Remove dead code
This code became dead when the builtin audio codec factories were
rewritten in https://codereview.webrtc.org/2997713002/.
BUG=webrtc:7821, webrtc:7822
Review-Url: https://codereview.webrtc.org/3003603002
Cr-Original-Commit-Position: refs/heads/master@{#19535}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: d1d79f6866fcde94bb0354cec7d6ecaaf72de235
diff --git a/api/audio_codecs/g722/BUILD.gn b/api/audio_codecs/g722/BUILD.gn
index 2c1349a..f3108e7 100644
--- a/api/audio_codecs/g722/BUILD.gn
+++ b/api/audio_codecs/g722/BUILD.gn
@@ -26,6 +26,7 @@
deps = [
":audio_encoder_g722_config",
"..:audio_codecs_api",
+ "../../..:webrtc_common",
"../../../modules/audio_coding:g722",
"../../../rtc_base:rtc_base_approved",
]
diff --git a/api/audio_codecs/g722/audio_encoder_g722.cc b/api/audio_codecs/g722/audio_encoder_g722.cc
index 09b3faf..b9df585 100644
--- a/api/audio_codecs/g722/audio_encoder_g722.cc
+++ b/api/audio_codecs/g722/audio_encoder_g722.cc
@@ -13,15 +13,34 @@
#include <memory>
#include <vector>
+#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h"
+#include "webrtc/rtc_base/safe_minmax.h"
+#include "webrtc/rtc_base/string_to_number.h"
namespace webrtc {
rtc::Optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
const SdpAudioFormat& format) {
- return AudioEncoderG722Impl::SdpToConfig(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::checked_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 = rtc::SafeClamp<int>(whole_packets * 10, 10, 60);
+ }
+ }
+ return config.IsOk() ? rtc::Optional<AudioEncoderG722Config>(config)
+ : rtc::Optional<AudioEncoderG722Config>();
}
void AudioEncoderG722::AppendSupportedEncoders(
diff --git a/api/audio_codecs/ilbc/BUILD.gn b/api/audio_codecs/ilbc/BUILD.gn
index 6ef8856..ab9681a 100644
--- a/api/audio_codecs/ilbc/BUILD.gn
+++ b/api/audio_codecs/ilbc/BUILD.gn
@@ -26,6 +26,7 @@
deps = [
":audio_encoder_ilbc_config",
"..:audio_codecs_api",
+ "../../..:webrtc_common",
"../../../modules/audio_coding:ilbc",
"../../../rtc_base:rtc_base_approved",
]
diff --git a/api/audio_codecs/ilbc/audio_encoder_ilbc.cc b/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
index 13a1c2e..fd11f00 100644
--- a/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
+++ b/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
@@ -13,9 +13,12 @@
#include <memory>
#include <vector>
+#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h"
+#include "webrtc/rtc_base/safe_minmax.h"
+#include "webrtc/rtc_base/string_to_number.h"
namespace webrtc {
namespace {
@@ -37,7 +40,22 @@
rtc::Optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
const SdpAudioFormat& format) {
- return AudioEncoderIlbcImpl::SdpToConfig(format);
+ if (STR_CASE_CMP(format.name.c_str(), "ILBC") != 0 ||
+ format.clockrate_hz != 8000 || format.num_channels != 1) {
+ return rtc::Optional<AudioEncoderIlbcConfig>();
+ }
+
+ AudioEncoderIlbcConfig config;
+ 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 = rtc::SafeClamp<int>(whole_packets * 10, 20, 60);
+ }
+ }
+ return config.IsOk() ? rtc::Optional<AudioEncoderIlbcConfig>(config)
+ : rtc::Optional<AudioEncoderIlbcConfig>();
}
void AudioEncoderIlbc::AppendSupportedEncoders(
diff --git a/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
index 8bff2ec..711eed7 100644
--- a/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
+++ b/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -16,7 +16,6 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/string_to_number.h"
namespace webrtc {
@@ -31,35 +30,6 @@
return config;
}
-template <typename T>
-typename T::Config CreateConfig(int payload_type,
- const SdpAudioFormat& format) {
- typename T::Config config;
- config.frame_size_ms = 20;
- 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));
- }
- }
- config.num_channels = format.num_channels;
- config.payload_type = payload_type;
- return config;
-}
-
-template <typename T>
-rtc::Optional<AudioCodecInfo> QueryAudioEncoderImpl(
- const SdpAudioFormat& format) {
- if (STR_CASE_CMP(format.name.c_str(), T::GetPayloadName()) == 0 &&
- format.clockrate_hz == 8000 && format.num_channels >= 1 &&
- CreateConfig<T>(0, format).IsOk()) {
- return rtc::Optional<AudioCodecInfo>({8000, format.num_channels, 64000});
- }
- return rtc::Optional<AudioCodecInfo>();
-}
-
} // namespace
bool AudioEncoderPcm::Config::IsOk() const {
@@ -138,15 +108,6 @@
AudioEncoderPcmA::AudioEncoderPcmA(const CodecInst& codec_inst)
: AudioEncoderPcmA(CreateConfig<AudioEncoderPcmA>(codec_inst)) {}
-AudioEncoderPcmA::AudioEncoderPcmA(int payload_type,
- const SdpAudioFormat& format)
- : AudioEncoderPcmA(CreateConfig<AudioEncoderPcmA>(payload_type, format)) {}
-
-rtc::Optional<AudioCodecInfo> AudioEncoderPcmA::QueryAudioEncoder(
- const SdpAudioFormat& format) {
- return QueryAudioEncoderImpl<AudioEncoderPcmA>(format);
-}
-
size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) {
@@ -164,15 +125,6 @@
AudioEncoderPcmU::AudioEncoderPcmU(const CodecInst& codec_inst)
: AudioEncoderPcmU(CreateConfig<AudioEncoderPcmU>(codec_inst)) {}
-AudioEncoderPcmU::AudioEncoderPcmU(int payload_type,
- const SdpAudioFormat& format)
- : AudioEncoderPcmU(CreateConfig<AudioEncoderPcmU>(payload_type, format)) {}
-
-rtc::Optional<AudioCodecInfo> AudioEncoderPcmU::QueryAudioEncoder(
- const SdpAudioFormat& format) {
- return QueryAudioEncoderImpl<AudioEncoderPcmU>(format);
-}
-
size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) {
diff --git a/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
index be8164a..22a15a1 100644
--- a/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
+++ b/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
@@ -14,7 +14,6 @@
#include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/rtc_base/constructormagic.h"
namespace webrtc {
@@ -81,11 +80,6 @@
explicit AudioEncoderPcmA(const Config& config)
: AudioEncoderPcm(config, kSampleRateHz) {}
explicit AudioEncoderPcmA(const CodecInst& codec_inst);
- AudioEncoderPcmA(int payload_type, const SdpAudioFormat& format);
-
- static constexpr const char* GetPayloadName() { return "PCMA"; }
- static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
- const SdpAudioFormat& format);
protected:
size_t EncodeCall(const int16_t* audio,
@@ -110,11 +104,6 @@
explicit AudioEncoderPcmU(const Config& config)
: AudioEncoderPcm(config, kSampleRateHz) {}
explicit AudioEncoderPcmU(const CodecInst& codec_inst);
- AudioEncoderPcmU(int payload_type, const SdpAudioFormat& format);
-
- static constexpr const char* GetPayloadName() { return "PCMU"; }
- static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
- const SdpAudioFormat& format);
protected:
size_t EncodeCall(const int16_t* audio,
diff --git a/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
index f936e81..4c3e82d 100644
--- a/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
+++ b/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
@@ -17,7 +17,6 @@
#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/rtc_base/string_to_number.h"
namespace webrtc {
@@ -34,27 +33,6 @@
} // 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.IsOk() ? rtc::Optional<AudioEncoderG722Config>(config)
- : rtc::Optional<AudioEncoderG722Config>();
-}
-
AudioEncoderG722Impl::AudioEncoderG722Impl(const AudioEncoderG722Config& config,
int payload_type)
: num_channels_(config.num_channels),
@@ -78,26 +56,8 @@
AudioEncoderG722Impl::AudioEncoderG722Impl(const CodecInst& codec_inst)
: AudioEncoderG722Impl(CreateConfig(codec_inst), codec_inst.pltype) {}
-AudioEncoderG722Impl::AudioEncoderG722Impl(int payload_type,
- const SdpAudioFormat& format)
- : AudioEncoderG722Impl(*SdpToConfig(format), payload_type) {}
-
AudioEncoderG722Impl::~AudioEncoderG722Impl() = default;
-rtc::Optional<AudioCodecInfo> AudioEncoderG722Impl::QueryAudioEncoder(
- const SdpAudioFormat& format) {
- if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0) {
- 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),
- rtc::dchecked_cast<size_t>(config_opt->num_channels), 64000});
- }
- }
- return rtc::Optional<AudioCodecInfo>();
-}
-
int AudioEncoderG722Impl::SampleRateHz() const {
return kSampleRateHz;
}
diff --git a/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/modules/audio_coding/codecs/g722/audio_encoder_g722.h
index 00d1c0f..b4d9ef0 100644
--- a/modules/audio_coding/codecs/g722/audio_encoder_g722.h
+++ b/modules/audio_coding/codecs/g722/audio_encoder_g722.h
@@ -14,7 +14,6 @@
#include <memory>
#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/audio_codecs/g722/audio_encoder_g722_config.h"
#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
#include "webrtc/rtc_base/buffer.h"
@@ -26,18 +25,10 @@
class AudioEncoderG722Impl final : public AudioEncoder {
public:
- static rtc::Optional<AudioEncoderG722Config> SdpToConfig(
- const SdpAudioFormat& format);
-
AudioEncoderG722Impl(const AudioEncoderG722Config& config, int payload_type);
explicit AudioEncoderG722Impl(const CodecInst& codec_inst);
- AudioEncoderG722Impl(int payload_type, const SdpAudioFormat& format);
~AudioEncoderG722Impl() override;
- static constexpr const char* GetPayloadName() { return "G722"; }
- static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
- const SdpAudioFormat& format);
-
int SampleRateHz() const override;
size_t NumChannels() const override;
int RtpTimestampRateHz() const override;
diff --git a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
index 5a0090a..2a6dda7 100644
--- a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
+++ b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
@@ -16,7 +16,6 @@
#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/rtc_base/string_to_number.h"
namespace webrtc {
@@ -47,26 +46,6 @@
} // namespace
-rtc::Optional<AudioEncoderIlbcConfig> AudioEncoderIlbcImpl::SdpToConfig(
- const SdpAudioFormat& format) {
- if (STR_CASE_CMP(format.name.c_str(), "ilbc") != 0 ||
- format.clockrate_hz != 8000 || format.num_channels != 1) {
- return rtc::Optional<AudioEncoderIlbcConfig>();
- }
-
- AudioEncoderIlbcConfig config;
- 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(20, std::min(whole_packets * 10, 60));
- }
- }
- return config.IsOk() ? rtc::Optional<AudioEncoderIlbcConfig>(config)
- : rtc::Optional<AudioEncoderIlbcConfig>();
-}
-
AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config,
int payload_type)
: frame_size_ms_(config.frame_size_ms),
@@ -81,29 +60,10 @@
AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const CodecInst& codec_inst)
: AudioEncoderIlbcImpl(CreateConfig(codec_inst), codec_inst.pltype) {}
-AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(int payload_type,
- const SdpAudioFormat& format)
- : AudioEncoderIlbcImpl(*SdpToConfig(format), payload_type) {}
-
AudioEncoderIlbcImpl::~AudioEncoderIlbcImpl() {
RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
}
-rtc::Optional<AudioCodecInfo> AudioEncoderIlbcImpl::QueryAudioEncoder(
- const SdpAudioFormat& format) {
- if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0) {
- const auto config_opt = SdpToConfig(format);
- if (format.clockrate_hz == 8000 && format.num_channels == 1 &&
- config_opt) {
- RTC_DCHECK(config_opt->IsOk());
- return rtc::Optional<AudioCodecInfo>(
- {rtc::dchecked_cast<int>(kSampleRateHz), 1,
- GetIlbcBitrate(config_opt->frame_size_ms)});
- }
- }
- return rtc::Optional<AudioCodecInfo>();
-}
-
int AudioEncoderIlbcImpl::SampleRateHz() const {
return kSampleRateHz;
}
diff --git a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
index c83be61..6a80a69 100644
--- a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
+++ b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
@@ -12,7 +12,6 @@
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
#include "webrtc/rtc_base/constructormagic.h"
@@ -23,18 +22,10 @@
class AudioEncoderIlbcImpl final : public AudioEncoder {
public:
- static rtc::Optional<AudioEncoderIlbcConfig> SdpToConfig(
- const SdpAudioFormat& format);
-
AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config, int payload_type);
explicit AudioEncoderIlbcImpl(const CodecInst& codec_inst);
- AudioEncoderIlbcImpl(int payload_type, const SdpAudioFormat& format);
~AudioEncoderIlbcImpl() override;
- static constexpr const char* GetPayloadName() { return "ILBC"; }
- static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
- const SdpAudioFormat& format);
-
int SampleRateHz() const override;
size_t NumChannels() const override;
size_t Num10MsFramesInNextPacket() const override;
diff --git a/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index 2cc23db..c12d734 100644
--- a/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -14,7 +14,6 @@
#include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h"
@@ -56,13 +55,8 @@
explicit AudioEncoderIsacT(
const CodecInst& codec_inst,
const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo);
- AudioEncoderIsacT(int payload_type, const SdpAudioFormat& format);
~AudioEncoderIsacT() override;
- static constexpr const char* GetPayloadName() { return "ISAC"; }
- static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
- const SdpAudioFormat& format);
-
int SampleRateHz() const override;
size_t NumChannels() const override;
size_t Num10MsFramesInNextPacket() const override;
diff --git a/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index bda379c..854f2ee 100644
--- a/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -13,14 +13,8 @@
#include "webrtc/common_types.h"
#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/string_to_number.h"
namespace webrtc {
-namespace { // NOLINT (not a "regular" header file)
-int GetIsacMaxBitrate(int clockrate_hz) {
- return (clockrate_hz == 32000) ? 56000 : 32000;
-}
-} // namespace
template <typename T>
typename AudioEncoderIsacT<T>::Config CreateIsacConfig(
@@ -39,33 +33,6 @@
}
template <typename T>
-typename AudioEncoderIsacT<T>::Config CreateIsacConfig(
- int payload_type,
- const SdpAudioFormat& format) {
- typename AudioEncoderIsacT<T>::Config config;
- config.payload_type = payload_type;
- config.sample_rate_hz = format.clockrate_hz;
-
- // We only support different frame sizes at 16000 Hz.
- if (config.sample_rate_hz == 16000) {
- auto ptime_iter = format.parameters.find("ptime");
- if (ptime_iter != format.parameters.end()) {
- auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
- if (ptime && *ptime >= 60) {
- config.frame_size_ms = 60;
- } else {
- config.frame_size_ms = 30;
- }
- }
- }
-
- // Set the default bitrate for ISAC to the maximum bitrate allowed at this
- // clockrate. At this point, adaptive mode is not used by WebRTC.
- config.bit_rate = GetIsacMaxBitrate(format.clockrate_hz);
- return config;
-}
-
-template <typename T>
bool AudioEncoderIsacT<T>::Config::IsOk() const {
if (max_bit_rate < 32000 && max_bit_rate != -1)
return false;
@@ -106,25 +73,6 @@
: AudioEncoderIsacT(CreateIsacConfig<T>(codec_inst, bwinfo)) {}
template <typename T>
-AudioEncoderIsacT<T>::AudioEncoderIsacT(int payload_type,
- const SdpAudioFormat& format)
- : AudioEncoderIsacT(CreateIsacConfig<T>(payload_type, format)) {}
-
-template <typename T>
-rtc::Optional<AudioCodecInfo> AudioEncoderIsacT<T>::QueryAudioEncoder(
- const SdpAudioFormat& format) {
- if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0) {
- Config config = CreateIsacConfig<T>(0, format);
- if (config.IsOk()) {
- return rtc::Optional<AudioCodecInfo>(
- {config.sample_rate_hz, 1, config.bit_rate, 10000,
- GetIsacMaxBitrate(format.clockrate_hz)});
- }
- }
- return rtc::Optional<AudioCodecInfo>();
-}
-
-template <typename T>
AudioEncoderIsacT<T>::~AudioEncoderIsacT() {
RTC_CHECK_EQ(0, T::Free(isac_state_));
}
diff --git a/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc b/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
index 897eed2..7b4a919 100644
--- a/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
+++ b/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
@@ -16,7 +16,6 @@
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/rtc_base/string_to_number.h"
namespace webrtc {
@@ -35,6 +34,7 @@
}
namespace {
+
AudioEncoderPcm16B::Config CreateConfig(const CodecInst& codec_inst) {
AudioEncoderPcm16B::Config config;
config.num_channels = codec_inst.channels;
@@ -45,23 +45,6 @@
return config;
}
-AudioEncoderPcm16B::Config CreateConfig(int payload_type,
- const SdpAudioFormat& format) {
- AudioEncoderPcm16B::Config config;
- config.num_channels = format.num_channels;
- config.sample_rate_hz = format.clockrate_hz;
- config.frame_size_ms = 10;
- 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));
- }
- }
- config.payload_type = payload_type;
- return config;
-}
} // namespace
bool AudioEncoderPcm16B::Config::IsOk() const {
@@ -74,24 +57,4 @@
AudioEncoderPcm16B::AudioEncoderPcm16B(const CodecInst& codec_inst)
: AudioEncoderPcm16B(CreateConfig(codec_inst)) {}
-AudioEncoderPcm16B::AudioEncoderPcm16B(int payload_type,
- const SdpAudioFormat& format)
- : AudioEncoderPcm16B(CreateConfig(payload_type, format)) {}
-
-rtc::Optional<AudioCodecInfo> AudioEncoderPcm16B::QueryAudioEncoder(
- const SdpAudioFormat& format) {
- if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0 &&
- format.num_channels >= 1) {
- Config config = CreateConfig(0, format);
- if (config.IsOk()) {
- constexpr int bits_per_sample = 16;
- return rtc::Optional<AudioCodecInfo>(
- {config.sample_rate_hz, config.num_channels,
- config.sample_rate_hz * bits_per_sample *
- rtc::dchecked_cast<int>(config.num_channels)});
- }
- }
- return rtc::Optional<AudioCodecInfo>();
-}
-
} // namespace webrtc
diff --git a/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h b/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
index 79f56c9..25d548c 100644
--- a/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
+++ b/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
@@ -31,11 +31,6 @@
explicit AudioEncoderPcm16B(const Config& config)
: AudioEncoderPcm(config, config.sample_rate_hz) {}
explicit AudioEncoderPcm16B(const CodecInst& codec_inst);
- AudioEncoderPcm16B(int payload_type, const SdpAudioFormat& format);
-
- static constexpr const char* GetPayloadName() { return "L16"; }
- static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
- const SdpAudioFormat& format);
protected:
size_t EncodeCall(const int16_t* audio,
