diff --git a/doc/examples/muxing.c b/doc/examples/muxing.c index e26c25da16..5eb0135499 100644 --- a/doc/examples/muxing.c +++ b/doc/examples/muxing.c @@ -156,15 +156,8 @@ static AVStream *add_stream(OutputStream *ost, AVFormatContext *oc, static float t, tincr, tincr2; -AVFrame *audio_frame; -static uint8_t **src_samples_data; -static int src_samples_linesize; static int src_nb_samples; -static int max_dst_nb_samples; -uint8_t **dst_samples_data; -int dst_samples_linesize; -int dst_samples_size; int samples_count; struct SwrContext *swr_ctx = NULL; @@ -176,13 +169,6 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st) c = st->codec; - /* allocate and init a re-usable frame */ - audio_frame = av_frame_alloc(); - if (!audio_frame) { - fprintf(stderr, "Could not allocate audio frame\n"); - exit(1); - } - /* open it */ ret = avcodec_open2(c, codec, NULL); if (ret < 0) { @@ -196,20 +182,10 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st) /* increment frequency by 110 Hz per second */ tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate; - src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ? - 10000 : c->frame_size; - - ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels, - src_nb_samples, AV_SAMPLE_FMT_S16, 0); - if (ret < 0) { - fprintf(stderr, "Could not allocate source samples\n"); - exit(1); - } - - /* compute the number of converted samples: buffering is avoided - * ensuring that the output buffer will contain at least all the - * converted input samples */ - max_dst_nb_samples = src_nb_samples; + if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE) + src_nb_samples = 10000; + else + src_nb_samples = c->frame_size; /* create resampler context */ if (c->sample_fmt != AV_SAMPLE_FMT_S16) { @@ -232,29 +208,25 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st) fprintf(stderr, "Failed to initialize the resampling context\n"); exit(1); } - - ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels, - max_dst_nb_samples, c->sample_fmt, 0); - if (ret < 0) { - fprintf(stderr, "Could not allocate destination samples\n"); - exit(1); - } - } else { - dst_samples_data = src_samples_data; } - dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples, - c->sample_fmt, 0); } /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and * 'nb_channels' channels. */ -static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels) +static void get_audio_frame(AVFrame *frame, int nb_channels) { - int j, i, v; - int16_t *q; + int j, i, v, ret; + int16_t *q = (int16_t*)frame->data[0]; - q = samples; - for (j = 0; j < frame_size; j++) { + /* 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(frame); + if (ret < 0) + exit(1); + + for (j = 0; j < frame->nb_samples; j++) { v = (int)(sin(t) * 10000); for (i = 0; i < nb_channels; i++) *q++ = v; @@ -267,50 +239,63 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush) { AVCodecContext *c; AVPacket pkt = { 0 }; // data and size must be 0; - int got_packet, ret, dst_nb_samples; + AVFrame *frame = av_frame_alloc(); + int got_packet, ret; + int dst_nb_samples; av_init_packet(&pkt); c = st->codec; if (!flush) { - get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels); + frame->sample_rate = c->sample_rate; + frame->nb_samples = src_nb_samples; + frame->format = AV_SAMPLE_FMT_S16; + frame->channel_layout = c->channel_layout; + ret = av_frame_get_buffer(frame, 0); + if (ret < 0) { + fprintf(stderr, "Could not allocate an audio frame.\n"); + exit(1); + } + + get_audio_frame(frame, c->channels); /* convert samples from native format to destination codec format, using the resampler */ if (swr_ctx) { + AVFrame *tmp_frame = av_frame_alloc(); + /* compute destination number of samples */ dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples, c->sample_rate, c->sample_rate, AV_ROUND_UP); - if (dst_nb_samples > max_dst_nb_samples) { - av_free(dst_samples_data[0]); - ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels, - dst_nb_samples, c->sample_fmt, 0); - if (ret < 0) - exit(1); - max_dst_nb_samples = dst_nb_samples; - dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples, - c->sample_fmt, 0); + tmp_frame->sample_rate = c->sample_rate; + tmp_frame->nb_samples = dst_nb_samples; + tmp_frame->format = c->sample_fmt; + tmp_frame->channel_layout = c->channel_layout; + ret = av_frame_get_buffer(tmp_frame, 0); + if (ret < 0) { + fprintf(stderr, "Could not allocate an audio frame.\n"); + exit(1); } /* convert to destination format */ ret = swr_convert(swr_ctx, - dst_samples_data, dst_nb_samples, - (const uint8_t **)src_samples_data, src_nb_samples); + tmp_frame->data, dst_nb_samples, + (const uint8_t **)frame->data, src_nb_samples); if (ret < 0) { fprintf(stderr, "Error while converting\n"); exit(1); } + av_frame_free(&frame); + frame = tmp_frame; } else { dst_nb_samples = src_nb_samples; } - audio_frame->nb_samples = dst_nb_samples; - audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base); - avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt, - dst_samples_data[0], dst_samples_size, 0); + frame->nb_samples = dst_nb_samples; + frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base); samples_count += dst_nb_samples; } - ret = avcodec_encode_audio2(c, &pkt, flush ? NULL : audio_frame, &got_packet); + ret = avcodec_encode_audio2(c, &pkt, flush ? NULL : frame, &got_packet); if (ret < 0) { fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret)); exit(1); @@ -333,13 +318,6 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush) static void close_audio(AVFormatContext *oc, AVStream *st) { avcodec_close(st->codec); - if (dst_samples_data != src_samples_data) { - av_free(dst_samples_data[0]); - av_free(dst_samples_data); - } - av_free(src_samples_data[0]); - av_free(src_samples_data); - av_frame_free(&audio_frame); } /**************************************************************/