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801500cf99
Specifically, don't expect the ACM to be able to create encoders; we have to give it an encoder that we make ourselves. The new way of creating encoders used a 32 kbit/s bitrate unconditionally for iSAC; I had to change it to 32 kbit/s for 16 kHz and 56 kbit/s for 32 kHz, which is what the old way of creating encoders has used since forever. I also had to change some test expectations on Opus, because the new way defaults to 32 kbit/s for mono and 64 kbit/s for stereo (which I believe to be correct), while the old way defaults to 64 kbit/s in both cases. Bug: webrtc:8396 Change-Id: I3aab944175a8e27f4c63380e822b27e839bba7f2 Reviewed-on: https://webrtc-review.googlesource.com/94540 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24375}
76 lines
2.7 KiB
C++
76 lines
2.7 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 "absl/memory/memory.h"
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#include "common_types.h" // NOLINT(build/include)
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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 (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
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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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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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RTC_DCHECK(config.IsOk());
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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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return absl::make_unique<AudioEncoderIsacFloatImpl>(c);
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}
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} // namespace webrtc
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