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801500cf99
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}
67 lines
2.2 KiB
C++
67 lines
2.2 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_
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#define API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_
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#include <memory>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/audio_codecs/audio_codec_pair_id.h"
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#include "api/audio_codecs/audio_encoder.h"
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#include "api/audio_codecs/audio_format.h"
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namespace webrtc {
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// iSAC encoder API (floating-point implementation) for use as a template
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// parameter to CreateAudioEncoderFactory<...>().
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//
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// NOTE: This struct is still under development and may change without notice.
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struct AudioEncoderIsacFloat {
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struct Config {
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bool IsOk() const {
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switch (sample_rate_hz) {
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case 16000:
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if (frame_size_ms != 30 && frame_size_ms != 60) {
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return false;
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}
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if (bit_rate < 10000 || bit_rate > 32000) {
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return false;
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}
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return true;
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case 32000:
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if (frame_size_ms != 30) {
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return false;
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}
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if (bit_rate < 10000 || bit_rate > 56000) {
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return false;
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}
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return true;
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default:
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return false;
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}
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}
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int sample_rate_hz = 16000;
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int frame_size_ms = 30;
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int bit_rate = 32000; // Limit on short-term average bit rate, in bits/s.
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};
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static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
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static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
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static AudioCodecInfo QueryAudioEncoder(const Config& config);
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static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
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const Config& config,
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int payload_type,
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absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
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};
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} // namespace webrtc
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#endif // API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_
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