tor-browser

The Tor Browser
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audio_encoder_g722.cc (2691B)

      1 /*
      2 *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
      3 *
      4 *  Use of this source code is governed by a BSD-style license
      5 *  that can be found in the LICENSE file in the root of the source
      6 *  tree. An additional intellectual property rights grant can be found
      7 *  in the file PATENTS.  All contributing project authors may
      8 *  be found in the AUTHORS file in the root of the source tree.
      9 */
     10 
     11 #include "api/audio_codecs/g722/audio_encoder_g722.h"
     12 
     13 #include <cstddef>
     14 #include <map>
     15 #include <memory>
     16 #include <optional>
     17 #include <vector>
     18 
     19 #include "absl/strings/match.h"
     20 #include "api/audio_codecs/audio_codec_pair_id.h"
     21 #include "api/audio_codecs/audio_encoder.h"
     22 #include "api/audio_codecs/audio_format.h"
     23 #include "api/audio_codecs/g722/audio_encoder_g722_config.h"
     24 #include "api/field_trials_view.h"
     25 #include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
     26 #include "rtc_base/checks.h"
     27 #include "rtc_base/numerics/safe_conversions.h"
     28 #include "rtc_base/numerics/safe_minmax.h"
     29 #include "rtc_base/string_to_number.h"
     30 
     31 namespace webrtc {
     32 
     33 std::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
     34    const SdpAudioFormat& format) {
     35  if (!absl::EqualsIgnoreCase(format.name, "g722") ||
     36      format.clockrate_hz != 8000) {
     37    return std::nullopt;
     38  }
     39 
     40  AudioEncoderG722Config config;
     41  config.num_channels = checked_cast<int>(format.num_channels);
     42  auto ptime_iter = format.parameters.find("ptime");
     43  if (ptime_iter != format.parameters.end()) {
     44    auto ptime = StringToNumber<int>(ptime_iter->second);
     45    if (ptime && *ptime > 0) {
     46      const int whole_packets = *ptime / 10;
     47      config.frame_size_ms = SafeClamp<int>(whole_packets * 10, 10, 60);
     48    }
     49  }
     50  if (!config.IsOk()) {
     51    RTC_DCHECK_NOTREACHED();
     52    return std::nullopt;
     53  }
     54  return config;
     55 }
     56 
     57 void AudioEncoderG722::AppendSupportedEncoders(
     58    std::vector<AudioCodecSpec>* specs) {
     59  const SdpAudioFormat fmt = {"G722", 8000, 1};
     60  const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
     61  specs->push_back({fmt, info});
     62 }
     63 
     64 AudioCodecInfo AudioEncoderG722::QueryAudioEncoder(
     65    const AudioEncoderG722Config& config) {
     66  RTC_DCHECK(config.IsOk());
     67  return {16000, dchecked_cast<size_t>(config.num_channels),
     68          64000 * config.num_channels};
     69 }
     70 
     71 std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
     72    const AudioEncoderG722Config& config,
     73    int payload_type,
     74    std::optional<AudioCodecPairId> /*codec_pair_id*/,
     75    const FieldTrialsView* /* field_trials */) {
     76  if (!config.IsOk()) {
     77    RTC_DCHECK_NOTREACHED();
     78    return nullptr;
     79  }
     80  return std::make_unique<AudioEncoderG722Impl>(config, payload_type);
     81 }
     82 
     83 }  // namespace webrtc