G722 implementation of the AudioEncoderFactoryTemplate API

webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc

 /*
  *  Copyright (c) 2014 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/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
 
 #include <algorithm>
 
 #include <limits>
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/base/string_to_number.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 
 namespace webrtc {
 
 namespace {
 
 const size_t kSampleRateHz = 16000;
 
-AudioEncoderG722::Config CreateConfig(const CodecInst& codec_inst) {
-  AudioEncoderG722::Config config;
-  config.num_channels = codec_inst.channels;
+AudioEncoderG722Config CreateConfig(const CodecInst& codec_inst) {
+  AudioEncoderG722Config config;
+  config.num_channels = rtc::dchecked_cast<int>(codec_inst.channels);
   config.frame_size_ms = codec_inst.pacsize / 16;
-  config.payload_type = codec_inst.pltype;
   return config;
 }
 
-AudioEncoderG722::Config CreateConfig(int payload_type,
-                                      const SdpAudioFormat& format) {
-  AudioEncoderG722::Config config;
-  config.payload_type = payload_type;
-  config.num_channels = format.num_channels;
+}  // namespace
+
+rtc::Optional<AudioEncoderG722Config> AudioEncoderG722Impl::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (STR_CASE_CMP(format.name.c_str(), "g722") != 0 ||
+      format.clockrate_hz != 8000) {
+    return rtc::Optional<AudioEncoderG722Config>();
+  }
+
+  AudioEncoderG722Config config;
+  config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
   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 = std::max(10, std::min(whole_packets * 10, 60));
     }
   }
-  return config;
+  return config.IsOk() ? rtc::Optional<AudioEncoderG722Config>(config)
+                       : rtc::Optional<AudioEncoderG722Config>();
 }
 
-}  // namespace
-
-bool AudioEncoderG722::Config::IsOk() const {
-  return (frame_size_ms > 0) && (frame_size_ms % 10 == 0) &&
-      (num_channels >= 1);
-}
-
-AudioEncoderG722::AudioEncoderG722(const Config& config)
+AudioEncoderG722Impl::AudioEncoderG722Impl(const AudioEncoderG722Config& config,
+                                           int payload_type)
     : num_channels_(config.num_channels),
-      payload_type_(config.payload_type),
+      payload_type_(payload_type),
       num_10ms_frames_per_packet_(
           static_cast<size_t>(config.frame_size_ms / 10)),
       num_10ms_frames_buffered_(0),
       first_timestamp_in_buffer_(0),
       encoders_(new EncoderState[num_channels_]),
       interleave_buffer_(2 * num_channels_) {
   RTC_CHECK(config.IsOk());
   const size_t samples_per_channel =
       kSampleRateHz / 100 * num_10ms_frames_per_packet_;
   for (size_t i = 0; i < num_channels_; ++i) {
     encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
     encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2);
   }
   Reset();
 }
 
-AudioEncoderG722::AudioEncoderG722(const CodecInst& codec_inst)
-    : AudioEncoderG722(CreateConfig(codec_inst)) {}
+AudioEncoderG722Impl::AudioEncoderG722Impl(const CodecInst& codec_inst)
+    : AudioEncoderG722Impl(CreateConfig(codec_inst), codec_inst.pltype) {}
 
-AudioEncoderG722::AudioEncoderG722(int payload_type,
-                                   const SdpAudioFormat& format)
-    : AudioEncoderG722(CreateConfig(payload_type, format)) {}
+AudioEncoderG722Impl::AudioEncoderG722Impl(int payload_type,
+                                           const SdpAudioFormat& format)
+    : AudioEncoderG722Impl(*SdpToConfig(format), payload_type) {}
 
-AudioEncoderG722::~AudioEncoderG722() = default;
+AudioEncoderG722Impl::~AudioEncoderG722Impl() = default;
 
-rtc::Optional<AudioCodecInfo> AudioEncoderG722::QueryAudioEncoder(
+rtc::Optional<AudioCodecInfo> AudioEncoderG722Impl::QueryAudioEncoder(
     const SdpAudioFormat& format) {
   if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0) {
-    Config config = CreateConfig(0, format);
-    if (format.clockrate_hz == 8000 && config.IsOk()) {
+    const auto config_opt = SdpToConfig(format);
+    if (format.clockrate_hz == 8000 && config_opt) {
+      RTC_DCHECK(config_opt->IsOk());
       return rtc::Optional<AudioCodecInfo>(
-          {rtc::dchecked_cast<int>(kSampleRateHz), config.num_channels, 64000});
+          {rtc::dchecked_cast<int>(kSampleRateHz),
+           rtc::dchecked_cast<size_t>(config_opt->num_channels), 64000});
     }
   }
   return rtc::Optional<AudioCodecInfo>();
 }
 
-int AudioEncoderG722::SampleRateHz() const {
+int AudioEncoderG722Impl::SampleRateHz() const {
   return kSampleRateHz;
 }
 
-size_t AudioEncoderG722::NumChannels() const {
+size_t AudioEncoderG722Impl::NumChannels() const {
   return num_channels_;
 }
 
-int AudioEncoderG722::RtpTimestampRateHz() const {
+int AudioEncoderG722Impl::RtpTimestampRateHz() const {
   // The RTP timestamp rate for G.722 is 8000 Hz, even though it is a 16 kHz
   // codec.
   return kSampleRateHz / 2;
 }
 
-size_t AudioEncoderG722::Num10MsFramesInNextPacket() const {
+size_t AudioEncoderG722Impl::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
-size_t AudioEncoderG722::Max10MsFramesInAPacket() const {
+size_t AudioEncoderG722Impl::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
-int AudioEncoderG722::GetTargetBitrate() const {
+int AudioEncoderG722Impl::GetTargetBitrate() const {
   // 4 bits/sample, 16000 samples/s/channel.
   return static_cast<int>(64000 * NumChannels());
 }
 
-void AudioEncoderG722::Reset() {
+void AudioEncoderG722Impl::Reset() {
   num_10ms_frames_buffered_ = 0;
   for (size_t i = 0; i < num_channels_; ++i)
     RTC_CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder));
 }
 
-AudioEncoder::EncodedInfo AudioEncoderG722::EncodeImpl(
+AudioEncoder::EncodedInfo AudioEncoderG722Impl::EncodeImpl(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     rtc::Buffer* encoded) {
   if (num_10ms_frames_buffered_ == 0)
     first_timestamp_in_buffer_ = rtp_timestamp;
 
   // Deinterleave samples and save them in each channel's buffer.
   const size_t start = kSampleRateHz / 100 * num_10ms_frames_buffered_;
   for (size_t i = 0; i < kSampleRateHz / 100; ++i)
     for (size_t j = 0; j < num_channels_; ++j)
@@ skipping 35 matching lines @@
         }
 
         return bytes_to_encode;
       });
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
   info.encoder_type = CodecType::kG722;
   return info;
 }
 
-AudioEncoderG722::EncoderState::EncoderState() {
+AudioEncoderG722Impl::EncoderState::EncoderState() {
   RTC_CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder));
 }
 
-AudioEncoderG722::EncoderState::~EncoderState() {
+AudioEncoderG722Impl::EncoderState::~EncoderState() {
   RTC_CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
 }
 
-size_t AudioEncoderG722::SamplesPerChannel() const {
+size_t AudioEncoderG722Impl::SamplesPerChannel() const {
   return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
 }
 
 }  // namespace webrtc