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https://github.com/FFmpeg/FFmpeg.git
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output example: convert audio to the format supported by the encoder
This commit is contained in:
Anton Khirnov
parent
884f7c975f
commit
56f98e340f
@ -36,7 +36,9 @@
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#include "libavutil/channel_layout.h"
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#include "libavutil/mathematics.h"
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#include "libavutil/opt.h"
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#include "libavformat/avformat.h"
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#include "libavresample/avresample.h"
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#include "libswscale/swscale.h"
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/* 5 seconds stream duration */
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@ -60,6 +62,7 @@ typedef struct OutputStream {
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float t, tincr, tincr2;
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struct SwsContext *sws_ctx;
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AVAudioResampleContext *avr;
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} OutputStream;
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/**************************************************************/
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@ -73,6 +76,7 @@ static void add_audio_stream(OutputStream *ost, AVFormatContext *oc,
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{
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AVCodecContext *c;
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AVCodec *codec;
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int ret;
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/* find the audio encoder */
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codec = avcodec_find_encoder(codec_id);
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@ -90,23 +94,75 @@ static void add_audio_stream(OutputStream *ost, AVFormatContext *oc,
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c = ost->st->codec;
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/* put sample parameters */
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c->sample_fmt = AV_SAMPLE_FMT_S16;
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c->sample_fmt = codec->sample_fmts ? codec->sample_fmts[0] : AV_SAMPLE_FMT_S16;
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c->sample_rate = codec->supported_samplerates ? codec->supported_samplerates[0] : 44100;
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c->channel_layout = codec->channel_layouts ? codec->channel_layouts[0] : AV_CH_LAYOUT_STEREO;
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c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
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c->bit_rate = 64000;
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c->sample_rate = 44100;
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c->channels = 2;
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c->channel_layout = AV_CH_LAYOUT_STEREO;
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ost->st->time_base = (AVRational){ 1, c->sample_rate };
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// some formats want stream headers to be separate
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if (oc->oformat->flags & AVFMT_GLOBALHEADER)
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c->flags |= CODEC_FLAG_GLOBAL_HEADER;
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/* initialize sample format conversion;
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* to simplify the code, we always pass the data through lavr, even
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* if the encoder supports the generated format directly -- the price is
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* some extra data copying;
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*/
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ost->avr = avresample_alloc_context();
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if (!ost->avr) {
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fprintf(stderr, "Error allocating the resampling context\n");
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exit(1);
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}
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av_opt_set_int(ost->avr, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
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av_opt_set_int(ost->avr, "in_sample_rate", 44100, 0);
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av_opt_set_int(ost->avr, "in_channel_layout", AV_CH_LAYOUT_STEREO, 0);
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av_opt_set_int(ost->avr, "out_sample_fmt", c->sample_fmt, 0);
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av_opt_set_int(ost->avr, "out_sample_rate", c->sample_rate, 0);
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av_opt_set_int(ost->avr, "out_channel_layout", c->channel_layout, 0);
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ret = avresample_open(ost->avr);
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if (ret < 0) {
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fprintf(stderr, "Error opening the resampling context\n");
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exit(1);
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}
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}
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static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
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uint64_t channel_layout,
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int sample_rate, int nb_samples)
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{
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AVFrame *frame = av_frame_alloc();
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int ret;
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if (!frame) {
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fprintf(stderr, "Error allocating an audio frame\n");
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exit(1);
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}
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frame->format = sample_fmt;
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frame->channel_layout = channel_layout;
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frame->sample_rate = sample_rate;
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frame->nb_samples = nb_samples;
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if (nb_samples) {
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ret = av_frame_get_buffer(frame, 0);
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if (ret < 0) {
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fprintf(stderr, "Error allocating an audio buffer\n");
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exit(1);
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}
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}
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return frame;
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}
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static void open_audio(AVFormatContext *oc, OutputStream *ost)
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{
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AVCodecContext *c;
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int ret;
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int nb_samples;
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c = ost->st->codec;
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@ -122,47 +178,32 @@ static void open_audio(AVFormatContext *oc, OutputStream *ost)
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/* increment frequency by 110 Hz per second */
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ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
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ost->frame = av_frame_alloc();
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if (!ost->frame)
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exit(1);
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ost->frame->sample_rate = c->sample_rate;
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ost->frame->format = AV_SAMPLE_FMT_S16;
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ost->frame->channel_layout = c->channel_layout;
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if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
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ost->frame->nb_samples = 10000;
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nb_samples = 10000;
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else
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ost->frame->nb_samples = c->frame_size;
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nb_samples = c->frame_size;
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ret = av_frame_get_buffer(ost->frame, 0);
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if (ret < 0) {
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fprintf(stderr, "Could not allocate an audio frame.\n");
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exit(1);
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}
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ost->frame = alloc_audio_frame(c->sample_fmt, c->channel_layout,
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c->sample_rate, nb_samples);
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ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, AV_CH_LAYOUT_STEREO,
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44100, nb_samples);
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}
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/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
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* 'nb_channels' channels. */
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static AVFrame *get_audio_frame(OutputStream *ost)
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{
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int j, i, v, ret;
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int16_t *q = (int16_t*)ost->frame->data[0];
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AVFrame *frame = ost->tmp_frame;
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int j, i, v;
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int16_t *q = (int16_t*)frame->data[0];
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/* check if we want to generate more frames */
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if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
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STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
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return NULL;
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/* when we pass a frame to the encoder, it may keep a reference to it
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* internally;
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* make sure we do not overwrite it here
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*/
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ret = av_frame_make_writable(ost->frame);
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if (ret < 0)
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exit(1);
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for (j = 0; j < ost->frame->nb_samples; j++) {
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for (j = 0; j < frame->nb_samples; j++) {
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v = (int)(sin(ost->t) * 10000);
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for (i = 0; i < ost->st->codec->channels; i++)
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*q++ = v;
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@ -170,33 +211,26 @@ static AVFrame *get_audio_frame(OutputStream *ost)
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ost->tincr += ost->tincr2;
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}
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ost->frame->pts = ost->next_pts;
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ost->next_pts += ost->frame->nb_samples;
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return ost->frame;
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return frame;
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}
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/*
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* encode one audio frame and send it to the muxer
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/* if a frame is provided, send it to the encoder, otherwise flush the encoder;
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* return 1 when encoding is finished, 0 otherwise
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*/
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static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
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static int encode_audio_frame(AVFormatContext *oc, OutputStream *ost,
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AVFrame *frame)
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{
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AVCodecContext *c;
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AVPacket pkt = { 0 }; // data and size must be 0;
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AVFrame *frame;
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int got_packet;
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av_init_packet(&pkt);
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c = ost->st->codec;
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frame = get_audio_frame(ost);
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avcodec_encode_audio2(c, &pkt, frame, &got_packet);
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avcodec_encode_audio2(ost->st->codec, &pkt, frame, &got_packet);
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if (got_packet) {
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pkt.stream_index = ost->st->index;
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av_packet_rescale_ts(&pkt, ost->st->codec->time_base, ost->st->time_base);
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/* Write the compressed frame to the media file. */
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if (av_interleaved_write_frame(oc, &pkt) != 0) {
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fprintf(stderr, "Error while writing audio frame\n");
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@ -207,6 +241,72 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
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return (frame || got_packet) ? 0 : 1;
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}
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/*
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* encode one audio frame and send it to the muxer
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* return 1 when encoding is finished, 0 otherwise
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*/
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static int process_audio_stream(AVFormatContext *oc, OutputStream *ost)
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{
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AVFrame *frame;
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int got_output = 0;
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int ret;
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frame = get_audio_frame(ost);
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got_output |= !!frame;
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/* feed the data to lavr */
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if (frame) {
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ret = avresample_convert(ost->avr, NULL, 0, 0,
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frame->extended_data, frame->linesize[0],
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frame->nb_samples);
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if (ret < 0) {
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fprintf(stderr, "Error feeding audio data to the resampler\n");
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exit(1);
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}
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}
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while ((frame && avresample_available(ost->avr) >= ost->frame->nb_samples) ||
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(!frame && avresample_get_out_samples(ost->avr, 0))) {
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/* when we pass a frame to the encoder, it may keep a reference to it
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* internally;
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* make sure we do not overwrite it here
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*/
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ret = av_frame_make_writable(ost->frame);
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if (ret < 0)
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exit(1);
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/* the difference between the two avresample calls here is that the
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* first one just reads the already converted data that is buffered in
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* the lavr output buffer, while the second one also flushes the
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* resampler */
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if (frame) {
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ret = avresample_read(ost->avr, ost->frame->extended_data,
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ost->frame->nb_samples);
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} else {
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ret = avresample_convert(ost->avr, ost->frame->extended_data,
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ost->frame->linesize[0], ost->frame->nb_samples,
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NULL, 0, 0);
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}
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if (ret < 0) {
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fprintf(stderr, "Error while resampling\n");
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exit(1);
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} else if (frame && ret != ost->frame->nb_samples) {
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fprintf(stderr, "Too few samples returned from lavr\n");
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exit(1);
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}
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ost->frame->nb_samples = ret;
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ost->frame->pts = ost->next_pts;
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ost->next_pts += ost->frame->nb_samples;
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got_output |= encode_audio_frame(oc, ost, ret ? ost->frame : NULL);
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}
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return !got_output;
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}
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/**************************************************************/
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/* video output */
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@ -447,6 +547,7 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
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av_frame_free(&ost->frame);
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av_frame_free(&ost->tmp_frame);
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sws_freeContext(ost->sws_ctx);
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avresample_free(&ost->avr);
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}
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/**************************************************************/
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@ -535,7 +636,7 @@ int main(int argc, char **argv)
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audio_st.next_pts, audio_st.st->codec->time_base) <= 0)) {
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encode_video = !write_video_frame(oc, &video_st);
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} else {
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encode_audio = !write_audio_frame(oc, &audio_st);
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encode_audio = !process_audio_stream(oc, &audio_st);
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}
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}
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