[FFmpeg-cvslog] output example: convert audio to the format supported by the encoder
Anton Khirnov
git at videolan.org
Sat Jul 26 23:38:18 CEST 2014
ffmpeg | branch: master | Anton Khirnov <anton at khirnov.net> | Thu Jul 24 17:47:26 2014 +0000| [56f98e340fca894a76d1ddbe33118b8d8c4db34a] | committer: Anton Khirnov
output example: convert audio to the format supported by the encoder
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=56f98e340fca894a76d1ddbe33118b8d8c4db34a
---
doc/examples/output.c | 193 +++++++++++++++++++++++++++++++++++++------------
1 file changed, 147 insertions(+), 46 deletions(-)
diff --git a/doc/examples/output.c b/doc/examples/output.c
index 239fe5b..0534554 100644
--- a/doc/examples/output.c
+++ b/doc/examples/output.c
@@ -36,7 +36,9 @@
#include "libavutil/channel_layout.h"
#include "libavutil/mathematics.h"
+#include "libavutil/opt.h"
#include "libavformat/avformat.h"
+#include "libavresample/avresample.h"
#include "libswscale/swscale.h"
/* 5 seconds stream duration */
@@ -60,6 +62,7 @@ typedef struct OutputStream {
float t, tincr, tincr2;
struct SwsContext *sws_ctx;
+ AVAudioResampleContext *avr;
} OutputStream;
/**************************************************************/
@@ -73,6 +76,7 @@ static void add_audio_stream(OutputStream *ost, AVFormatContext *oc,
{
AVCodecContext *c;
AVCodec *codec;
+ int ret;
/* find the audio encoder */
codec = avcodec_find_encoder(codec_id);
@@ -90,23 +94,75 @@ static void add_audio_stream(OutputStream *ost, AVFormatContext *oc,
c = ost->st->codec;
/* put sample parameters */
- c->sample_fmt = AV_SAMPLE_FMT_S16;
- c->bit_rate = 64000;
- c->sample_rate = 44100;
- c->channels = 2;
- c->channel_layout = AV_CH_LAYOUT_STEREO;
+ c->sample_fmt = codec->sample_fmts ? codec->sample_fmts[0] : AV_SAMPLE_FMT_S16;
+ c->sample_rate = codec->supported_samplerates ? codec->supported_samplerates[0] : 44100;
+ c->channel_layout = codec->channel_layouts ? codec->channel_layouts[0] : AV_CH_LAYOUT_STEREO;
+ c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
+ c->bit_rate = 64000;
ost->st->time_base = (AVRational){ 1, c->sample_rate };
// some formats want stream headers to be separate
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
+
+ /* initialize sample format conversion;
+ * to simplify the code, we always pass the data through lavr, even
+ * if the encoder supports the generated format directly -- the price is
+ * some extra data copying;
+ */
+ ost->avr = avresample_alloc_context();
+ if (!ost->avr) {
+ fprintf(stderr, "Error allocating the resampling context\n");
+ exit(1);
+ }
+
+ av_opt_set_int(ost->avr, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
+ av_opt_set_int(ost->avr, "in_sample_rate", 44100, 0);
+ av_opt_set_int(ost->avr, "in_channel_layout", AV_CH_LAYOUT_STEREO, 0);
+ av_opt_set_int(ost->avr, "out_sample_fmt", c->sample_fmt, 0);
+ av_opt_set_int(ost->avr, "out_sample_rate", c->sample_rate, 0);
+ av_opt_set_int(ost->avr, "out_channel_layout", c->channel_layout, 0);
+
+ ret = avresample_open(ost->avr);
+ if (ret < 0) {
+ fprintf(stderr, "Error opening the resampling context\n");
+ exit(1);
+ }
+}
+
+static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
+ uint64_t channel_layout,
+ int sample_rate, int nb_samples)
+{
+ AVFrame *frame = av_frame_alloc();
+ int ret;
+
+ if (!frame) {
+ fprintf(stderr, "Error allocating an audio frame\n");
+ exit(1);
+ }
+
+ frame->format = sample_fmt;
+ frame->channel_layout = channel_layout;
+ frame->sample_rate = sample_rate;
+ frame->nb_samples = nb_samples;
+
+ if (nb_samples) {
+ ret = av_frame_get_buffer(frame, 0);
+ if (ret < 0) {
+ fprintf(stderr, "Error allocating an audio buffer\n");
+ exit(1);
+ }
+ }
+
+ return frame;
}
static void open_audio(AVFormatContext *oc, OutputStream *ost)
{
AVCodecContext *c;
- int ret;
+ int nb_samples;
c = ost->st->codec;
@@ -122,47 +178,32 @@ static void open_audio(AVFormatContext *oc, OutputStream *ost)
/* increment frequency by 110 Hz per second */
ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
- ost->frame = av_frame_alloc();
- if (!ost->frame)
- exit(1);
-
- ost->frame->sample_rate = c->sample_rate;
- ost->frame->format = AV_SAMPLE_FMT_S16;
- ost->frame->channel_layout = c->channel_layout;
-
if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
- ost->frame->nb_samples = 10000;
+ nb_samples = 10000;
else
- ost->frame->nb_samples = c->frame_size;
+ nb_samples = c->frame_size;
- ret = av_frame_get_buffer(ost->frame, 0);
- if (ret < 0) {
- fprintf(stderr, "Could not allocate an audio frame.\n");
- exit(1);
- }
+ ost->frame = alloc_audio_frame(c->sample_fmt, c->channel_layout,
+ c->sample_rate, nb_samples);
+ ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, AV_CH_LAYOUT_STEREO,
+ 44100, nb_samples);
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
* 'nb_channels' channels. */
static AVFrame *get_audio_frame(OutputStream *ost)
{
- int j, i, v, ret;
- int16_t *q = (int16_t*)ost->frame->data[0];
+ AVFrame *frame = ost->tmp_frame;
+ int j, i, v;
+ int16_t *q = (int16_t*)frame->data[0];
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL;
- /* when we pass a frame to the encoder, it may keep a reference to it
- * internally;
- * make sure we do not overwrite it here
- */
- ret = av_frame_make_writable(ost->frame);
- if (ret < 0)
- exit(1);
- for (j = 0; j < ost->frame->nb_samples; j++) {
+ for (j = 0; j < frame->nb_samples; j++) {
v = (int)(sin(ost->t) * 10000);
for (i = 0; i < ost->st->codec->channels; i++)
*q++ = v;
@@ -170,33 +211,26 @@ static AVFrame *get_audio_frame(OutputStream *ost)
ost->tincr += ost->tincr2;
}
- ost->frame->pts = ost->next_pts;
- ost->next_pts += ost->frame->nb_samples;
-
- return ost->frame;
+ return frame;
}
-/*
- * encode one audio frame and send it to the muxer
+/* if a frame is provided, send it to the encoder, otherwise flush the encoder;
* return 1 when encoding is finished, 0 otherwise
*/
-static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
+static int encode_audio_frame(AVFormatContext *oc, OutputStream *ost,
+ AVFrame *frame)
{
- AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
- AVFrame *frame;
int got_packet;
av_init_packet(&pkt);
- c = ost->st->codec;
-
- frame = get_audio_frame(ost);
-
- avcodec_encode_audio2(c, &pkt, frame, &got_packet);
+ avcodec_encode_audio2(ost->st->codec, &pkt, frame, &got_packet);
if (got_packet) {
pkt.stream_index = ost->st->index;
+ av_packet_rescale_ts(&pkt, ost->st->codec->time_base, ost->st->time_base);
+
/* Write the compressed frame to the media file. */
if (av_interleaved_write_frame(oc, &pkt) != 0) {
fprintf(stderr, "Error while writing audio frame\n");
@@ -207,6 +241,72 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
return (frame || got_packet) ? 0 : 1;
}
+/*
+ * encode one audio frame and send it to the muxer
+ * return 1 when encoding is finished, 0 otherwise
+ */
+static int process_audio_stream(AVFormatContext *oc, OutputStream *ost)
+{
+ AVFrame *frame;
+ int got_output = 0;
+ int ret;
+
+ frame = get_audio_frame(ost);
+ got_output |= !!frame;
+
+ /* feed the data to lavr */
+ if (frame) {
+ ret = avresample_convert(ost->avr, NULL, 0, 0,
+ frame->extended_data, frame->linesize[0],
+ frame->nb_samples);
+ if (ret < 0) {
+ fprintf(stderr, "Error feeding audio data to the resampler\n");
+ exit(1);
+ }
+ }
+
+ while ((frame && avresample_available(ost->avr) >= ost->frame->nb_samples) ||
+ (!frame && avresample_get_out_samples(ost->avr, 0))) {
+ /* when we pass a frame to the encoder, it may keep a reference to it
+ * internally;
+ * make sure we do not overwrite it here
+ */
+ ret = av_frame_make_writable(ost->frame);
+ if (ret < 0)
+ exit(1);
+
+ /* the difference between the two avresample calls here is that the
+ * first one just reads the already converted data that is buffered in
+ * the lavr output buffer, while the second one also flushes the
+ * resampler */
+ if (frame) {
+ ret = avresample_read(ost->avr, ost->frame->extended_data,
+ ost->frame->nb_samples);
+ } else {
+ ret = avresample_convert(ost->avr, ost->frame->extended_data,
+ ost->frame->linesize[0], ost->frame->nb_samples,
+ NULL, 0, 0);
+ }
+
+ if (ret < 0) {
+ fprintf(stderr, "Error while resampling\n");
+ exit(1);
+ } else if (frame && ret != ost->frame->nb_samples) {
+ fprintf(stderr, "Too few samples returned from lavr\n");
+ exit(1);
+ }
+
+ ost->frame->nb_samples = ret;
+
+ ost->frame->pts = ost->next_pts;
+ ost->next_pts += ost->frame->nb_samples;
+
+ got_output |= encode_audio_frame(oc, ost, ret ? ost->frame : NULL);
+ }
+
+ return !got_output;
+}
+
/**************************************************************/
/* video output */
@@ -447,6 +547,7 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
av_frame_free(&ost->frame);
av_frame_free(&ost->tmp_frame);
sws_freeContext(ost->sws_ctx);
+ avresample_free(&ost->avr);
}
/**************************************************************/
@@ -535,7 +636,7 @@ int main(int argc, char **argv)
audio_st.next_pts, audio_st.st->codec->time_base) <= 0)) {
encode_video = !write_video_frame(oc, &video_st);
} else {
- encode_audio = !write_audio_frame(oc, &audio_st);
+ encode_audio = !process_audio_stream(oc, &audio_st);
}
}
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