[FFmpeg-devel] [PATCH] examples/muxing: add support to audio resampling
Stefano Sabatini
stefasab at gmail.com
Tue Jul 2 18:21:09 CEST 2013
Allows to encode to output in case the destination sample format is
different from AV_SAMPLE_FMT_S16.
---
doc/examples/muxing.c | 116 +++++++++++++++++++++++++++++++++++++++-----------
1 file changed, 92 insertions(+), 24 deletions(-)
diff --git a/doc/examples/muxing.c b/doc/examples/muxing.c
index c4ffee8..9edcaa7 100644
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -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
@@ -50,8 +52,6 @@ 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,
@@ -78,7 +78,7 @@ 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;
@@ -125,9 +125,16 @@ 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 +156,51 @@ 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 +225,45 @@ 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 +289,8 @@ 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]);
}
/**************************************************************/
--
1.8.1.2
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