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ffmpeg.wasm-core
Commit 2aa2b4ac authored by Stefano Sabatini's avatar Stefano Sabatini
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examples/muxing: add support to audio resampling 

Allows to encode to output in case the destination sample format is
different from AV_SAMPLE_FMT_S16.
parent 561e0513
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Showing with 93 additions and 28 deletions
doc/examples/muxing.c
View file @ 2aa2b4ac
......@@ -34,9 +34,11 @@
#include <string.h>
#include <math.h>
#include <libavutil/opt.h>
#include <libavutil/mathematics.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.h>
/* 5 seconds stream duration */
#define STREAM_DURATION 200.0
......@@ -46,13 +48,6 @@
static int sws_flags = SWS_BICUBIC;
/**************************************************************/
/* audio output */
static float t, tincr, tincr2;
static int16_t *samples;
static int audio_input_frame_size;
/* Add an output stream. */
static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
enum AVCodecID codec_id)
......@@ -78,7 +73,7 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
switch ((*codec)->type) {
case AVMEDIA_TYPE_AUDIO:
c->sample_fmt = AV_SAMPLE_FMT_S16;
c->sample_fmt = AV_SAMPLE_FMT_FLTP;
c->bit_rate = 64000;
c->sample_rate = 44100;
c->channels = 2;
......@@ -126,8 +121,17 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
/* audio output */
static float t, tincr, tincr2;
static int16_t *samples;
static int audio_input_frame_size;
static uint8_t **src_samples_data;
static int src_samples_linesize;
static int src_nb_samples;
static int max_dst_nb_samples;
uint8_t **dst_samples_data;
int dst_samples_linesize;
int dst_samples_size;
struct SwrContext *swr_ctx = NULL;
static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
{
......@@ -149,17 +153,51 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
/* increment frequency by 110 Hz per second */
tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
audio_input_frame_size = 10000;
else
audio_input_frame_size = c->frame_size;
samples = av_malloc(audio_input_frame_size *
av_get_bytes_per_sample(c->sample_fmt) *
c->channels);
if (!samples) {
fprintf(stderr, "Could not allocate audio samples buffer\n");
src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
10000 : c->frame_size;
ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
src_nb_samples, c->sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate source samples\n");
exit(1);
}
/* create resampler context */
if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
/* set options */
av_opt_set_int (swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int (swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int (swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int (swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
}
}
/* compute the number of converted samples: buffering is avoided
* ensuring that the output buffer will contain at least all the
* converted input samples */
max_dst_nb_samples = src_nb_samples;
ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
max_dst_nb_samples, c->sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
exit(1);
}
dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
c->sample_fmt, 0);
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
......@@ -184,18 +222,45 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
AVFrame *frame = avcodec_alloc_frame();
int got_packet, ret;
int got_packet, ret, dst_nb_samples;
av_init_packet(&pkt);
c = st->codec;
get_audio_frame(samples, audio_input_frame_size, c->channels);
frame->nb_samples = audio_input_frame_size;
get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
/* convert samples from native format to destination codec format, using the resampler */
if (swr_ctx) {
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_free(dst_samples_data[0]);
ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
dst_nb_samples, c->sample_fmt, 0);
if (ret < 0)
exit(1);
max_dst_nb_samples = dst_nb_samples;
dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
c->sample_fmt, 0);
}
/* convert to destination format */
ret = swr_convert(swr_ctx,
dst_samples_data, dst_nb_samples,
(const uint8_t **)src_samples_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
} else {
dst_samples_data[0] = src_samples_data[0];
dst_nb_samples = src_nb_samples;
}
frame->nb_samples = dst_nb_samples;
avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
(uint8_t *)samples,
audio_input_frame_size *
av_get_bytes_per_sample(c->sample_fmt) *
c->channels, 1);
dst_samples_data[0], dst_samples_size, 0);
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (ret < 0) {
......@@ -221,8 +286,8 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
static void close_audio(AVFormatContext *oc, AVStream *st)
{
avcodec_close(st->codec);
av_free(samples);
av_free(src_samples_data[0]);
av_free(dst_samples_data[0]);
}
/**************************************************************/
......
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