Specifically, don't expect the ACM to be able to create encoders; we have to give it an encoder that we make ourselves. The new way of creating encoders used a 32 kbit/s bitrate unconditionally for iSAC; I had to change it to 32 kbit/s for 16 kHz and 56 kbit/s for 32 kHz, which is what the old way of creating encoders has used since forever. I also had to change some test expectations on Opus, because the new way defaults to 32 kbit/s for mono and 64 kbit/s for stereo (which I believe to be correct), while the old way defaults to 64 kbit/s in both cases. Bug: webrtc:8396 Change-Id: I3aab944175a8e27f4c63380e822b27e839bba7f2 Reviewed-on: https://webrtc-review.googlesource.com/94540 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24375} |
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| .. | ||
| audio_decoder_isac_fix.cc | ||
| audio_decoder_isac_fix.h | ||
| audio_decoder_isac_float.cc | ||
| audio_decoder_isac_float.h | ||
| audio_decoder_isac.h | ||
| audio_encoder_isac_fix.cc | ||
| audio_encoder_isac_fix.h | ||
| audio_encoder_isac_float.cc | ||
| audio_encoder_isac_float.h | ||
| audio_encoder_isac.h | ||
| BUILD.gn | ||