mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-12 13:20:44 +01:00
bb547203bf
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
73 lines
2.6 KiB
C++
73 lines
2.6 KiB
C++
/*
|
|
* 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 "webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h"
|
|
|
|
#include "webrtc/common_types.h"
|
|
#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
|
|
#include "webrtc/rtc_base/ptr_util.h"
|
|
#include "webrtc/rtc_base/string_to_number.h"
|
|
|
|
namespace webrtc {
|
|
|
|
rtc::Optional<AudioEncoderIsacFloat::Config> AudioEncoderIsacFloat::SdpToConfig(
|
|
const SdpAudioFormat& format) {
|
|
if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
|
|
(format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
|
|
format.num_channels == 1) {
|
|
Config config;
|
|
config.sample_rate_hz = format.clockrate_hz;
|
|
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;
|
|
}
|
|
}
|
|
}
|
|
return rtc::Optional<Config>(config);
|
|
} else {
|
|
return rtc::Optional<Config>();
|
|
}
|
|
}
|
|
|
|
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) {
|
|
RTC_DCHECK(config.IsOk());
|
|
AudioEncoderIsacFloatImpl::Config c;
|
|
c.sample_rate_hz = config.sample_rate_hz;
|
|
c.frame_size_ms = config.frame_size_ms;
|
|
c.payload_type = payload_type;
|
|
return rtc::MakeUnique<AudioEncoderIsacFloatImpl>(c);
|
|
}
|
|
|
|
} // namespace webrtc
|