lavr: change the type of the data buffers to uint8_t**.
This is more consistent with what the rest of Libav does. This breaks API.
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parent
30223b3bf2
commit
e7ba5b1de0
4
avplay.c
4
avplay.c
@ -1961,9 +1961,9 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
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is->audio_buf1 = tmp_out;
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out_samples = avresample_convert(is->avr,
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(void **)&is->audio_buf1,
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&is->audio_buf1,
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out_linesize, nb_samples,
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(void **)is->frame->data,
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is->frame->data,
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is->frame->linesize[0],
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is->frame->nb_samples);
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if (out_samples < 0) {
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@ -133,7 +133,7 @@ static int request_frame(AVFilterLink *link)
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nb_samples);
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if (!buf)
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return AVERROR(ENOMEM);
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ret = avresample_convert(s->avr, (void**)buf->extended_data,
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ret = avresample_convert(s->avr, buf->extended_data,
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buf->linesize[0], nb_samples, NULL, 0, 0);
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if (ret <= 0) {
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avfilter_unref_bufferp(&buf);
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@ -149,7 +149,7 @@ static int request_frame(AVFilterLink *link)
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static int write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
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{
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int ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
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int ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,
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buf->linesize[0], buf->audio->nb_samples);
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avfilter_unref_buffer(buf);
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return ret;
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@ -210,7 +210,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
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goto fail;
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}
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avresample_read(s->avr, (void**)buf_out->extended_data, out_size);
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avresample_read(s->avr, buf_out->extended_data, out_size);
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buf_out->pts = s->pts;
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if (delta > 0) {
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@ -230,7 +230,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
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avresample_read(s->avr, NULL, avresample_available(s->avr));
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s->pts = pts - avresample_get_delay(s->avr);
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ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
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ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,
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buf->linesize[0], buf->audio->nb_samples);
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fail:
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@ -149,7 +149,7 @@ static int request_frame(AVFilterLink *outlink)
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if (!buf)
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return AVERROR(ENOMEM);
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ret = avresample_convert(s->avr, (void**)buf->extended_data,
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ret = avresample_convert(s->avr, buf->extended_data,
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buf->linesize[0], nb_samples,
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NULL, 0, 0);
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if (ret <= 0) {
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@ -186,9 +186,9 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
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goto fail;
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}
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ret = avresample_convert(s->avr, (void**)buf_out->extended_data,
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ret = avresample_convert(s->avr, buf_out->extended_data,
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buf_out->linesize[0], nb_samples,
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(void**)buf->extended_data, buf->linesize[0],
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buf->extended_data, buf->linesize[0],
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buf->audio->nb_samples);
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if (ret < 0) {
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avfilter_unref_buffer(buf_out);
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@ -62,7 +62,7 @@ int ff_audio_data_set_channels(AudioData *a, int channels)
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return 0;
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}
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int ff_audio_data_init(AudioData *a, void **src, int plane_size, int channels,
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int ff_audio_data_init(AudioData *a, uint8_t **src, int plane_size, int channels,
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int nb_samples, enum AVSampleFormat sample_fmt,
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int read_only, const char *name)
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{
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@ -73,7 +73,7 @@ int ff_audio_data_set_channels(AudioData *a, int channels);
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* @param name name for debug logging (can be NULL)
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* @return 0 on success, negative AVERROR value on error
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*/
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int ff_audio_data_init(AudioData *a, void **src, int plane_size, int channels,
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int ff_audio_data_init(AudioData *a, uint8_t **src, int plane_size, int channels,
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int nb_samples, enum AVSampleFormat sample_fmt,
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int read_only, const char *name);
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@ -305,8 +305,8 @@ int main(int argc, char **argv)
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goto end;
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}
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ret = avresample_convert(s, (void **)out_data, out_linesize, out_rate * 6,
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(void **) in_data, in_linesize, in_rate * 6);
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ret = avresample_convert(s, out_data, out_linesize, out_rate * 6,
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in_data, in_linesize, in_rate * 6);
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if (ret < 0) {
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char errbuf[256];
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av_strerror(ret, errbuf, sizeof(errbuf));
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@ -234,8 +234,8 @@ int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
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* not including converted samples added to the internal
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* output FIFO
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*/
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int avresample_convert(AVAudioResampleContext *avr, void **output,
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int out_plane_size, int out_samples, void **input,
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int avresample_convert(AVAudioResampleContext *avr, uint8_t **output,
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int out_plane_size, int out_samples, uint8_t **input,
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int in_plane_size, int in_samples);
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/**
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@ -287,6 +287,6 @@ int avresample_available(AVAudioResampleContext *avr);
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* @param nb_samples number of samples to read from the FIFO
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* @return the number of samples written to output
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*/
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int avresample_read(AVAudioResampleContext *avr, void **output, int nb_samples);
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int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples);
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#endif /* AVRESAMPLE_AVRESAMPLE_H */
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@ -247,8 +247,8 @@ static int handle_buffered_output(AVAudioResampleContext *avr,
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}
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int attribute_align_arg avresample_convert(AVAudioResampleContext *avr,
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void **output, int out_plane_size,
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int out_samples, void **input,
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uint8_t **output, int out_plane_size,
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int out_samples, uint8_t **input,
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int in_plane_size, int in_samples)
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{
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AudioData input_buffer;
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@ -410,11 +410,11 @@ int avresample_available(AVAudioResampleContext *avr)
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return av_audio_fifo_size(avr->out_fifo);
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}
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int avresample_read(AVAudioResampleContext *avr, void **output, int nb_samples)
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int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples)
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{
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if (!output)
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return av_audio_fifo_drain(avr->out_fifo, nb_samples);
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return av_audio_fifo_read(avr->out_fifo, output, nb_samples);
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return av_audio_fifo_read(avr->out_fifo, (void**)output, nb_samples);
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}
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unsigned avresample_version(void)
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