AcmReceiver: Eliminate AcmReceiver::decoders_
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2351183002
Cr-Commit-Position: refs/heads/master@{#14335}
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index 85cbd8a..afd1ff4 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -180,9 +180,12 @@
int32_t AcmReceiver::AddCodec(int acm_codec_id,
uint8_t payload_type,
size_t channels,
- int sample_rate_hz,
+ int /*sample_rate_hz*/,
AudioDecoder* audio_decoder,
const std::string& name) {
+ // TODO(kwiberg): This function has been ignoring the |sample_rate_hz|
+ // argument for a long time. Arguably, it should simply be removed.
+
const auto neteq_decoder = [acm_codec_id, channels]() -> NetEqDecoder {
if (acm_codec_id == -1)
return NetEqDecoder::kDecoderArbitrary; // External decoder.
@@ -194,29 +197,22 @@
RTC_DCHECK(ned) << "Invalid codec ID: " << static_cast<int>(*cid);
return *ned;
}();
+ const rtc::Optional<SdpAudioFormat> new_format =
+ RentACodec::NetEqDecoderToSdpAudioFormat(neteq_decoder);
rtc::CritScope lock(&crit_sect_);
- // The corresponding NetEq decoder ID.
- // If this codec has been registered before.
- auto it = decoders_.find(payload_type);
- if (it != decoders_.end()) {
- const Decoder& decoder = it->second;
- if (acm_codec_id != -1 && decoder.acm_codec_id == acm_codec_id &&
- decoder.channels == channels &&
- decoder.sample_rate_hz == sample_rate_hz) {
- // Re-registering the same codec. Do nothing and return.
- return 0;
- }
+ const SdpAudioFormat* const old_format =
+ neteq_->GetDecoderFormat(payload_type);
+ if (old_format && new_format && *old_format == *new_format) {
+ // Re-registering the same codec. Do nothing and return.
+ return 0;
+ }
- // Changing codec. First unregister the old codec, then register the new
- // one.
- if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
- LOG(LERROR) << "Cannot remove payload " << static_cast<int>(payload_type);
- return -1;
- }
-
- decoders_.erase(it);
+ if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK &&
+ neteq_->LastError() != NetEq::kDecoderNotFound) {
+ LOG(LERROR) << "Cannot remove payload " << static_cast<int>(payload_type);
+ return -1;
}
int ret_val;
@@ -232,13 +228,6 @@
<< " channels: " << channels;
return -1;
}
-
- Decoder decoder;
- decoder.acm_codec_id = acm_codec_id;
- decoder.payload_type = payload_type;
- decoder.channels = channels;
- decoder.sample_rate_hz = sample_rate_hz;
- decoders_[payload_type] = decoder;
return 0;
}
@@ -249,18 +238,14 @@
void AcmReceiver::RemoveAllCodecs() {
rtc::CritScope lock(&crit_sect_);
neteq_->RemoveAllPayloadTypes();
- decoders_.clear();
last_audio_decoder_ = rtc::Optional<CodecInst>();
last_packet_sample_rate_hz_ = rtc::Optional<int>();
}
int AcmReceiver::RemoveCodec(uint8_t payload_type) {
rtc::CritScope lock(&crit_sect_);
- auto it = decoders_.find(payload_type);
- if (it == decoders_.end()) { // Such a payload-type is not registered.
- return 0;
- }
- if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
+ if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK &&
+ neteq_->LastError() != NetEq::kDecoderNotFound) {
LOG(LERROR) << "AcmReceiver::RemoveCodec" << static_cast<int>(payload_type);
return -1;
}
@@ -268,7 +253,6 @@
last_audio_decoder_ = rtc::Optional<CodecInst>();
last_packet_sample_rate_hz_ = rtc::Optional<int>();
}
- decoders_.erase(it);
return 0;
}
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index 53def2b..94f15a2 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -274,8 +274,6 @@
std::unique_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
CallStatistics call_stats_ GUARDED_BY(crit_sect_);
NetEq* neteq_;
- // Decoders map is keyed by payload type
- std::map<uint8_t, Decoder> decoders_ GUARDED_BY(crit_sect_);
Clock* clock_; // TODO(henrik.lundin) Make const if possible.
bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_);
diff --git a/webrtc/modules/audio_coding/codecs/audio_format.cc b/webrtc/modules/audio_coding/codecs/audio_format.cc
index 86d5d80..ebd7cb0 100644
--- a/webrtc/modules/audio_coding/codecs/audio_format.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_format.cc
@@ -10,6 +10,8 @@
#include "webrtc/modules/audio_coding/codecs/audio_format.h"
+#include "webrtc/common_types.h"
+
namespace webrtc {
SdpAudioFormat::SdpAudioFormat(const SdpAudioFormat&) = default;
@@ -33,6 +35,12 @@
SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default;
SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default;
+bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b) {
+ return STR_CASE_CMP(a.name.c_str(), b.name.c_str()) == 0 &&
+ a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels &&
+ a.parameters == b.parameters;
+}
+
void swap(SdpAudioFormat& a, SdpAudioFormat& b) {
using std::swap;
swap(a.name, b.name);
diff --git a/webrtc/modules/audio_coding/codecs/audio_format.h b/webrtc/modules/audio_coding/codecs/audio_format.h
index 43f82dc..1199cc2 100644
--- a/webrtc/modules/audio_coding/codecs/audio_format.h
+++ b/webrtc/modules/audio_coding/codecs/audio_format.h
@@ -35,6 +35,11 @@
SdpAudioFormat& operator=(const SdpAudioFormat&);
SdpAudioFormat& operator=(SdpAudioFormat&&);
+ friend bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b);
+ friend bool operator!=(const SdpAudioFormat& a, const SdpAudioFormat& b) {
+ return !(a == b);
+ }
+
std::string name;
int clockrate_hz;
int num_channels;
diff --git a/webrtc/modules/audio_coding/neteq/decoder_database.h b/webrtc/modules/audio_coding/neteq/decoder_database.h
index 3728d1d..296d059 100644
--- a/webrtc/modules/audio_coding/neteq/decoder_database.h
+++ b/webrtc/modules/audio_coding/neteq/decoder_database.h
@@ -64,9 +64,8 @@
return decoder ? decoder->SampleRateHz() : cng_decoder_->sample_rate_hz;
}
- const SdpAudioFormat& GetFormat() const {
- RTC_DCHECK(audio_format_);
- return *audio_format_;
+ const SdpAudioFormat* GetFormat() const {
+ return audio_format_ ? &*audio_format_ : nullptr;
}
// Returns true if |codec_type| is comfort noise.
diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
index 952ab23..98bb37c 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -259,6 +259,11 @@
// value if we have no decoder for that payload type.
virtual rtc::Optional<CodecInst> GetDecoder(int payload_type) const = 0;
+ // Returns the decoder format for the given payload type. Returns null if no
+ // such payload type was registered, or if it was registered without
+ // providing an SdpAudioFormat.
+ virtual const SdpAudioFormat* GetDecoderFormat(int payload_type) const = 0;
+
// Not implemented.
virtual int SetTargetNumberOfChannels() = 0;
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 98588f4..221b07c 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -457,6 +457,18 @@
return rtc::Optional<CodecInst>(ci);
}
+const SdpAudioFormat* NetEqImpl::GetDecoderFormat(int payload_type) const {
+ rtc::CritScope lock(&crit_sect_);
+ const DecoderDatabase::DecoderInfo* const di =
+ decoder_database_->GetDecoderInfo(payload_type);
+ if (!di) {
+ return nullptr; // Payload type not registered.
+ }
+ // This will return null if the payload type was registered without an
+ // SdpAudioFormat.
+ return di->GetFormat();
+}
+
int NetEqImpl::SetTargetNumberOfChannels() {
return kNotImplemented;
}
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index 7903ba6..dd35301 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -173,6 +173,8 @@
rtc::Optional<CodecInst> GetDecoder(int payload_type) const override;
+ const SdpAudioFormat* GetDecoderFormat(int payload_type) const override;
+
int SetTargetNumberOfChannels() override;
int SetTargetSampleRate() override;
diff --git a/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc b/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
index 1f28639..fc3a846 100644
--- a/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
+++ b/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
@@ -50,7 +50,7 @@
// support timestamp scaling of them.
denominator_ = numerator_;
} else {
- denominator_ = info->GetFormat().clockrate_hz;
+ denominator_ = info->GetFormat()->clockrate_hz;
}
}
if (numerator_ != denominator_) {