diff options
author | Anton Khirnov <anton@khirnov.net> | 2014-06-24 10:56:27 +0400 |
---|---|---|
committer | Anton Khirnov <anton@khirnov.net> | 2014-06-26 18:07:02 +0400 |
commit | edd5f957646dcbf1bb55718bc7bf1e5481c25bcb (patch) | |
tree | 763c5a698dae393e812bf47776d4ec70b8204934 | |
parent | 5e7b125b6ae36893dfd9cb5661c99b67363cbb38 (diff) |
output example: use OutputStream for audio streams as well
-rw-r--r-- | doc/examples/output.c | 81 |
1 files changed, 39 insertions, 42 deletions
diff --git a/doc/examples/output.c b/doc/examples/output.c index 1f8ff9642f..768f1b664c 100644 --- a/doc/examples/output.c +++ b/doc/examples/output.c @@ -53,21 +53,21 @@ typedef struct OutputStream { AVFrame *frame; AVFrame *tmp_frame; + + float t, tincr, tincr2; + int audio_input_frame_size; } OutputStream; /**************************************************************/ /* audio output */ -static float t, tincr, tincr2; -static int audio_input_frame_size; - /* * add an audio output stream */ -static AVStream *add_audio_stream(AVFormatContext *oc, enum AVCodecID codec_id) +static void add_audio_stream(OutputStream *ost, AVFormatContext *oc, + enum AVCodecID codec_id) { AVCodecContext *c; - AVStream *st; AVCodec *codec; /* find the audio encoder */ @@ -77,13 +77,13 @@ static AVStream *add_audio_stream(AVFormatContext *oc, enum AVCodecID codec_id) exit(1); } - st = avformat_new_stream(oc, codec); - if (!st) { + ost->st = avformat_new_stream(oc, codec); + if (!ost->st) { fprintf(stderr, "Could not alloc stream\n"); exit(1); } - c = st->codec; + c = ost->st->codec; /* put sample parameters */ c->sample_fmt = AV_SAMPLE_FMT_S16; @@ -95,15 +95,13 @@ static AVStream *add_audio_stream(AVFormatContext *oc, enum AVCodecID codec_id) // some formats want stream headers to be separate if (oc->oformat->flags & AVFMT_GLOBALHEADER) c->flags |= CODEC_FLAG_GLOBAL_HEADER; - - return st; } -static void open_audio(AVFormatContext *oc, AVStream *st) +static void open_audio(AVFormatContext *oc, OutputStream *ost) { AVCodecContext *c; - c = st->codec; + c = ost->st->codec; /* open it */ if (avcodec_open2(c, NULL, NULL) < 0) { @@ -112,20 +110,20 @@ static void open_audio(AVFormatContext *oc, AVStream *st) } /* init signal generator */ - t = 0; - tincr = 2 * M_PI * 110.0 / c->sample_rate; + ost->t = 0; + ost->tincr = 2 * M_PI * 110.0 / c->sample_rate; /* increment frequency by 110 Hz per second */ - tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate; + ost->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; + ost->audio_input_frame_size = 10000; else - audio_input_frame_size = c->frame_size; + ost->audio_input_frame_size = c->frame_size; } /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and * 'nb_channels' channels. */ -static void get_audio_frame(AVFrame *frame, int nb_channels) +static void get_audio_frame(OutputStream *ost, AVFrame *frame, int nb_channels) { int j, i, v, ret; int16_t *q = (int16_t*)frame->data[0]; @@ -139,15 +137,15 @@ static void get_audio_frame(AVFrame *frame, int nb_channels) exit(1); for (j = 0; j < frame->nb_samples; j++) { - v = (int)(sin(t) * 10000); + v = (int)(sin(ost->t) * 10000); for (i = 0; i < nb_channels; i++) *q++ = v; - t += tincr; - tincr += tincr2; + ost->t += ost->tincr; + ost->tincr += ost->tincr2; } } -static void write_audio_frame(AVFormatContext *oc, AVStream *st) +static void write_audio_frame(AVFormatContext *oc, OutputStream *ost) { AVCodecContext *c; AVPacket pkt = { 0 }; // data and size must be 0; @@ -155,10 +153,10 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st) int got_packet, ret; av_init_packet(&pkt); - c = st->codec; + c = ost->st->codec; frame->sample_rate = c->sample_rate; - frame->nb_samples = audio_input_frame_size; + frame->nb_samples = ost->audio_input_frame_size; frame->format = AV_SAMPLE_FMT_S16; frame->channel_layout = c->channel_layout; ret = av_frame_get_buffer(frame, 0); @@ -167,13 +165,13 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st) exit(1); } - get_audio_frame(frame, c->channels); + get_audio_frame(ost, frame, c->channels); avcodec_encode_audio2(c, &pkt, frame, &got_packet); if (!got_packet) return; - pkt.stream_index = st->index; + pkt.stream_index = ost->st->index; /* Write the compressed frame to the media file. */ if (av_interleaved_write_frame(oc, &pkt) != 0) { @@ -183,9 +181,9 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st) av_frame_free(&frame); } -static void close_audio(AVFormatContext *oc, AVStream *st) +static void close_audio(AVFormatContext *oc, OutputStream *ost) { - avcodec_close(st->codec); + avcodec_close(ost->st->codec); } /**************************************************************/ @@ -413,13 +411,12 @@ static void close_video(AVFormatContext *oc, OutputStream *ost) int main(int argc, char **argv) { - OutputStream video_st; + OutputStream video_st, audio_st; const char *filename; AVOutputFormat *fmt; AVFormatContext *oc; - AVStream *audio_st; double audio_pts, video_pts; - int have_video = 0; + int have_video = 0, have_audio = 0; int i; /* Initialize libavcodec, and register all codecs and formats. */ @@ -458,21 +455,21 @@ int main(int argc, char **argv) /* Add the audio and video streams using the default format codecs * and initialize the codecs. */ - audio_st = NULL; if (fmt->video_codec != AV_CODEC_ID_NONE) { add_video_stream(&video_st, oc, fmt->video_codec); have_video = 1; } if (fmt->audio_codec != AV_CODEC_ID_NONE) { - audio_st = add_audio_stream(oc, fmt->audio_codec); + add_audio_stream(&audio_st, oc, fmt->audio_codec); + have_audio = 1; } /* Now that all the parameters are set, we can open the audio and * video codecs and allocate the necessary encode buffers. */ if (have_video) open_video(oc, &video_st); - if (audio_st) - open_audio(oc, audio_st); + if (have_audio) + open_audio(oc, &audio_st); av_dump_format(oc, 0, filename, 1); @@ -489,8 +486,8 @@ int main(int argc, char **argv) for (;;) { /* Compute current audio and video time. */ - if (audio_st) - audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den; + if (have_audio) + audio_pts = (double)audio_st.st->pts.val * audio_st.st->time_base.num / audio_st.st->time_base.den; else audio_pts = 0.0; @@ -500,13 +497,13 @@ int main(int argc, char **argv) else video_pts = 0.0; - if ((!audio_st || audio_pts >= STREAM_DURATION) && + if ((!have_audio || audio_pts >= STREAM_DURATION) && (!have_video || video_pts >= STREAM_DURATION)) break; /* write interleaved audio and video frames */ - if (!have_video || (have_video && audio_st && audio_pts < video_pts)) { - write_audio_frame(oc, audio_st); + if (!have_video || (have_video && have_audio && audio_pts < video_pts)) { + write_audio_frame(oc, &audio_st); } else { write_video_frame(oc, &video_st); } @@ -521,8 +518,8 @@ int main(int argc, char **argv) /* Close each codec. */ if (have_video) close_video(oc, &video_st); - if (audio_st) - close_audio(oc, audio_st); + if (have_audio) + close_audio(oc, &audio_st); /* Free the streams. */ for (i = 0; i < oc->nb_streams; i++) { |