mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-23 12:43:46 +02:00
Merge commit '074a00d192c0e749d677b008b337da42597e780f'
* commit '074a00d192c0e749d677b008b337da42597e780f': lavr: add a public function for setting a custom channel map lavr: typedef internal structs in internal.h doc: Extend commit message section Conflicts: doc/APIchanges doc/developer.texi Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
commit
249fca3df9
@ -132,6 +132,10 @@ API changes, most recent first:
|
||||
2012-03-26 - a67d9cf - lavfi 2.66.100
|
||||
Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions.
|
||||
|
||||
2013-xx-xx - xxxxxxx - lavr 1.1.0
|
||||
Add avresample_set_channel_mapping() for input channel reordering,
|
||||
duplication, and silencing.
|
||||
|
||||
2012-xx-xx - xxxxxxx - lavu 52.2.1 - avstring.h
|
||||
Add av_basename() and av_dirname().
|
||||
|
||||
|
@ -228,6 +228,13 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
|
||||
You can commit unfinished stuff (for testing etc), but it must be disabled
|
||||
(#ifdef etc) by default so it does not interfere with other developers'
|
||||
work.
|
||||
@item
|
||||
The commit message should have a short first line in the form of
|
||||
a @samp{topic: short description} as a header, separated by a newline
|
||||
from the body consisting of an explanation of why the change is necessary.
|
||||
If the commit fixes a known bug on the bug tracker, the commit message
|
||||
should include its bug ID. Referring to the issue on the bug tracker does
|
||||
not exempt you from writing an excerpt of the bug in the commit message.
|
||||
@item
|
||||
You do not have to over-test things. If it works for you, and you think it
|
||||
should work for others, then commit. If your code has problems
|
||||
|
@ -30,7 +30,6 @@
|
||||
#include "audio_convert.h"
|
||||
#include "audio_data.h"
|
||||
#include "dither.h"
|
||||
#include "internal.h"
|
||||
|
||||
enum ConvFuncType {
|
||||
CONV_FUNC_TYPE_FLAT,
|
||||
@ -51,6 +50,7 @@ struct AudioConvert {
|
||||
DitherContext *dc;
|
||||
enum AVSampleFormat in_fmt;
|
||||
enum AVSampleFormat out_fmt;
|
||||
int apply_map;
|
||||
int channels;
|
||||
int planes;
|
||||
int ptr_align;
|
||||
@ -260,7 +260,8 @@ void ff_audio_convert_free(AudioConvert **ac)
|
||||
AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
|
||||
enum AVSampleFormat out_fmt,
|
||||
enum AVSampleFormat in_fmt,
|
||||
int channels, int sample_rate)
|
||||
int channels, int sample_rate,
|
||||
int apply_map)
|
||||
{
|
||||
AudioConvert *ac;
|
||||
int in_planar, out_planar;
|
||||
@ -273,11 +274,13 @@ AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
|
||||
ac->out_fmt = out_fmt;
|
||||
ac->in_fmt = in_fmt;
|
||||
ac->channels = channels;
|
||||
ac->apply_map = apply_map;
|
||||
|
||||
if (avr->dither_method != AV_RESAMPLE_DITHER_NONE &&
|
||||
av_get_packed_sample_fmt(out_fmt) == AV_SAMPLE_FMT_S16 &&
|
||||
av_get_bytes_per_sample(in_fmt) > 2) {
|
||||
ac->dc = ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate);
|
||||
ac->dc = ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate,
|
||||
apply_map);
|
||||
if (!ac->dc) {
|
||||
av_free(ac);
|
||||
return NULL;
|
||||
@ -310,6 +313,7 @@ int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in)
|
||||
{
|
||||
int use_generic = 1;
|
||||
int len = in->nb_samples;
|
||||
int p;
|
||||
|
||||
if (ac->dc) {
|
||||
/* dithered conversion */
|
||||
@ -336,9 +340,46 @@ int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in)
|
||||
av_get_sample_fmt_name(ac->out_fmt),
|
||||
use_generic ? ac->func_descr_generic : ac->func_descr);
|
||||
|
||||
if (ac->apply_map) {
|
||||
ChannelMapInfo *map = &ac->avr->ch_map_info;
|
||||
|
||||
if (!av_sample_fmt_is_planar(ac->out_fmt)) {
|
||||
av_log(ac->avr, AV_LOG_ERROR, "cannot remap packed format during conversion\n");
|
||||
return AVERROR(EINVAL);
|
||||
}
|
||||
|
||||
if (map->do_remap) {
|
||||
if (av_sample_fmt_is_planar(ac->in_fmt)) {
|
||||
conv_func_flat *convert = use_generic ? ac->conv_flat_generic :
|
||||
ac->conv_flat;
|
||||
|
||||
for (p = 0; p < ac->planes; p++)
|
||||
if (map->channel_map[p] >= 0)
|
||||
convert(out->data[p], in->data[map->channel_map[p]], len);
|
||||
} else {
|
||||
uint8_t *data[AVRESAMPLE_MAX_CHANNELS];
|
||||
conv_func_deinterleave *convert = use_generic ?
|
||||
ac->conv_deinterleave_generic :
|
||||
ac->conv_deinterleave;
|
||||
|
||||
for (p = 0; p < ac->channels; p++)
|
||||
data[map->input_map[p]] = out->data[p];
|
||||
|
||||
convert(data, in->data[0], len, ac->channels);
|
||||
}
|
||||
}
|
||||
if (map->do_copy || map->do_zero) {
|
||||
for (p = 0; p < ac->planes; p++) {
|
||||
if (map->channel_copy[p])
|
||||
memcpy(out->data[p], out->data[map->channel_copy[p]],
|
||||
len * out->stride);
|
||||
else if (map->channel_zero[p])
|
||||
av_samples_set_silence(&out->data[p], 0, len, 1, ac->out_fmt);
|
||||
}
|
||||
}
|
||||
} else {
|
||||
switch (ac->func_type) {
|
||||
case CONV_FUNC_TYPE_FLAT: {
|
||||
int p;
|
||||
if (!in->is_planar)
|
||||
len *= in->channels;
|
||||
if (use_generic) {
|
||||
@ -363,6 +404,7 @@ int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in)
|
||||
ac->conv_deinterleave(out->data, in->data[0], len, ac->channels);
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
out->nb_samples = in->nb_samples;
|
||||
return 0;
|
||||
|
@ -23,10 +23,9 @@
|
||||
|
||||
#include "libavutil/samplefmt.h"
|
||||
#include "avresample.h"
|
||||
#include "internal.h"
|
||||
#include "audio_data.h"
|
||||
|
||||
typedef struct AudioConvert AudioConvert;
|
||||
|
||||
/**
|
||||
* Set conversion function if the parameters match.
|
||||
*
|
||||
@ -59,12 +58,14 @@ void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt,
|
||||
* @param in_fmt input sample format
|
||||
* @param channels number of channels
|
||||
* @param sample_rate sample rate (used for dithering)
|
||||
* @param apply_map apply channel map during conversion
|
||||
* @return newly-allocated AudioConvert context
|
||||
*/
|
||||
AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
|
||||
enum AVSampleFormat out_fmt,
|
||||
enum AVSampleFormat in_fmt,
|
||||
int channels, int sample_rate);
|
||||
int channels, int sample_rate,
|
||||
int apply_map);
|
||||
|
||||
/**
|
||||
* Free AudioConvert.
|
||||
|
@ -213,7 +213,7 @@ void ff_audio_data_free(AudioData **a)
|
||||
av_freep(a);
|
||||
}
|
||||
|
||||
int ff_audio_data_copy(AudioData *dst, AudioData *src)
|
||||
int ff_audio_data_copy(AudioData *dst, AudioData *src, ChannelMapInfo *map)
|
||||
{
|
||||
int ret, p;
|
||||
|
||||
@ -221,6 +221,11 @@ int ff_audio_data_copy(AudioData *dst, AudioData *src)
|
||||
if (dst->sample_fmt != src->sample_fmt || dst->channels < src->channels)
|
||||
return AVERROR(EINVAL);
|
||||
|
||||
if (map && !src->is_planar) {
|
||||
av_log(src, AV_LOG_ERROR, "cannot remap packed format during copy\n");
|
||||
return AVERROR(EINVAL);
|
||||
}
|
||||
|
||||
/* if the input is empty, just empty the output */
|
||||
if (!src->nb_samples) {
|
||||
dst->nb_samples = 0;
|
||||
@ -233,8 +238,29 @@ int ff_audio_data_copy(AudioData *dst, AudioData *src)
|
||||
return ret;
|
||||
|
||||
/* copy data */
|
||||
for (p = 0; p < src->planes; p++)
|
||||
memcpy(dst->data[p], src->data[p], src->nb_samples * src->stride);
|
||||
if (map) {
|
||||
if (map->do_remap) {
|
||||
for (p = 0; p < src->planes; p++) {
|
||||
if (map->channel_map[p] >= 0)
|
||||
memcpy(dst->data[p], src->data[map->channel_map[p]],
|
||||
src->nb_samples * src->stride);
|
||||
}
|
||||
}
|
||||
if (map->do_copy || map->do_zero) {
|
||||
for (p = 0; p < src->planes; p++) {
|
||||
if (map->channel_copy[p])
|
||||
memcpy(dst->data[p], dst->data[map->channel_copy[p]],
|
||||
src->nb_samples * src->stride);
|
||||
else if (map->channel_zero[p])
|
||||
av_samples_set_silence(&dst->data[p], 0, src->nb_samples,
|
||||
1, dst->sample_fmt);
|
||||
}
|
||||
}
|
||||
} else {
|
||||
for (p = 0; p < src->planes; p++)
|
||||
memcpy(dst->data[p], src->data[p], src->nb_samples * src->stride);
|
||||
}
|
||||
|
||||
dst->nb_samples = src->nb_samples;
|
||||
|
||||
return 0;
|
||||
|
@ -27,11 +27,12 @@
|
||||
#include "libavutil/log.h"
|
||||
#include "libavutil/samplefmt.h"
|
||||
#include "avresample.h"
|
||||
#include "internal.h"
|
||||
|
||||
/**
|
||||
* Audio buffer used for intermediate storage between conversion phases.
|
||||
*/
|
||||
typedef struct AudioData {
|
||||
struct AudioData {
|
||||
const AVClass *class; /**< AVClass for logging */
|
||||
uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; /**< data plane pointers */
|
||||
uint8_t *buffer; /**< data buffer */
|
||||
@ -50,7 +51,7 @@ typedef struct AudioData {
|
||||
int ptr_align; /**< minimum data pointer alignment */
|
||||
int samples_align; /**< allocated samples alignment */
|
||||
const char *name; /**< name for debug logging */
|
||||
} AudioData;
|
||||
};
|
||||
|
||||
int ff_audio_data_set_channels(AudioData *a, int channels);
|
||||
|
||||
@ -117,9 +118,10 @@ void ff_audio_data_free(AudioData **a);
|
||||
*
|
||||
* @param out output AudioData
|
||||
* @param in input AudioData
|
||||
* @param map channel map, NULL if not remapping
|
||||
* @return 0 on success, negative AVERROR value on error
|
||||
*/
|
||||
int ff_audio_data_copy(AudioData *out, AudioData *in);
|
||||
int ff_audio_data_copy(AudioData *out, AudioData *in, ChannelMapInfo *map);
|
||||
|
||||
/**
|
||||
* Append data from one AudioData to the end of another.
|
||||
|
@ -25,13 +25,12 @@
|
||||
|
||||
#include "libavutil/samplefmt.h"
|
||||
#include "avresample.h"
|
||||
#include "internal.h"
|
||||
#include "audio_data.h"
|
||||
|
||||
typedef void (mix_func)(uint8_t **src, void **matrix, int len, int out_ch,
|
||||
int in_ch);
|
||||
|
||||
typedef struct AudioMix AudioMix;
|
||||
|
||||
/**
|
||||
* Set mixing function if the parameters match.
|
||||
*
|
||||
|
@ -258,6 +258,36 @@ int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
|
||||
int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
|
||||
int stride);
|
||||
|
||||
/**
|
||||
* Set a customized input channel mapping.
|
||||
*
|
||||
* This function can only be called when the allocated context is not open.
|
||||
* Also, the input channel layout must have already been set.
|
||||
*
|
||||
* Calling avresample_close() on the context will clear the channel mapping.
|
||||
*
|
||||
* The map for each input channel specifies the channel index in the source to
|
||||
* use for that particular channel, or -1 to mute the channel. Source channels
|
||||
* can be duplicated by using the same index for multiple input channels.
|
||||
*
|
||||
* Examples:
|
||||
*
|
||||
* Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to Libav order (L,R,C,LFE,Ls,Rs):
|
||||
* { 1, 2, 0, 5, 3, 4 }
|
||||
*
|
||||
* Muting the 3rd channel in 4-channel input:
|
||||
* { 0, 1, -1, 3 }
|
||||
*
|
||||
* Duplicating the left channel of stereo input:
|
||||
* { 0, 0 }
|
||||
*
|
||||
* @param avr audio resample context
|
||||
* @param channel_map customized input channel mapping
|
||||
* @return 0 on success, negative AVERROR code on failure
|
||||
*/
|
||||
int avresample_set_channel_mapping(AVAudioResampleContext *avr,
|
||||
const int *channel_map);
|
||||
|
||||
/**
|
||||
* Set compensation for resampling.
|
||||
*
|
||||
|
@ -53,6 +53,8 @@ typedef struct DitherState {
|
||||
struct DitherContext {
|
||||
DitherDSPContext ddsp;
|
||||
enum AVResampleDitherMethod method;
|
||||
int apply_map;
|
||||
ChannelMapInfo *ch_map_info;
|
||||
|
||||
int mute_dither_threshold; // threshold for disabling dither
|
||||
int mute_reset_threshold; // threshold for resetting noise shaping
|
||||
@ -251,17 +253,23 @@ int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src)
|
||||
return ret;
|
||||
}
|
||||
|
||||
if (src->sample_fmt != AV_SAMPLE_FMT_FLTP) {
|
||||
if (src->sample_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) {
|
||||
/* make sure flt_data is large enough for the input */
|
||||
ret = ff_audio_data_realloc(c->flt_data, src->nb_samples);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
flt_data = c->flt_data;
|
||||
}
|
||||
|
||||
if (src->sample_fmt != AV_SAMPLE_FMT_FLTP) {
|
||||
/* convert input samples to fltp and scale to s16 range */
|
||||
ret = ff_audio_convert(c->ac_in, flt_data, src);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
} else if (c->apply_map) {
|
||||
ret = ff_audio_data_copy(flt_data, src, c->ch_map_info);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
} else {
|
||||
flt_data = src;
|
||||
}
|
||||
@ -333,7 +341,7 @@ static void dither_init(DitherDSPContext *ddsp,
|
||||
DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
|
||||
enum AVSampleFormat out_fmt,
|
||||
enum AVSampleFormat in_fmt,
|
||||
int channels, int sample_rate)
|
||||
int channels, int sample_rate, int apply_map)
|
||||
{
|
||||
AVLFG seed_gen;
|
||||
DitherContext *c;
|
||||
@ -350,6 +358,10 @@ DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
|
||||
if (!c)
|
||||
return NULL;
|
||||
|
||||
c->apply_map = apply_map;
|
||||
if (apply_map)
|
||||
c->ch_map_info = &avr->ch_map_info;
|
||||
|
||||
if (avr->dither_method == AV_RESAMPLE_DITHER_TRIANGULAR_NS &&
|
||||
sample_rate != 48000 && sample_rate != 44100) {
|
||||
av_log(avr, AV_LOG_WARNING, "sample rate must be 48000 or 44100 Hz "
|
||||
@ -379,19 +391,20 @@ DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
|
||||
goto fail;
|
||||
|
||||
c->ac_out = ff_audio_convert_alloc(avr, out_fmt, AV_SAMPLE_FMT_S16P,
|
||||
channels, sample_rate);
|
||||
channels, sample_rate, 0);
|
||||
if (!c->ac_out)
|
||||
goto fail;
|
||||
}
|
||||
|
||||
if (in_fmt != AV_SAMPLE_FMT_FLTP) {
|
||||
if (in_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) {
|
||||
c->flt_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_FLTP,
|
||||
"dither flt buffer");
|
||||
if (!c->flt_data)
|
||||
goto fail;
|
||||
|
||||
}
|
||||
if (in_fmt != AV_SAMPLE_FMT_FLTP) {
|
||||
c->ac_in = ff_audio_convert_alloc(avr, AV_SAMPLE_FMT_FLTP, in_fmt,
|
||||
channels, sample_rate);
|
||||
channels, sample_rate, c->apply_map);
|
||||
if (!c->ac_in)
|
||||
goto fail;
|
||||
}
|
||||
|
@ -66,7 +66,7 @@ typedef struct DitherDSPContext {
|
||||
DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
|
||||
enum AVSampleFormat out_fmt,
|
||||
enum AVSampleFormat in_fmt,
|
||||
int channels, int sample_rate);
|
||||
int channels, int sample_rate, int apply_map);
|
||||
|
||||
/**
|
||||
* Free a DitherContext.
|
||||
|
@ -26,10 +26,29 @@
|
||||
#include "libavutil/opt.h"
|
||||
#include "libavutil/samplefmt.h"
|
||||
#include "avresample.h"
|
||||
#include "audio_convert.h"
|
||||
#include "audio_data.h"
|
||||
#include "audio_mix.h"
|
||||
#include "resample.h"
|
||||
|
||||
typedef struct AudioData AudioData;
|
||||
typedef struct AudioConvert AudioConvert;
|
||||
typedef struct AudioMix AudioMix;
|
||||
typedef struct ResampleContext ResampleContext;
|
||||
|
||||
enum RemapPoint {
|
||||
REMAP_NONE,
|
||||
REMAP_IN_COPY,
|
||||
REMAP_IN_CONVERT,
|
||||
REMAP_OUT_COPY,
|
||||
REMAP_OUT_CONVERT,
|
||||
};
|
||||
|
||||
typedef struct ChannelMapInfo {
|
||||
int channel_map[AVRESAMPLE_MAX_CHANNELS]; /**< source index of each output channel, -1 if not remapped */
|
||||
int do_remap; /**< remap needed */
|
||||
int channel_copy[AVRESAMPLE_MAX_CHANNELS]; /**< dest index to copy from */
|
||||
int do_copy; /**< copy needed */
|
||||
int channel_zero[AVRESAMPLE_MAX_CHANNELS]; /**< dest index to zero */
|
||||
int do_zero; /**< zeroing needed */
|
||||
int input_map[AVRESAMPLE_MAX_CHANNELS]; /**< dest index of each input channel */
|
||||
} ChannelMapInfo;
|
||||
|
||||
struct AVAudioResampleContext {
|
||||
const AVClass *av_class; /**< AVClass for logging and AVOptions */
|
||||
@ -64,6 +83,7 @@ struct AVAudioResampleContext {
|
||||
int resample_needed; /**< resampling is needed */
|
||||
int in_convert_needed; /**< input sample format conversion is needed */
|
||||
int out_convert_needed; /**< output sample format conversion is needed */
|
||||
int in_copy_needed; /**< input data copy is needed */
|
||||
|
||||
AudioData *in_buffer; /**< buffer for converted input */
|
||||
AudioData *resample_out_buffer; /**< buffer for output from resampler */
|
||||
@ -81,6 +101,10 @@ struct AVAudioResampleContext {
|
||||
* only used if avresample_set_matrix() is called before avresample_open()
|
||||
*/
|
||||
double *mix_matrix;
|
||||
|
||||
int use_channel_map;
|
||||
enum RemapPoint remap_point;
|
||||
ChannelMapInfo ch_map_info;
|
||||
};
|
||||
|
||||
#endif /* AVRESAMPLE_INTERNAL_H */
|
||||
|
@ -23,6 +23,7 @@
|
||||
#include "libavutil/libm.h"
|
||||
#include "libavutil/log.h"
|
||||
#include "internal.h"
|
||||
#include "resample.h"
|
||||
#include "audio_data.h"
|
||||
|
||||
struct ResampleContext {
|
||||
|
@ -22,10 +22,9 @@
|
||||
#define AVRESAMPLE_RESAMPLE_H
|
||||
|
||||
#include "avresample.h"
|
||||
#include "internal.h"
|
||||
#include "audio_data.h"
|
||||
|
||||
typedef struct ResampleContext ResampleContext;
|
||||
|
||||
/**
|
||||
* Allocate and initialize a ResampleContext.
|
||||
*
|
||||
|
@ -26,8 +26,11 @@
|
||||
#include "libavutil/opt.h"
|
||||
|
||||
#include "avresample.h"
|
||||
#include "audio_data.h"
|
||||
#include "internal.h"
|
||||
#include "audio_data.h"
|
||||
#include "audio_convert.h"
|
||||
#include "audio_mix.h"
|
||||
#include "resample.h"
|
||||
|
||||
int avresample_open(AVAudioResampleContext *avr)
|
||||
{
|
||||
@ -93,20 +96,84 @@ int avresample_open(AVAudioResampleContext *avr)
|
||||
av_get_sample_fmt_name(avr->internal_sample_fmt));
|
||||
}
|
||||
|
||||
/* set sample format conversion parameters */
|
||||
/* treat all mono as planar for easier comparison */
|
||||
if (avr->in_channels == 1)
|
||||
avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
|
||||
if (avr->out_channels == 1)
|
||||
avr->out_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
|
||||
avr->in_convert_needed = (avr->resample_needed || avr->mixing_needed) &&
|
||||
avr->in_sample_fmt != avr->internal_sample_fmt;
|
||||
|
||||
/* we may need to add an extra conversion in order to remap channels if
|
||||
the output format is not planar */
|
||||
if (avr->use_channel_map && !avr->mixing_needed && !avr->resample_needed &&
|
||||
!av_sample_fmt_is_planar(avr->out_sample_fmt)) {
|
||||
avr->internal_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
|
||||
}
|
||||
|
||||
/* set sample format conversion parameters */
|
||||
if (avr->resample_needed || avr->mixing_needed)
|
||||
avr->in_convert_needed = avr->in_sample_fmt != avr->internal_sample_fmt;
|
||||
else
|
||||
avr->in_convert_needed = avr->use_channel_map &&
|
||||
!av_sample_fmt_is_planar(avr->out_sample_fmt);
|
||||
|
||||
if (avr->resample_needed || avr->mixing_needed || avr->in_convert_needed)
|
||||
avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt;
|
||||
else
|
||||
avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt;
|
||||
|
||||
avr->in_copy_needed = !avr->in_convert_needed && (avr->mixing_needed ||
|
||||
(avr->use_channel_map && avr->resample_needed));
|
||||
|
||||
if (avr->use_channel_map) {
|
||||
if (avr->in_copy_needed) {
|
||||
avr->remap_point = REMAP_IN_COPY;
|
||||
av_dlog(avr, "remap channels during in_copy\n");
|
||||
} else if (avr->in_convert_needed) {
|
||||
avr->remap_point = REMAP_IN_CONVERT;
|
||||
av_dlog(avr, "remap channels during in_convert\n");
|
||||
} else if (avr->out_convert_needed) {
|
||||
avr->remap_point = REMAP_OUT_CONVERT;
|
||||
av_dlog(avr, "remap channels during out_convert\n");
|
||||
} else {
|
||||
avr->remap_point = REMAP_OUT_COPY;
|
||||
av_dlog(avr, "remap channels during out_copy\n");
|
||||
}
|
||||
|
||||
#ifdef DEBUG
|
||||
{
|
||||
int ch;
|
||||
av_dlog(avr, "output map: ");
|
||||
if (avr->ch_map_info.do_remap)
|
||||
for (ch = 0; ch < avr->in_channels; ch++)
|
||||
av_dlog(avr, " % 2d", avr->ch_map_info.channel_map[ch]);
|
||||
else
|
||||
av_dlog(avr, "n/a");
|
||||
av_dlog(avr, "\n");
|
||||
av_dlog(avr, "copy map: ");
|
||||
if (avr->ch_map_info.do_copy)
|
||||
for (ch = 0; ch < avr->in_channels; ch++)
|
||||
av_dlog(avr, " % 2d", avr->ch_map_info.channel_copy[ch]);
|
||||
else
|
||||
av_dlog(avr, "n/a");
|
||||
av_dlog(avr, "\n");
|
||||
av_dlog(avr, "zero map: ");
|
||||
if (avr->ch_map_info.do_zero)
|
||||
for (ch = 0; ch < avr->in_channels; ch++)
|
||||
av_dlog(avr, " % 2d", avr->ch_map_info.channel_zero[ch]);
|
||||
else
|
||||
av_dlog(avr, "n/a");
|
||||
av_dlog(avr, "\n");
|
||||
av_dlog(avr, "input map: ");
|
||||
for (ch = 0; ch < avr->in_channels; ch++)
|
||||
av_dlog(avr, " % 2d", avr->ch_map_info.input_map[ch]);
|
||||
av_dlog(avr, "\n");
|
||||
}
|
||||
#endif
|
||||
} else
|
||||
avr->remap_point = REMAP_NONE;
|
||||
|
||||
/* allocate buffers */
|
||||
if (avr->mixing_needed || avr->in_convert_needed) {
|
||||
if (avr->in_copy_needed || avr->in_convert_needed) {
|
||||
avr->in_buffer = ff_audio_data_alloc(FFMAX(avr->in_channels, avr->out_channels),
|
||||
0, avr->internal_sample_fmt,
|
||||
"in_buffer");
|
||||
@ -143,7 +210,8 @@ int avresample_open(AVAudioResampleContext *avr)
|
||||
if (avr->in_convert_needed) {
|
||||
avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt,
|
||||
avr->in_sample_fmt, avr->in_channels,
|
||||
avr->in_sample_rate);
|
||||
avr->in_sample_rate,
|
||||
avr->remap_point == REMAP_IN_CONVERT);
|
||||
if (!avr->ac_in) {
|
||||
ret = AVERROR(ENOMEM);
|
||||
goto error;
|
||||
@ -157,7 +225,8 @@ int avresample_open(AVAudioResampleContext *avr)
|
||||
src_fmt = avr->in_sample_fmt;
|
||||
avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt,
|
||||
avr->out_channels,
|
||||
avr->out_sample_rate);
|
||||
avr->out_sample_rate,
|
||||
avr->remap_point == REMAP_OUT_CONVERT);
|
||||
if (!avr->ac_out) {
|
||||
ret = AVERROR(ENOMEM);
|
||||
goto error;
|
||||
@ -197,6 +266,8 @@ void avresample_close(AVAudioResampleContext *avr)
|
||||
ff_audio_resample_free(&avr->resample);
|
||||
ff_audio_mix_free(&avr->am);
|
||||
av_freep(&avr->mix_matrix);
|
||||
|
||||
avr->use_channel_map = 0;
|
||||
}
|
||||
|
||||
void avresample_free(AVAudioResampleContext **avr)
|
||||
@ -239,7 +310,9 @@ static int handle_buffered_output(AVAudioResampleContext *avr,
|
||||
data in the output FIFO */
|
||||
av_dlog(avr, "[copy] %s to output\n", converted->name);
|
||||
output->nb_samples = 0;
|
||||
ret = ff_audio_data_copy(output, converted);
|
||||
ret = ff_audio_data_copy(output, converted,
|
||||
avr->remap_point == REMAP_OUT_COPY ?
|
||||
&avr->ch_map_info : NULL);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
av_dlog(avr, "[end conversion]\n");
|
||||
@ -303,11 +376,24 @@ int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
|
||||
/* in some rare cases we can copy input to output and upmix
|
||||
directly in the output buffer */
|
||||
av_dlog(avr, "[copy] %s to output\n", current_buffer->name);
|
||||
ret = ff_audio_data_copy(&output_buffer, current_buffer);
|
||||
ret = ff_audio_data_copy(&output_buffer, current_buffer,
|
||||
avr->remap_point == REMAP_OUT_COPY ?
|
||||
&avr->ch_map_info : NULL);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
current_buffer = &output_buffer;
|
||||
} else if (avr->mixing_needed || avr->in_convert_needed) {
|
||||
} else if (avr->remap_point == REMAP_OUT_COPY &&
|
||||
(!direct_output || out_samples < in_samples)) {
|
||||
/* if remapping channels during output copy, we may need to
|
||||
* use an intermediate buffer in order to remap before adding
|
||||
* samples to the output fifo */
|
||||
av_dlog(avr, "[copy] %s to out_buffer\n", current_buffer->name);
|
||||
ret = ff_audio_data_copy(avr->out_buffer, current_buffer,
|
||||
&avr->ch_map_info);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
current_buffer = avr->out_buffer;
|
||||
} else if (avr->in_copy_needed || avr->in_convert_needed) {
|
||||
/* if needed, copy or convert input to in_buffer, and downmix if
|
||||
applicable */
|
||||
if (avr->in_convert_needed) {
|
||||
@ -322,7 +408,9 @@ int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
|
||||
return ret;
|
||||
} else {
|
||||
av_dlog(avr, "[copy] %s to in_buffer\n", current_buffer->name);
|
||||
ret = ff_audio_data_copy(avr->in_buffer, current_buffer);
|
||||
ret = ff_audio_data_copy(avr->in_buffer, current_buffer,
|
||||
avr->remap_point == REMAP_IN_COPY ?
|
||||
&avr->ch_map_info : NULL);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
}
|
||||
@ -467,6 +555,57 @@ int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
|
||||
return 0;
|
||||
}
|
||||
|
||||
int avresample_set_channel_mapping(AVAudioResampleContext *avr,
|
||||
const int *channel_map)
|
||||
{
|
||||
ChannelMapInfo *info = &avr->ch_map_info;
|
||||
int in_channels, ch, i;
|
||||
|
||||
in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
|
||||
if (in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS) {
|
||||
av_log(avr, AV_LOG_ERROR, "Invalid input channel layout\n");
|
||||
return AVERROR(EINVAL);
|
||||
}
|
||||
|
||||
memset(info, 0, sizeof(*info));
|
||||
memset(info->input_map, -1, sizeof(info->input_map));
|
||||
|
||||
for (ch = 0; ch < in_channels; ch++) {
|
||||
if (channel_map[ch] >= in_channels) {
|
||||
av_log(avr, AV_LOG_ERROR, "Invalid channel map\n");
|
||||
return AVERROR(EINVAL);
|
||||
}
|
||||
if (channel_map[ch] < 0) {
|
||||
info->channel_zero[ch] = 1;
|
||||
info->channel_map[ch] = -1;
|
||||
info->do_zero = 1;
|
||||
} else if (info->input_map[channel_map[ch]] >= 0) {
|
||||
info->channel_copy[ch] = info->input_map[channel_map[ch]];
|
||||
info->channel_map[ch] = -1;
|
||||
info->do_copy = 1;
|
||||
} else {
|
||||
info->channel_map[ch] = channel_map[ch];
|
||||
info->input_map[channel_map[ch]] = ch;
|
||||
info->do_remap = 1;
|
||||
}
|
||||
}
|
||||
/* Fill-in unmapped input channels with unmapped output channels.
|
||||
This is used when remapping during conversion from interleaved to
|
||||
planar format. */
|
||||
for (ch = 0, i = 0; ch < in_channels && i < in_channels; ch++, i++) {
|
||||
while (ch < in_channels && info->input_map[ch] >= 0)
|
||||
ch++;
|
||||
while (i < in_channels && info->channel_map[i] >= 0)
|
||||
i++;
|
||||
if (ch >= in_channels || i >= in_channels)
|
||||
break;
|
||||
info->input_map[ch] = i;
|
||||
}
|
||||
|
||||
avr->use_channel_map = 1;
|
||||
return 0;
|
||||
}
|
||||
|
||||
int avresample_available(AVAudioResampleContext *avr)
|
||||
{
|
||||
return av_audio_fifo_size(avr->out_fifo);
|
||||
|
@ -20,8 +20,8 @@
|
||||
#define AVRESAMPLE_VERSION_H
|
||||
|
||||
#define LIBAVRESAMPLE_VERSION_MAJOR 1
|
||||
#define LIBAVRESAMPLE_VERSION_MINOR 0
|
||||
#define LIBAVRESAMPLE_VERSION_MICRO 1
|
||||
#define LIBAVRESAMPLE_VERSION_MINOR 1
|
||||
#define LIBAVRESAMPLE_VERSION_MICRO 0
|
||||
|
||||
#define LIBAVRESAMPLE_VERSION_INT AV_VERSION_INT(LIBAVRESAMPLE_VERSION_MAJOR, \
|
||||
LIBAVRESAMPLE_VERSION_MINOR, \
|
||||
|
Loading…
Reference in New Issue
Block a user