fbeb76ab51
This reverts commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1. Reason for revert: breaks a downstream project Original change's description: > [ACM] iSAC audio codec removed > > Note: this CL has to leave behind one part of iSAC, which is its VAD > currently used by AGC1 in APM. The target visibility has been > restricted and the VAD will be removed together with AGC1 when the > time comes. > > Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319 > > Bug: webrtc:14450 > Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421 > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38652} Bug: webrtc:14450 Change-Id: Ice138004e84e8c5f896684e8d01133d4b2a77bb7 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283800 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38655}
55 lines
1.8 KiB
C++
55 lines
1.8 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_FIX_H_
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#define API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_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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#include "api/field_trials_view.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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// iSAC encoder API (fixed-point implementation) for use as a template
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// parameter to CreateAudioEncoderFactory<...>().
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struct RTC_EXPORT AudioEncoderIsacFix {
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struct Config {
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bool IsOk() const {
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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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}
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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(Config config);
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static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
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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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const FieldTrialsView* field_trials = nullptr);
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};
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} // namespace webrtc
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#endif // API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_H_
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