RepoMirrors/ffmpeg
1
0
Fork 0
mirror of https://git.ffmpeg.org/ffmpeg.git synced 2024-12-28 02:12:28 +00:00
Code Issues Projects Releases Wiki Activity

extend resampling API, add S16 internal conversion

Browse Source 
Originally committed as revision 17163 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Baptiste Coudurier 2009-02-11 22:57:10 +00:00
parent b5fdaebb44
commit d1e3c6fd40
3 changed files with 168 additions and 15 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

16
ffmpeg.c
View File

@ -555,12 +555,12 @@ static void do_audio_out(AVFormatContext *s,
ost->audio_resample = 1;
if (ost->audio_resample && !ost->resample) {
if (dec->sample_fmt != SAMPLE_FMT_S16) {
fprintf(stderr, "Audio resampler only works with 16 bits per sample, patch welcome.\n");
av_exit(1);
}
ost->resample = audio_resample_init(enc->channels, dec->channels,
enc->sample_rate, dec->sample_rate);
if (dec->sample_fmt != SAMPLE_FMT_S16)
fprintf(stderr, "Warning, using s16 intermediate sample format for resampling\n");
ost->resample = av_audio_resample_init(enc->channels, dec->channels,
enc->sample_rate, dec->sample_rate,
enc->sample_fmt, dec->sample_fmt,
16, 10, 0, 0.8);
if (!ost->resample) {
fprintf(stderr, "Can not resample %d channels @ %d Hz to %d channels @ %d Hz\n",
dec->channels, dec->sample_rate,
@ -570,7 +570,7 @@ static void do_audio_out(AVFormatContext *s,
}
#define MAKE_SFMT_PAIR(a,b) ((a)+SAMPLE_FMT_NB*(b))
if (dec->sample_fmt!=enc->sample_fmt &&
if (!ost->audio_resample && dec->sample_fmt!=enc->sample_fmt &&
MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt)!=ost->reformat_pair) {
if (!audio_out2)
audio_out2 = av_malloc(audio_out_size);
@ -647,7 +647,7 @@ static void do_audio_out(AVFormatContext *s,
size_out = size;
}
if (dec->sample_fmt!=enc->sample_fmt) {
if (!ost->audio_resample && dec->sample_fmt!=enc->sample_fmt) {
const void *ibuf[6]= {buftmp};
void *obuf[6]= {audio_out2};
int istride[6]= {isize};

34
libavcodec/avcodec.h
View File

@ -30,7 +30,7 @@
#include "libavutil/avutil.h"
#define LIBAVCODEC_VERSION_MAJOR 52
#define LIBAVCODEC_VERSION_MINOR 14
#define LIBAVCODEC_VERSION_MINOR 15
#define LIBAVCODEC_VERSION_MICRO 0
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
@ -2443,8 +2443,36 @@ struct AVResampleContext;
typedef struct ReSampleContext ReSampleContext;
ReSampleContext *audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate);
#if LIBAVCODEC_VERSION_MAJOR < 53
/**
* @deprecated Use av_audio_resample_init() instead.
*/
attribute_deprecated ReSampleContext *audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate);
#endif
/**
* Initializes audio resampling context
*
* @param output_channels number of output channels
* @param input_channels number of input channels
* @param output_rate output sample rate
* @param input_rate input sample rate
* @param sample_fmt_out requested output sample format
* @param sample_fmt_in input sample format
* @param filter_length length of each FIR filter in the filterbank relative to the cutoff freq
* @param log2_phase_count log2 of the number of entries in the polyphase filterbank
* @param linear If 1 then the used FIR filter will be linearly interpolated
between the 2 closest, if 0 the closest will be used
* @param cutoff cutoff frequency, 1.0 corresponds to half the output sampling rate
* @return allocated ReSampleContext, NULL if error occured
*/
ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate,
enum SampleFormat sample_fmt_out,
enum SampleFormat sample_fmt_in,
int filter_length, int log2_phase_count,
int linear, double cutoff);
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples);
void audio_resample_close(ReSampleContext *s);

133
libavcodec/resample.c
View File

@ -25,16 +25,32 @@
*/
#include "avcodec.h"
#include "audioconvert.h"
#include "opt.h"
struct AVResampleContext;
static const char *context_to_name(void *ptr)
{
return "audioresample";
}
static const AVOption options[] = {{NULL}};
static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options };
struct ReSampleContext {
const AVClass *av_class;
struct AVResampleContext *resample_context;
short *temp[2];
int temp_len;
float ratio;
/* channel convert */
int input_channels, output_channels, filter_channels;
AVAudioConvert *convert_ctx[2];
enum SampleFormat sample_fmt[2]; ///< input and output sample format
unsigned sample_size[2]; ///< size of one sample in sample_fmt
short *buffer[2]; ///< buffers used for conversion to S16
unsigned buffer_size[2]; ///< sizes of allocated buffers
};
/* n1: number of samples */
@ -126,8 +142,12 @@ static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
}
}
ReSampleContext *audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate)
ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate,
enum SampleFormat sample_fmt_out,
enum SampleFormat sample_fmt_in,
int filter_length, int log2_phase_count,
int linear, double cutoff)
{
ReSampleContext *s;
@ -153,6 +173,34 @@ ReSampleContext *audio_resample_init(int output_channels, int input_channels,
if (s->output_channels < s->filter_channels)
s->filter_channels = s->output_channels;
s->sample_fmt [0] = sample_fmt_in;
s->sample_fmt [1] = sample_fmt_out;
s->sample_size[0] = av_get_bits_per_sample_format(s->sample_fmt[0])>>3;
s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3;
if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1,
s->sample_fmt[0], 1, NULL, 0))) {
av_log(s, AV_LOG_ERROR,
"Cannot convert %s sample format to s16 sample format\n",
avcodec_get_sample_fmt_name(s->sample_fmt[0]));
av_free(s);
return NULL;
}
}
if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
SAMPLE_FMT_S16, 1, NULL, 0))) {
av_log(s, AV_LOG_ERROR,
"Cannot convert s16 sample format to %s sample format\n",
avcodec_get_sample_fmt_name(s->sample_fmt[1]));
av_audio_convert_free(s->convert_ctx[0]);
av_free(s);
return NULL;
}
}
/*
* AC-3 output is the only case where filter_channels could be greater than 2.
* input channels can't be greater than 2, so resample the 2 channels and then
@ -162,11 +210,25 @@ ReSampleContext *audio_resample_init(int output_channels, int input_channels,
s->filter_channels = 2;
#define TAPS 16
s->resample_context= av_resample_init(output_rate, input_rate, TAPS, 10, 0, 0.8);
s->resample_context= av_resample_init(output_rate, input_rate,
filter_length, log2_phase_count, linear, cutoff);
s->av_class= &audioresample_context_class;
return s;
}
#if LIBAVCODEC_VERSION_MAJOR < 53
ReSampleContext *audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate)
{
return av_audio_resample_init(output_channels, input_channels,
output_rate, input_rate,
SAMPLE_FMT_S16, SAMPLE_FMT_S16,
TAPS, 10, 0, 0.8);
}
#endif
/* resample audio. 'nb_samples' is the number of input samples */
/* XXX: optimize it ! */
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
@ -175,6 +237,7 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
short *bufin[2];
short *bufout[2];
short *buftmp2[2], *buftmp3[2];
short *output_bak = NULL;
int lenout;
if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
@ -183,6 +246,52 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
return nb_samples;
}
if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
int istride[1] = { s->sample_size[0] };
int ostride[1] = { 2 };
const void *ibuf[1] = { input };
void *obuf[1];
unsigned input_size = nb_samples*s->input_channels*s->sample_size[0];
if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
av_free(s->buffer[0]);
s->buffer_size[0] = input_size;
s->buffer[0] = av_malloc(s->buffer_size[0]);
if (!s->buffer[0]) {
av_log(s, AV_LOG_ERROR, "Could not allocate buffer\n");
return 0;
}
}
obuf[0] = s->buffer[0];
if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
ibuf, istride, nb_samples*s->input_channels) < 0) {
av_log(s, AV_LOG_ERROR, "Audio sample format conversion failed\n");
return 0;
}
input = s->buffer[0];
}
lenout= 4*nb_samples * s->ratio + 16;
if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
output_bak = output;
if (!s->buffer_size[1] || s->buffer_size[1] < lenout) {
av_free(s->buffer[1]);
s->buffer_size[1] = lenout;
s->buffer[1] = av_malloc(s->buffer_size[1]);
if (!s->buffer[1]) {
av_log(s, AV_LOG_ERROR, "Could not allocate buffer\n");
return 0;
}
}
output = s->buffer[1];
}
/* XXX: move those malloc to resample init code */
for(i=0; i<s->filter_channels; i++){
bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
@ -191,7 +300,6 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
}
/* make some zoom to avoid round pb */
lenout= 4*nb_samples * s->ratio + 16;
bufout[0]= av_malloc( lenout * sizeof(short) );
bufout[1]= av_malloc( lenout * sizeof(short) );
@ -233,6 +341,19 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
}
if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
int istride[1] = { 2 };
int ostride[1] = { s->sample_size[1] };
const void *ibuf[1] = { output };
void *obuf[1] = { output_bak };
if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
ibuf, istride, nb_samples1*s->output_channels) < 0) {
av_log(s, AV_LOG_ERROR, "Audio sample format convertion failed\n");
return 0;
}
}
for(i=0; i<s->filter_channels; i++)
av_free(bufin[i]);
@ -246,5 +367,9 @@ void audio_resample_close(ReSampleContext *s)
av_resample_close(s->resample_context);
av_freep(&s->temp[0]);
av_freep(&s->temp[1]);
av_freep(&s->buffer[0]);
av_freep(&s->buffer[1]);
av_audio_convert_free(s->convert_ctx[0]);
av_audio_convert_free(s->convert_ctx[1]);
av_free(s);
}