ao: proper end_time calulation & ao_read_frame() #14058
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so that people can freely pass eof = NULL to read_buffer() related functions.
The returned samples count can vary. Use a pre-calculated end_time_ns is inaccurate. This commit make ao_read_data() accept start_time_ns and calculate end_time_ns with: end_time_ns = start_time_ns + MP_TIME_S_TO_NS(pad_silence ? samples : pos) / ao->samplerate
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to allow pull-based AO to directly get audio frames. This also allows zero-copy AO.
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This PR fixes the calculation of
ao->buffer_state->end_time_nswhenpad_silence == falseand partial data is returned for pull-based AO. Previously,end_time_nswas calculated by driver delay plus the duration of requested sample count. It was not accurate ifpad_silence == falsegiving the number of returned samples can vary. To fix this issue, this PR introduces an API change thatao_read_data()andao_read_data_converted()now acceptsstart_time_ns, which is the the expected delay until the first sample reaches the speakers, instead ofout_time_ns.ao_read_data()then updatesao->buffer_state->end_time_nsusingstart_time_nsand the actual sample count got from the buffer.This PR also introduces a new API
ao_read_frame()for pull-based AO. Now pull-based AO can directly getmp_aframe. For AO that can handle data with variable size, this new API bypass the copying of audio data from audio frames into data buffer inread_buffer(). A zero-copyao_avfoundationthat utilizesao_read_frame()is included in this PR.This PR depends on #13967. Before it is merged, I think I'd better keep this PR a draft, though my work is done already.