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}
73 lines
2.2 KiB
C++
73 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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#include "api/audio_codecs/isac/audio_encoder_isac_fix.h"
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#include <memory>
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#include "absl/strings/match.h"
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#include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
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#include "rtc_base/string_to_number.h"
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namespace webrtc {
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absl::optional<AudioEncoderIsacFix::Config> AudioEncoderIsacFix::SdpToConfig(
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const SdpAudioFormat& format) {
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if (absl::EqualsIgnoreCase(format.name, "ISAC") &&
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format.clockrate_hz == 16000 && format.num_channels == 1) {
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Config config;
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const auto ptime_iter = format.parameters.find("ptime");
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if (ptime_iter != format.parameters.end()) {
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const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
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if (ptime && *ptime >= 60) {
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config.frame_size_ms = 60;
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}
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}
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if (!config.IsOk()) {
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RTC_DCHECK_NOTREACHED();
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return absl::nullopt;
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}
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return config;
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} else {
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return absl::nullopt;
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}
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}
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void AudioEncoderIsacFix::AppendSupportedEncoders(
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std::vector<AudioCodecSpec>* specs) {
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const SdpAudioFormat fmt = {"ISAC", 16000, 1};
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const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
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specs->push_back({fmt, info});
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}
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AudioCodecInfo AudioEncoderIsacFix::QueryAudioEncoder(
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AudioEncoderIsacFix::Config config) {
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RTC_DCHECK(config.IsOk());
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return {16000, 1, 32000, 10000, 32000};
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}
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std::unique_ptr<AudioEncoder> AudioEncoderIsacFix::MakeAudioEncoder(
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AudioEncoderIsacFix::Config config,
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int payload_type,
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absl::optional<AudioCodecPairId> /*codec_pair_id*/,
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const FieldTrialsView* field_trials) {
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AudioEncoderIsacFixImpl::Config c;
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c.frame_size_ms = config.frame_size_ms;
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c.bit_rate = config.bit_rate;
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c.payload_type = payload_type;
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if (!config.IsOk()) {
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RTC_DCHECK_NOTREACHED();
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return nullptr;
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}
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return std::make_unique<AudioEncoderIsacFixImpl>(c);
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}
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} // namespace webrtc
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