mirror of https://git.ffmpeg.org/ffmpeg.git
Merge commit '3a70c0c95feacb3844d05eebd579fc8189a77eee'
* commit '3a70c0c95feacb3844d05eebd579fc8189a77eee': examples/transcode_aac: generate proper PTS and set the muxer timebase Merged-by: Michael Niedermayer <michaelni@gmx.at>
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commit
e5c28d4f9a
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@ -183,6 +183,10 @@ static int open_output_file(const char *filename,
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/** Allow the use of the experimental AAC encoder */
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(*output_codec_context)->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
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/** Set the sample rate for the container. */
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stream->time_base.den = input_codec_context->sample_rate;
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stream->time_base.num = 1;
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/**
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* Some container formats (like MP4) require global headers to be present
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* Mark the encoder so that it behaves accordingly.
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@ -539,6 +543,9 @@ static int init_output_frame(AVFrame **frame,
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return 0;
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}
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/** Global timestamp for the audio frames */
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static int64_t pts = 0;
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/** Encode one frame worth of audio to the output file. */
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static int encode_audio_frame(AVFrame *frame,
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AVFormatContext *output_format_context,
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@ -550,6 +557,12 @@ static int encode_audio_frame(AVFrame *frame,
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int error;
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init_packet(&output_packet);
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/** Set a timestamp based on the sample rate for the container. */
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if (frame) {
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frame->pts = pts;
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pts += frame->nb_samples;
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}
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/**
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* Encode the audio frame and store it in the temporary packet.
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* The output audio stream encoder is used to do this.
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