#include "webrtc/voice_engine/utility.h"
+#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-
-namespace webrtc
-{
-
-namespace voe
-{
-enum{kMaxTargetLen = 2*32*10}; // stereo 32KHz 10ms
-
-void Utility::MixWithSat(int16_t target[],
- int target_channel,
- const int16_t source[],
- int source_channel,
- int source_len)
-{
- assert((target_channel == 1) || (target_channel == 2));
- assert((source_channel == 1) || (source_channel == 2));
- assert(source_len <= kMaxTargetLen);
-
- if ((target_channel == 2) && (source_channel == 1))
- {
- // Convert source from mono to stereo.
- int32_t left = 0;
- int32_t right = 0;
- for (int i = 0; i < source_len; ++i) {
- left = source[i] + target[i*2];
- right = source[i] + target[i*2 + 1];
- target[i*2] = WebRtcSpl_SatW32ToW16(left);
- target[i*2 + 1] = WebRtcSpl_SatW32ToW16(right);
- }
- }
- else if ((target_channel == 1) && (source_channel == 2))
- {
- // Convert source from stereo to mono.
- int32_t temp = 0;
- for (int i = 0; i < source_len/2; ++i) {
- temp = ((source[i*2] + source[i*2 + 1])>>1) + target[i];
- target[i] = WebRtcSpl_SatW32ToW16(temp);
- }
- }
- else
- {
- int32_t temp = 0;
- for (int i = 0; i < source_len; ++i) {
- temp = source[i] + target[i];
- target[i] = WebRtcSpl_SatW32ToW16(temp);
- }
- }
-}
+#include "webrtc/common_types.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/voice_engine/voice_engine_defines.h"
-void Utility::MixSubtractWithSat(int16_t target[],
- const int16_t source[],
- uint16_t len)
-{
- int32_t temp(0);
- for (int i = 0; i < len; i++)
- {
- temp = target[i] - source[i];
- if (temp > 32767)
- target[i] = 32767;
- else if (temp < -32768)
- target[i] = -32768;
- else
- target[i] = (int16_t) temp;
- }
-}
+namespace webrtc {
+namespace voe {
-void Utility::MixAndScaleWithSat(int16_t target[],
- const int16_t source[], float scale,
- uint16_t len)
-{
- int32_t temp(0);
- for (int i = 0; i < len; i++)
- {
- temp = (int32_t) (target[i] + scale * source[i]);
- if (temp > 32767)
- target[i] = 32767;
- else if (temp < -32768)
- target[i] = -32768;
- else
- target[i] = (int16_t) temp;
- }
+// TODO(ajm): There is significant overlap between RemixAndResample and
+// ConvertToCodecFormat, but if we're to consolidate we should probably make a
+// real converter class.
+void RemixAndResample(const AudioFrame& src_frame,
+ PushResampler<int16_t>* resampler,
+ AudioFrame* dst_frame) {
+ const int16_t* audio_ptr = src_frame.data_;
+ int audio_ptr_num_channels = src_frame.num_channels_;
+ int16_t mono_audio[AudioFrame::kMaxDataSizeSamples];
+
+ // Downmix before resampling.
+ if (src_frame.num_channels_ == 2 && dst_frame->num_channels_ == 1) {
+ AudioFrameOperations::StereoToMono(src_frame.data_,
+ src_frame.samples_per_channel_,
+ mono_audio);
+ audio_ptr = mono_audio;
+ audio_ptr_num_channels = 1;
+ }
+
+ if (resampler->InitializeIfNeeded(src_frame.sample_rate_hz_,
+ dst_frame->sample_rate_hz_,
+ audio_ptr_num_channels) == -1) {
+ dst_frame->CopyFrom(src_frame);
+ LOG_FERR3(LS_ERROR, InitializeIfNeeded, src_frame.sample_rate_hz_,
+ dst_frame->sample_rate_hz_, audio_ptr_num_channels);
+ assert(false);
+ }
+
+ const int src_length = src_frame.samples_per_channel_ *
+ audio_ptr_num_channels;
+ int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
+ AudioFrame::kMaxDataSizeSamples);
+ if (out_length == -1) {
+ dst_frame->CopyFrom(src_frame);
+ LOG_FERR3(LS_ERROR, Resample, audio_ptr, src_length, dst_frame->data_);
+ assert(false);
+ }
+ dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
+
+ // Upmix after resampling.
+ if (src_frame.num_channels_ == 1 && dst_frame->num_channels_ == 2) {
+ // The audio in dst_frame really is mono at this point; MonoToStereo will
+ // set this back to stereo.
+ dst_frame->num_channels_ = 1;
+ AudioFrameOperations::MonoToStereo(dst_frame);
+ }
}
-void Utility::Scale(int16_t vector[], float scale, uint16_t len)
-{
- for (int i = 0; i < len; i++)
- {
- vector[i] = (int16_t) (scale * vector[i]);
- }
+void DownConvertToCodecFormat(const int16_t* src_data,
+ int samples_per_channel,
+ int num_channels,
+ int sample_rate_hz,
+ int codec_num_channels,
+ int codec_rate_hz,
+ int16_t* mono_buffer,
+ PushResampler<int16_t>* resampler,
+ AudioFrame* dst_af) {
+ assert(samples_per_channel <= kMaxMonoDataSizeSamples);
+ assert(num_channels == 1 || num_channels == 2);
+ assert(codec_num_channels == 1 || codec_num_channels == 2);
+
+ // Never upsample the capture signal here. This should be done at the
+ // end of the send chain.
+ int destination_rate = std::min(codec_rate_hz, sample_rate_hz);
+
+ // If no stereo codecs are in use, we downmix a stereo stream from the
+ // device early in the chain, before resampling.
+ if (num_channels == 2 && codec_num_channels == 1) {
+ AudioFrameOperations::StereoToMono(src_data, samples_per_channel,
+ mono_buffer);
+ src_data = mono_buffer;
+ num_channels = 1;
+ }
+
+ if (resampler->InitializeIfNeeded(
+ sample_rate_hz, destination_rate, num_channels) != 0) {
+ LOG_FERR3(LS_ERROR,
+ InitializeIfNeeded,
+ sample_rate_hz,
+ destination_rate,
+ num_channels);
+ assert(false);
+ }
+
+ const int in_length = samples_per_channel * num_channels;
+ int out_length = resampler->Resample(
+ src_data, in_length, dst_af->data_, AudioFrame::kMaxDataSizeSamples);
+ if (out_length == -1) {
+ LOG_FERR3(LS_ERROR, Resample, src_data, in_length, dst_af->data_);
+ assert(false);
+ }
+
+ dst_af->samples_per_channel_ = out_length / num_channels;
+ dst_af->sample_rate_hz_ = destination_rate;
+ dst_af->num_channels_ = num_channels;
+ dst_af->timestamp_ = -1;
+ dst_af->speech_type_ = AudioFrame::kNormalSpeech;
+ dst_af->vad_activity_ = AudioFrame::kVadUnknown;
}
-void Utility::ScaleWithSat(int16_t vector[], float scale,
- uint16_t len)
-{
- int32_t temp(0);
- for (int i = 0; i < len; i++)
- {
- temp = (int32_t) (scale * vector[i]);
- if (temp > 32767)
- vector[i] = 32767;
- else if (temp < -32768)
- vector[i] = -32768;
- else
- vector[i] = (int16_t) temp;
+void MixWithSat(int16_t target[],
+ int target_channel,
+ const int16_t source[],
+ int source_channel,
+ int source_len) {
+ assert(target_channel == 1 || target_channel == 2);
+ assert(source_channel == 1 || source_channel == 2);
+
+ if (target_channel == 2 && source_channel == 1) {
+ // Convert source from mono to stereo.
+ int32_t left = 0;
+ int32_t right = 0;
+ for (int i = 0; i < source_len; ++i) {
+ left = source[i] + target[i * 2];
+ right = source[i] + target[i * 2 + 1];
+ target[i * 2] = WebRtcSpl_SatW32ToW16(left);
+ target[i * 2 + 1] = WebRtcSpl_SatW32ToW16(right);
+ }
+ } else if (target_channel == 1 && source_channel == 2) {
+ // Convert source from stereo to mono.
+ int32_t temp = 0;
+ for (int i = 0; i < source_len / 2; ++i) {
+ temp = ((source[i * 2] + source[i * 2 + 1]) >> 1) + target[i];
+ target[i] = WebRtcSpl_SatW32ToW16(temp);
}
+ } else {
+ int32_t temp = 0;
+ for (int i = 0; i < source_len; ++i) {
+ temp = source[i] + target[i];
+ target[i] = WebRtcSpl_SatW32ToW16(temp);
+ }
+ }
}
} // namespace voe
-
} // namespace webrtc
projects / platform / framework / web / crosswalk.git / blobdiff