| /* |
| * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "api/audio_codecs/isac/audio_encoder_isac_float.h" |
| |
| #include <memory> |
| |
| #include "absl/strings/match.h" |
| #include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h" |
| #include "rtc_base/string_to_number.h" |
| |
| namespace webrtc { |
| |
| absl::optional<AudioEncoderIsacFloat::Config> |
| AudioEncoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) { |
| if (absl::EqualsIgnoreCase(format.name, "ISAC") && |
| (format.clockrate_hz == 16000 || format.clockrate_hz == 32000) && |
| format.num_channels == 1) { |
| Config config; |
| config.sample_rate_hz = format.clockrate_hz; |
| config.bit_rate = format.clockrate_hz == 16000 ? 32000 : 56000; |
| if (config.sample_rate_hz == 16000) { |
| // For sample rate 16 kHz, optionally use 60 ms frames, instead of the |
| // default 30 ms. |
| const auto ptime_iter = format.parameters.find("ptime"); |
| if (ptime_iter != format.parameters.end()) { |
| const auto ptime = rtc::StringToNumber<int>(ptime_iter->second); |
| if (ptime && *ptime >= 60) { |
| config.frame_size_ms = 60; |
| } |
| } |
| } |
| if (!config.IsOk()) { |
| RTC_DCHECK_NOTREACHED(); |
| return absl::nullopt; |
| } |
| return config; |
| } else { |
| return absl::nullopt; |
| } |
| } |
| |
| void AudioEncoderIsacFloat::AppendSupportedEncoders( |
| std::vector<AudioCodecSpec>* specs) { |
| for (int sample_rate_hz : {16000, 32000}) { |
| const SdpAudioFormat fmt = {"ISAC", sample_rate_hz, 1}; |
| const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt)); |
| specs->push_back({fmt, info}); |
| } |
| } |
| |
| AudioCodecInfo AudioEncoderIsacFloat::QueryAudioEncoder( |
| const AudioEncoderIsacFloat::Config& config) { |
| RTC_DCHECK(config.IsOk()); |
| constexpr int min_bitrate = 10000; |
| const int max_bitrate = config.sample_rate_hz == 16000 ? 32000 : 56000; |
| const int default_bitrate = max_bitrate; |
| return {config.sample_rate_hz, 1, default_bitrate, min_bitrate, max_bitrate}; |
| } |
| |
| std::unique_ptr<AudioEncoder> AudioEncoderIsacFloat::MakeAudioEncoder( |
| const AudioEncoderIsacFloat::Config& config, |
| int payload_type, |
| absl::optional<AudioCodecPairId> /*codec_pair_id*/, |
| const FieldTrialsView* field_trials) { |
| AudioEncoderIsacFloatImpl::Config c; |
| c.payload_type = payload_type; |
| c.sample_rate_hz = config.sample_rate_hz; |
| c.frame_size_ms = config.frame_size_ms; |
| c.bit_rate = config.bit_rate; |
| if (!config.IsOk()) { |
| RTC_DCHECK_NOTREACHED(); |
| return nullptr; |
| } |
| return std::make_unique<AudioEncoderIsacFloatImpl>(c); |
| } |
| |
| } // namespace webrtc |