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output example: convert audio to the format supported by the encoder

tags/n2.4
Anton Khirnov 11 years ago
parent
884f7c975f
commit
56f98e340f
1 changed files with 147 additions and 46 deletions
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  1. +147
    -46
      doc/examples/output.c

+ 147
- 46
doc/examples/output.c View File

@@ -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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