diff options
Diffstat (limited to 'third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc')
-rw-r--r-- | third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc | 88 |
1 files changed, 88 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc new file mode 100644 index 0000000000..b497948491 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc @@ -0,0 +1,88 @@ +/* + * 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/ilbc/audio_encoder_ilbc.h" + +#include <memory> +#include <vector> + +#include "absl/strings/match.h" +#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h" +#include "rtc_base/numerics/safe_conversions.h" +#include "rtc_base/numerics/safe_minmax.h" +#include "rtc_base/string_to_number.h" + +namespace webrtc { +namespace { +int GetIlbcBitrate(int ptime) { + switch (ptime) { + case 20: + case 40: + // 38 bytes per frame of 20 ms => 15200 bits/s. + return 15200; + case 30: + case 60: + // 50 bytes per frame of 30 ms => (approx) 13333 bits/s. + return 13333; + default: + RTC_CHECK_NOTREACHED(); + } +} +} // namespace + +absl::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig( + const SdpAudioFormat& format) { + if (!absl::EqualsIgnoreCase(format.name.c_str(), "ILBC") || + format.clockrate_hz != 8000 || format.num_channels != 1) { + return absl::nullopt; + } + + AudioEncoderIlbcConfig config; + auto ptime_iter = format.parameters.find("ptime"); + if (ptime_iter != format.parameters.end()) { + auto ptime = rtc::StringToNumber<int>(ptime_iter->second); + if (ptime && *ptime > 0) { + const int whole_packets = *ptime / 10; + config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 20, 60); + } + } + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return absl::nullopt; + } + return config; +} + +void AudioEncoderIlbc::AppendSupportedEncoders( + std::vector<AudioCodecSpec>* specs) { + const SdpAudioFormat fmt = {"ILBC", 8000, 1}; + const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt)); + specs->push_back({fmt, info}); +} + +AudioCodecInfo AudioEncoderIlbc::QueryAudioEncoder( + const AudioEncoderIlbcConfig& config) { + RTC_DCHECK(config.IsOk()); + return {8000, 1, GetIlbcBitrate(config.frame_size_ms)}; +} + +std::unique_ptr<AudioEncoder> AudioEncoderIlbc::MakeAudioEncoder( + const AudioEncoderIlbcConfig& config, + int payload_type, + absl::optional<AudioCodecPairId> /*codec_pair_id*/, + const FieldTrialsView* field_trials) { + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return nullptr; + } + return std::make_unique<AudioEncoderIlbcImpl>(config, payload_type); +} + +} // namespace webrtc |