? alsa_reorder_f32_out_50 : NULL :
layout == AV_CH_LAYOUT_5POINT1_BACK || layout == AV_CH_LAYOUT_5POINT1 ?
out ? alsa_reorder_f32_out_51 : NULL :
layout == AV_CH_LAYOUT_7POINT1 ?
out ? alsa_reorder_f32_out_71 : NULL :
NULL :
NULL;
}
av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
unsigned int *sample_rate,
int channels, enum CodecID *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;
int64_t layout = ctx->streams[0]->codec->channel_layout;
if (ctx->filename[0] == 0) audio_device = "default";
else audio_device = ctx->filename;
if (*codec_id == 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) {
s->reorder_func = find_reorder_func(*codec_id, layout,
mode == SND_PCM_STREAM_PLAYBACK);
if (s->reorder_func == REORDER_DUMMY) {
s->reorder_func = NULL;
} else 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;
} else {
char name[32];
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");
}
}
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;
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;
}