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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 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 "config_components.h"
#include <alsa/asoundlib.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <unistd.h>
#include "avdevice.h"
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/mem.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_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_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;
default: return SND_PCM_FORMAT_UNKNOWN;
}
}
#define MAKE_REORDER_FUNC(NAME, TYPE, CHANNELS, LAYOUT, MAP) \
static void alsa_reorder_ ## NAME ## _ ## LAYOUT(const void *in_v, \
void *out_v, \
int n) \
{ \
const TYPE *in = in_v; \
TYPE *out = out_v; \
\
while (n-- > 0) { \
MAP \
in += CHANNELS; \
out += CHANNELS; \
} \
}
#define MAKE_REORDER_FUNCS(CHANNELS, LAYOUT, MAP) \
MAKE_REORDER_FUNC(int8, int8_t, CHANNELS, LAYOUT, MAP) \
MAKE_REORDER_FUNC(int16, int16_t, CHANNELS, LAYOUT, MAP) \
MAKE_REORDER_FUNC(int32, int32_t, CHANNELS, LAYOUT, MAP) \
MAKE_REORDER_FUNC(f32, float, CHANNELS, LAYOUT, MAP)
MAKE_REORDER_FUNCS(5, out_50, \
out[0] = in[0]; \
out[1] = in[1]; \
out[2] = in[3]; \
out[3] = in[4]; \
out[4] = in[2]; \
)
MAKE_REORDER_FUNCS(6, out_51, \
out[0] = in[0]; \
out[1] = in[1]; \
out[2] = in[4]; \
out[3] = in[5]; \
out[4] = in[2]; \
out[5] = in[3]; \
)
MAKE_REORDER_FUNCS(8, out_71, \
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]; \
)
#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, AVChannelLayout *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 (!av_channel_layout_compare(layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_QUAD) ||
!av_channel_layout_compare(layout, &(AVChannelLayout)AV_CHANNEL_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 (!av_channel_layout_compare(layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT0_BACK) ||
!av_channel_layout_compare(layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT0))
PICK_REORDER(50)
else if (!av_channel_layout_compare(layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT1_BACK) ||
!av_channel_layout_compare(layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT1))
PICK_REORDER(51)
else if (!av_channel_layout_compare(layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_7POINT1))
PICK_REORDER(71)
return s->reorder_func ? 0 : AVERROR(ENOSYS);
}
static int ff_audio_samplerate_convert(int samplerate, int *sample_rates, int num)
{
int i;
for (i = 0; i < num && samplerate; i++) {
if (samplerate & AUDIO_SAMP_RATE_8K) {
samplerate &= ~AUDIO_SAMP_RATE_8K;
sample_rates[i] = 8000;
} else if (samplerate & AUDIO_SAMP_RATE_11K) {
samplerate &= ~AUDIO_SAMP_RATE_11K;
sample_rates[i] = 11025;
} else if (samplerate & AUDIO_SAMP_RATE_12K) {
samplerate &= ~AUDIO_SAMP_RATE_12K;
sample_rates[i] = 12000;
} else if (samplerate & AUDIO_SAMP_RATE_16K) {
samplerate &= ~AUDIO_SAMP_RATE_16K;
sample_rates[i] = 16000;
} else if (samplerate & AUDIO_SAMP_RATE_22K) {
samplerate &= ~AUDIO_SAMP_RATE_22K;
sample_rates[i] = 22050;
} else if (samplerate & AUDIO_SAMP_RATE_24K) {
samplerate &= ~AUDIO_SAMP_RATE_24K;
sample_rates[i] = 24000;
} else if (samplerate & AUDIO_SAMP_RATE_32K) {
samplerate &= ~AUDIO_SAMP_RATE_32K;
sample_rates[i] = 32000;
} else if (samplerate & AUDIO_SAMP_RATE_44K) {
samplerate &= ~AUDIO_SAMP_RATE_44K;
sample_rates[i] = 44100;
} else if (samplerate & AUDIO_SAMP_RATE_48K) {
samplerate &= ~AUDIO_SAMP_RATE_48K;
sample_rates[i] = 48000;
} else if (samplerate & AUDIO_SAMP_RATE_96K) {
samplerate &= ~AUDIO_SAMP_RATE_96K;
sample_rates[i] = 96000;
} else if (samplerate & AUDIO_SAMP_RATE_128K) {
samplerate &= ~AUDIO_SAMP_RATE_128K;
sample_rates[i] = 128000;
} else if (samplerate & AUDIO_SAMP_RATE_160K) {
samplerate &= ~AUDIO_SAMP_RATE_160K;
sample_rates[i] = 160000;
} else if (samplerate & AUDIO_SAMP_RATE_172K) {
samplerate &= ~AUDIO_SAMP_RATE_172K;
sample_rates[i] = 172000;
} else if (samplerate & AUDIO_SAMP_RATE_192K) {
samplerate &= ~AUDIO_SAMP_RATE_192K;
sample_rates[i] = 192000;
}
}
return i;
}
static int ff_audio_subfmt_to_avcodec(int subfmt)
{
switch (subfmt) {
case AUDIO_SUBFMT_PCM_U8:
return AV_CODEC_ID_PCM_U8;
case AUDIO_SUBFMT_PCM_S8:
return AV_CODEC_ID_PCM_S8;
case AUDIO_SUBFMT_PCM_U16_LE:
return AV_CODEC_ID_PCM_U16LE;
case AUDIO_SUBFMT_PCM_U16_BE:
return AV_CODEC_ID_PCM_U16BE;
case AUDIO_SUBFMT_PCM_S16_LE:
return AV_CODEC_ID_PCM_S16LE;
case AUDIO_SUBFMT_PCM_S16_BE:
return AV_CODEC_ID_PCM_S16BE;
case AUDIO_SUBFMT_PCM_U32_LE:
return AV_CODEC_ID_PCM_U32LE;
case AUDIO_SUBFMT_PCM_U32_BE:
return AV_CODEC_ID_PCM_U32BE;
case AUDIO_SUBFMT_PCM_S32_LE:
return AV_CODEC_ID_PCM_S32LE;
case AUDIO_SUBFMT_PCM_S32_BE:
return AV_CODEC_ID_PCM_S32BE;
case AUDIO_SUBFMT_PCM_MU_LAW:
return AV_CODEC_ID_PCM_MULAW;
case AUDIO_SUBFMT_PCM_A_LAW:
return AV_CODEC_ID_PCM_ALAW;
case AUDIO_SUBFMT_PCM_MP1:
return AV_CODEC_ID_MP1;
case AUDIO_SUBFMT_PCM_MP2:
return AV_CODEC_ID_MP2;
case AUDIO_SUBFMT_PCM_MP3:
return AV_CODEC_ID_MP3;
}
return AV_CODEC_ID_FIRST_AUDIO;
}
static int ff_audio_subfmt_to_smpfmt(int subfmt)
{
switch (subfmt) {
case AUDIO_SUBFMT_PCM_U8:
return AV_SAMPLE_FMT_U8;
case AUDIO_SUBFMT_PCM_S16_LE:
case AUDIO_SUBFMT_PCM_S16_BE:
return AV_SAMPLE_FMT_S16;
case AUDIO_SUBFMT_PCM_S32_LE:
case AUDIO_SUBFMT_PCM_S32_BE:
return AV_SAMPLE_FMT_S32;
}
return AV_SAMPLE_FMT_NONE;
}
static int ff_audio_fmt_to_avcodec(int *codec_id, int *formats)
{
int codec = AV_CODEC_ID_NONE;
int format = AUDIO_FMT_UNDEF;
if (*formats & (1 << (AUDIO_FMT_PCM - 1))) {
codec = AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE);
format = AUDIO_FMT_PCM;
*formats &= ~(1 << (AUDIO_FMT_PCM - 1));
} else if (*formats & (1 << (AUDIO_FMT_MP3 - 1))) {
codec = AV_CODEC_ID_MP3;
format = AUDIO_FMT_MP3;
*formats &= ~(1 << (AUDIO_FMT_MP3 - 1));
} else if (*formats & (1 << (AUDIO_FMT_AC3 - 1))) {
codec = AV_CODEC_ID_AC3;
format = AUDIO_FMT_AC3;
*formats &= ~(1 << (AUDIO_FMT_AC3 - 1));
} else if (*formats & (1 << (AUDIO_FMT_WMA - 1))) {
codec = AV_CODEC_ID_WMAV2;
format = AUDIO_FMT_WMA;
*formats &= ~(1 << (AUDIO_FMT_WMA - 1));
} else if (*formats & (1 << (AUDIO_FMT_DTS - 1))) {
codec = AV_CODEC_ID_DTS;
format = AUDIO_FMT_WMA;
*formats &= ~(1 << (AUDIO_FMT_DTS - 1));
} else if (*formats & (1 << (AUDIO_FMT_OGG_VORBIS - 1))) {
codec = AV_CODEC_ID_VORBIS;
format = AUDIO_FMT_OGG_VORBIS;
*formats &= ~(1 << (AUDIO_FMT_OGG_VORBIS - 1));
} else if (*formats & (1 << (AUDIO_FMT_FLAC - 1))) {
codec = AV_CODEC_ID_FLAC;
format = AUDIO_FMT_FLAC;
*formats &= ~(1 << (AUDIO_FMT_FLAC - 1));
} else if (*formats & (1 << (AUDIO_FMT_AMR - 1))) {
codec = AV_CODEC_ID_AMR_NB;
format = AUDIO_FMT_AMR;
*formats &= ~(1 << (AUDIO_FMT_AMR - 1));
} else if (*formats & (1 << (AUDIO_FMT_OTHER - 1))) {
format = AUDIO_FMT_OTHER;
*formats &= ~(1 << (AUDIO_FMT_OTHER - 1));
} else if (*formats & (1 << (AUDIO_FMT_OPUS - 1))) {
codec = AV_CODEC_ID_OPUS;
format = AUDIO_FMT_OPUS;
*formats &= ~(1 << (AUDIO_FMT_OPUS - 1));
} else if (*formats & (1 << (AUDIO_FMT_AMRWB - 1))) {
codec = AV_CODEC_ID_AMR_WB;
format = AUDIO_FMT_AMRWB;
*formats &= ~(1 << (AUDIO_FMT_AMRWB - 1));
}
*codec_id = codec;
return format;
}
static int ff_audio_pcm_ioctl(int fd, int cmd, unsigned long arg)
{
int ret;
ret = ioctl(fd, cmd, arg);
if (ret < 0) {
ret = -errno;
}
return ret;
}
#define ff_audio_pcm_ioctl(fd, cmd, arg) \
ff_audio_pcm_ioctl(fd, cmd, (unsigned long)(arg))
static int ff_audio_get_capabilities(char *device, int ac_type, int ac_subtype,
struct audio_caps_s *caps)
{
int ret;
int fd;
char path[32];
snprintf(path, sizeof(path), CONFIG_AUDIOUTILS_ALSA_LIB_DEV_PATH "/%s", device);
fd = open(path, O_RDWR | O_CLOEXEC);
if (fd < 0)
return -ENOENT;
caps->ac_len = sizeof(struct audio_caps_s);
caps->ac_type = ac_type;
caps->ac_subtype = ac_subtype;
ret = ff_audio_pcm_ioctl(fd, AUDIOIOC_GETCAPS, caps);
close(fd);
return ret;
}
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;
AVChannelLayout *layout = &ctx->streams[0]->codecpar->ch_layout;
const char *audio_device;
int res, dir = 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;
if (ctx->url[0] == 0) audio_device = "default";
else audio_device = ctx->url;
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;
res = snd_pcm_open(&h, audio_device, mode, SND_PCM_NONBLOCK);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot open audio device %s (%s)\n",
audio_device, snd_strerror(res));
return AVERROR(EIO);
}
snd_pcm_hw_params_alloca(&hw_params);
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_set_period_time(h, hw_params, s->period_time * 1000, dir);
snd_pcm_hw_params_set_periods(h, hw_params, s->periods, dir);
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;
}
if (channels > 2 && layout->order != AV_CHANNEL_ORDER_UNSPEC) {
if (find_reorder_func(s, *codec_id, layout, mode == SND_PCM_STREAM_PLAYBACK) < 0) {
char name[128];
av_channel_layout_describe(layout, name, sizeof(name));
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_array(s->reorder_buf_size, s->frame_size);
if (!s->reorder_buf)
goto fail;
}
}
s->pkt = av_packet_alloc();
if (!s->pkt)
goto fail;
s->h = h;
return 0;
fail:
snd_pcm_close(h);
return AVERROR(EIO);
}
av_cold int ff_alsa_close(AVFormatContext *s1)
{
AlsaData *s = s1->priv_data;
snd_pcm_drain(s->h);
av_freep(&s->reorder_buf);
if (CONFIG_ALSA_INDEV)
ff_timefilter_destroy(s->timefilter);
snd_pcm_close(s->h);
av_packet_free(&s->pkt);
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 ret=%d.\n", err);
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;
} else if (err == -EAGAIN) {
return 0;
}
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_array(s->reorder_buf, size, s->frame_size);
if (!r)
return AVERROR(ENOMEM);
s->reorder_buf = r;
s->reorder_buf_size = size;
return 0;
}
/* ported from alsa-utils/aplay.c */
int ff_alsa_get_device_list(AVDeviceInfoList *device_list, snd_pcm_stream_t stream_type)
{
return 0;
}
static int ff_audio_set_ranges(struct AVOptionRanges *ranges, int nb_ranges, int is_range,
int min_v[], int max_v[])
{
ranges->nb_components = 1;
ranges->nb_ranges = nb_ranges;
ranges->range = av_mallocz(nb_ranges * sizeof(AVOptionRange*));
if (!ranges->range)
return AVERROR(ENOMEM);
for (int i = 0; i < nb_ranges; i++) {
ranges->range[i] = av_mallocz(sizeof(AVOptionRange));
if (!ranges->range[i])
return AVERROR(ENOMEM);
ranges->range[i]->is_range = is_range;
ranges->range[i]->value_min = min_v[i];
ranges->range[i]->value_max = is_range ? max_v[i] : min_v[i];
}
return 0;
}
static int ff_audio_capbility_query_smpfmts(struct AVFormatContext *s1, int format, int values[])
{
struct audio_caps_s smpfmts;
int ret, x;
AlsaData *s = s1->priv_data;
if (((format & (1 << (AUDIO_FMT_PCM - 1))) == 0))
return AVERROR(EPERM);
ret = ff_audio_get_capabilities(s1->url, AUDIO_TYPE_QUERY, AUDIO_FMT_PCM, &smpfmts);
if (ret < 0)
return ret;
for (x = 0; x < sizeof(smpfmts.ac_controls.b); x++) {
if (smpfmts.ac_controls.b[x] == AUDIO_SUBFMT_END)
break;
ret = ff_audio_subfmt_to_smpfmt(smpfmts.ac_controls.b[x]);
if (ret >= 0)
values[x] = ret;
}
return x == 0 ? AVERROR(EPERM) : x;
}
static int ff_audio_capbility_query_codecs(struct AVFormatContext *s1,
int format, int codecs[], int num)
{
int ac_subtype = AUDIO_FMT_UNDEF;
int codec = AV_CODEC_ID_NONE;
struct audio_caps_s caps;
int i, nb_codecs = 0;
AlsaData *s = s1->priv_data;
while (nb_codecs < num && format) {
ac_subtype = ff_audio_fmt_to_avcodec(&codec, &format);
if (ff_audio_get_capabilities(s1->url, AUDIO_TYPE_QUERY, ac_subtype, &caps) < 0)
continue;
if (ac_subtype == AUDIO_FMT_OTHER) {
nb_codecs += ff_audio_capbility_query_codecs(s1, caps.ac_controls.w, &codecs[nb_codecs], num - nb_codecs);
continue;
}
for (i = 0; i < sizeof(caps.ac_controls.b) && nb_codecs < num; i++) {
if (caps.ac_controls.b[i] == AUDIO_SUBFMT_END) {
if (i == 0)
codecs[nb_codecs++] = codec;
break;
}
codecs[nb_codecs++] = ff_audio_subfmt_to_avcodec(caps.ac_controls.b[i]);
}
}
return nb_codecs;
}
int ff_audio_capbility_query_ranges(struct AVOptionRanges **ranges_, void *obj,
const char *key, int flags, bool playback)
{
struct AVFormatContext *s1 = obj;
struct audio_caps_s formats, others;
int ac_type = AUDIO_TYPE_QUERY;
struct AVOptionRanges *ranges;
int values0[64], values1[64];
int nb_ranges, is_range = 0;
int ret;
AlsaData *s = s1->priv_data;
ranges = av_mallocz(sizeof(struct AVOptionRanges));
if (!ranges)
return AVERROR(ENOMEM);
if (!strcmp(key, "sample_fmts") || !strcmp(key, "codecs")) {
ret = ff_audio_get_capabilities(s1->url, ac_type, AUDIO_TYPE_QUERY, &formats);
if (ret < 0)
goto err;
if (!strcmp(key, "sample_fmts")) {
ret = ff_audio_capbility_query_smpfmts(s1, formats.ac_format.hw, values0);
if (ret < 0)
goto err;
} else {
ret = ff_audio_capbility_query_codecs(s1, formats.ac_format.hw, values0, 64);
if (ret < 0)
goto err;
}
nb_ranges = ret;
} else if (!strcmp(key, "channels") || !strcmp(key, "sample_rates")) {
ac_type = playback ? AUDIO_TYPE_OUTPUT : AUDIO_TYPE_INPUT;
ret = ff_audio_get_capabilities(s1->url, ac_type, AUDIO_TYPE_QUERY, &others);
if (ret < 0)
goto err;
if (!strcmp(key, "channels")) {
if ((others.ac_channels & 0xf0) == 0) {
values0[0] = 1;
values1[0] = others.ac_channels;
} else {
values0[0] = others.ac_channels >> 4;
values1[0] = others.ac_channels & 0x0f;
}
nb_ranges = 1;
is_range = (values0[0] != values1[0]);
} else {
ret = ff_audio_samplerate_convert(others.ac_controls.hw[0], values0, 64);
if (ret < 0)
goto err;
nb_ranges = ret;
}
} else {
ret = -EINVAL;
goto err;
}
ret = ff_audio_set_ranges(ranges, nb_ranges, is_range, values0, values1);
if (ret < 0)
goto err;
*ranges_ = ranges;
return ranges->nb_components;
err:
av_opt_freep_ranges(&ranges);
return ret;
}
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