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@@ -34,6 +34,7 @@ |
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#include <string.h> |
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#include <math.h> |
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#include <libavutil/avassert.h> |
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#include <libavutil/channel_layout.h> |
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#include <libavutil/opt.h> |
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#include <libavutil/mathematics.h> |
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@@ -58,7 +59,6 @@ typedef struct OutputStream { |
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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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static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt) |
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@@ -179,10 +179,27 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, 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->audio_input_frame_size = 10000; |
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ost->frame->nb_samples = 10000; |
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else |
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ost->audio_input_frame_size = c->frame_size; |
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ost->frame->nb_samples = c->frame_size; |
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ost->tmp_frame = av_frame_alloc(); |
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if (!ost->frame) |
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exit(1); |
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ost->tmp_frame->sample_rate = c->sample_rate; |
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ost->tmp_frame->format = c->sample_fmt; |
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ost->tmp_frame->channel_layout = c->channel_layout; |
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ost->tmp_frame->nb_samples = ost->frame->nb_samples; |
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/* create resampler context */ |
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if (c->sample_fmt != AV_SAMPLE_FMT_S16) { |
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@@ -206,6 +223,17 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost) |
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exit(1); |
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} |
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} |
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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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ret = av_frame_get_buffer(ost->tmp_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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} |
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/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and |
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@@ -236,58 +264,37 @@ static void write_audio_frame(AVFormatContext *oc, OutputStream *ost, int flush) |
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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 = av_frame_alloc(); |
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int got_packet, ret; |
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int dst_nb_samples; |
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AVFrame *frame; |
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av_init_packet(&pkt); |
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c = ost->st->codec; |
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if (!flush) { |
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frame->sample_rate = c->sample_rate; |
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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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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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get_audio_frame(ost, frame, c->channels); |
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get_audio_frame(ost, ost->frame, c->channels); |
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/* convert samples from native format to destination codec format, using the resampler */ |
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if (swr_ctx) { |
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AVFrame *tmp_frame = av_frame_alloc(); |
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/* compute destination number of samples */ |
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dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + ost->audio_input_frame_size, |
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dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + ost->frame->nb_samples, |
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c->sample_rate, c->sample_rate, AV_ROUND_UP); |
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tmp_frame->sample_rate = c->sample_rate; |
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tmp_frame->nb_samples = dst_nb_samples; |
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tmp_frame->format = c->sample_fmt; |
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tmp_frame->channel_layout = c->channel_layout; |
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ret = av_frame_get_buffer(tmp_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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av_assert0(dst_nb_samples == ost->frame->nb_samples); |
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/* convert to destination format */ |
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ret = swr_convert(swr_ctx, |
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tmp_frame->data, dst_nb_samples, |
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(const uint8_t **)frame->data, ost->audio_input_frame_size); |
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ost->tmp_frame->data, dst_nb_samples, |
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(const uint8_t **)ost->frame->data, ost->frame->nb_samples); |
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if (ret < 0) { |
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fprintf(stderr, "Error while converting\n"); |
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exit(1); |
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} |
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av_frame_free(&frame); |
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frame = tmp_frame; |
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frame = ost->tmp_frame; |
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} else { |
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dst_nb_samples = ost->audio_input_frame_size; |
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dst_nb_samples = ost->frame->nb_samples; |
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frame = ost->frame; |
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} |
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frame->nb_samples = dst_nb_samples; |
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frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base); |
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samples_count += dst_nb_samples; |
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} |
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@@ -301,6 +308,7 @@ static void write_audio_frame(AVFormatContext *oc, OutputStream *ost, int flush) |
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if (!got_packet) { |
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if (flush) |
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audio_is_eof = 1; |
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return; |
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} |
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@@ -312,11 +320,6 @@ static void write_audio_frame(AVFormatContext *oc, OutputStream *ost, int flush) |
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} |
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} |
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static void close_audio(AVFormatContext *oc, OutputStream *ost) |
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{ |
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avcodec_close(ost->st->codec); |
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} |
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/**************************************************************/ |
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/* video output */ |
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@@ -477,7 +480,7 @@ static void write_video_frame(AVFormatContext *oc, OutputStream *ost, int flush) |
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frame_count++; |
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} |
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static void close_video(AVFormatContext *oc, OutputStream *ost) |
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static void close_stream(AVFormatContext *oc, OutputStream *ost) |
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{ |
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avcodec_close(ost->st->codec); |
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av_frame_free(&ost->frame); |
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@@ -489,7 +492,7 @@ 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, audio_st; |
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OutputStream video_st = { 0 }, audio_st = { 0 }; |
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const char *filename; |
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AVOutputFormat *fmt; |
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AVFormatContext *oc; |
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@@ -592,9 +595,9 @@ 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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close_stream(oc, &video_st); |
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if (have_audio) |
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close_audio(oc, &audio_st); |
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close_stream(oc, &audio_st); |
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if (!(fmt->flags & AVFMT_NOFILE)) |
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/* Close the output file. */ |
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