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deb1b1bc70
We should avoid creating codecs with invalid parameters, since this can expose security issues. For many codecs the IsOk() method to check the codec config is only called in DCHECKs. This CL ensures IsOk() is always called, also in non-debug builds. Bug: chromium:1265806 Change-Id: Ibd3c6c65d3bb547cd2603e11808ac40ac693a8b1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238801 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35422}
84 lines
2.8 KiB
C++
84 lines
2.8 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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#include "api/audio_codecs/isac/audio_encoder_isac_float.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/main/include/audio_encoder_isac.h"
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#include "rtc_base/string_to_number.h"
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namespace webrtc {
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absl::optional<AudioEncoderIsacFloat::Config>
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AudioEncoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) {
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if (absl::EqualsIgnoreCase(format.name, "ISAC") &&
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(format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
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format.num_channels == 1) {
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Config config;
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config.sample_rate_hz = format.clockrate_hz;
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config.bit_rate = format.clockrate_hz == 16000 ? 32000 : 56000;
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if (config.sample_rate_hz == 16000) {
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// For sample rate 16 kHz, optionally use 60 ms frames, instead of the
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// default 30 ms.
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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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}
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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 AudioEncoderIsacFloat::AppendSupportedEncoders(
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std::vector<AudioCodecSpec>* specs) {
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for (int sample_rate_hz : {16000, 32000}) {
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const SdpAudioFormat fmt = {"ISAC", sample_rate_hz, 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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}
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AudioCodecInfo AudioEncoderIsacFloat::QueryAudioEncoder(
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const AudioEncoderIsacFloat::Config& config) {
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RTC_DCHECK(config.IsOk());
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constexpr int min_bitrate = 10000;
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const int max_bitrate = config.sample_rate_hz == 16000 ? 32000 : 56000;
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const int default_bitrate = max_bitrate;
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return {config.sample_rate_hz, 1, default_bitrate, min_bitrate, max_bitrate};
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}
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std::unique_ptr<AudioEncoder> AudioEncoderIsacFloat::MakeAudioEncoder(
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const AudioEncoderIsacFloat::Config& config,
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int payload_type,
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absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
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AudioEncoderIsacFloatImpl::Config c;
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c.payload_type = payload_type;
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c.sample_rate_hz = config.sample_rate_hz;
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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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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<AudioEncoderIsacFloatImpl>(c);
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}
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} // namespace webrtc
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