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https://github.com/FFmpeg/FFmpeg.git
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output example: use OutputStream for audio streams as well
This commit is contained in:
Anton Khirnov
parent
5e7b125b6a
commit
edd5f95764
@ -53,21 +53,21 @@ typedef struct OutputStream {
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AVFrame *frame;
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AVFrame *tmp_frame;
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float t, tincr, tincr2;
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int audio_input_frame_size;
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} OutputStream;
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/**************************************************************/
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/* audio output */
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static float t, tincr, tincr2;
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static int audio_input_frame_size;
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/*
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* add an audio output stream
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*/
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static AVStream *add_audio_stream(AVFormatContext *oc, enum AVCodecID codec_id)
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static void add_audio_stream(OutputStream *ost, AVFormatContext *oc,
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enum AVCodecID codec_id)
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{
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AVCodecContext *c;
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AVStream *st;
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AVCodec *codec;
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/* find the audio encoder */
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@ -77,13 +77,13 @@ static AVStream *add_audio_stream(AVFormatContext *oc, enum AVCodecID codec_id)
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exit(1);
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}
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st = avformat_new_stream(oc, codec);
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if (!st) {
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ost->st = avformat_new_stream(oc, codec);
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if (!ost->st) {
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fprintf(stderr, "Could not alloc stream\n");
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exit(1);
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}
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c = st->codec;
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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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@ -95,15 +95,13 @@ static AVStream *add_audio_stream(AVFormatContext *oc, enum AVCodecID codec_id)
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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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return st;
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}
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static void open_audio(AVFormatContext *oc, AVStream *st)
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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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c = st->codec;
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c = ost->st->codec;
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/* open it */
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if (avcodec_open2(c, NULL, NULL) < 0) {
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@ -112,20 +110,20 @@ static void open_audio(AVFormatContext *oc, AVStream *st)
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}
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/* init signal generator */
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t = 0;
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tincr = 2 * M_PI * 110.0 / c->sample_rate;
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ost->t = 0;
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ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
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/* increment frequency by 110 Hz per second */
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tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
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ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
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if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
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audio_input_frame_size = 10000;
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ost->audio_input_frame_size = 10000;
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else
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audio_input_frame_size = c->frame_size;
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ost->audio_input_frame_size = c->frame_size;
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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 void get_audio_frame(AVFrame *frame, int nb_channels)
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static void get_audio_frame(OutputStream *ost, AVFrame *frame, int nb_channels)
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{
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int j, i, v, ret;
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int16_t *q = (int16_t*)frame->data[0];
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@ -139,15 +137,15 @@ static void get_audio_frame(AVFrame *frame, int nb_channels)
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exit(1);
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for (j = 0; j < frame->nb_samples; j++) {
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v = (int)(sin(t) * 10000);
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v = (int)(sin(ost->t) * 10000);
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for (i = 0; i < nb_channels; i++)
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*q++ = v;
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t += tincr;
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tincr += tincr2;
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ost->t += ost->tincr;
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ost->tincr += ost->tincr2;
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}
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}
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static void write_audio_frame(AVFormatContext *oc, AVStream *st)
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static void write_audio_frame(AVFormatContext *oc, OutputStream *ost)
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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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@ -155,10 +153,10 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
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int got_packet, ret;
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av_init_packet(&pkt);
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c = st->codec;
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c = ost->st->codec;
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frame->sample_rate = c->sample_rate;
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frame->nb_samples = audio_input_frame_size;
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frame->nb_samples = ost->audio_input_frame_size;
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frame->format = AV_SAMPLE_FMT_S16;
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frame->channel_layout = c->channel_layout;
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ret = av_frame_get_buffer(frame, 0);
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@ -167,13 +165,13 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
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exit(1);
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}
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get_audio_frame(frame, c->channels);
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get_audio_frame(ost, frame, c->channels);
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avcodec_encode_audio2(c, &pkt, frame, &got_packet);
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if (!got_packet)
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return;
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pkt.stream_index = st->index;
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pkt.stream_index = ost->st->index;
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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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@ -183,9 +181,9 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
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av_frame_free(&frame);
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}
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static void close_audio(AVFormatContext *oc, AVStream *st)
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static void close_audio(AVFormatContext *oc, OutputStream *ost)
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{
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avcodec_close(st->codec);
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avcodec_close(ost->st->codec);
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}
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/**************************************************************/
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@ -413,13 +411,12 @@ static void close_video(AVFormatContext *oc, OutputStream *ost)
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int main(int argc, char **argv)
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{
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OutputStream video_st;
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OutputStream video_st, audio_st;
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const char *filename;
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AVOutputFormat *fmt;
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AVFormatContext *oc;
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AVStream *audio_st;
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double audio_pts, video_pts;
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int have_video = 0;
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int have_video = 0, have_audio = 0;
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int i;
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/* Initialize libavcodec, and register all codecs and formats. */
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@ -458,21 +455,21 @@ int main(int argc, char **argv)
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/* Add the audio and video streams using the default format codecs
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* and initialize the codecs. */
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audio_st = NULL;
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if (fmt->video_codec != AV_CODEC_ID_NONE) {
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add_video_stream(&video_st, oc, fmt->video_codec);
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have_video = 1;
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}
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if (fmt->audio_codec != AV_CODEC_ID_NONE) {
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audio_st = add_audio_stream(oc, fmt->audio_codec);
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add_audio_stream(&audio_st, oc, fmt->audio_codec);
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have_audio = 1;
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}
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/* Now that all the parameters are set, we can open the audio and
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* video codecs and allocate the necessary encode buffers. */
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if (have_video)
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open_video(oc, &video_st);
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if (audio_st)
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open_audio(oc, audio_st);
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if (have_audio)
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open_audio(oc, &audio_st);
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av_dump_format(oc, 0, filename, 1);
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@ -489,8 +486,8 @@ int main(int argc, char **argv)
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for (;;) {
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/* Compute current audio and video time. */
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if (audio_st)
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audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
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if (have_audio)
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audio_pts = (double)audio_st.st->pts.val * audio_st.st->time_base.num / audio_st.st->time_base.den;
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else
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audio_pts = 0.0;
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@ -500,13 +497,13 @@ int main(int argc, char **argv)
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else
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video_pts = 0.0;
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if ((!audio_st || audio_pts >= STREAM_DURATION) &&
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if ((!have_audio || audio_pts >= STREAM_DURATION) &&
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(!have_video || video_pts >= STREAM_DURATION))
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break;
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/* write interleaved audio and video frames */
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if (!have_video || (have_video && audio_st && audio_pts < video_pts)) {
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write_audio_frame(oc, audio_st);
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if (!have_video || (have_video && have_audio && audio_pts < video_pts)) {
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write_audio_frame(oc, &audio_st);
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} else {
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write_video_frame(oc, &video_st);
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}
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@ -521,8 +518,8 @@ int main(int argc, char **argv)
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/* Close each codec. */
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if (have_video)
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close_video(oc, &video_st);
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if (audio_st)
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close_audio(oc, audio_st);
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if (have_audio)
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close_audio(oc, &audio_st);
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/* Free the streams. */
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for (i = 0; i < oc->nb_streams; i++) {
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