Enable 8 bit and float pcm record formats for AudioFlinger
Update sampling rate handling as well.
Bug: 19570772
Change-Id: I872248e64c0578b2e48869a68fee0d51bd0640c3
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index f3206cb..5002099 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -45,6 +45,8 @@
#include "AudioFlinger.h"
#include "ServiceUtilities.h"
+#include <media/AudioResamplerPublic.h>
+
#include <media/EffectsFactoryApi.h>
#include <audio_effects/effect_visualizer.h>
#include <audio_effects/effect_ns.h>
@@ -1140,19 +1142,46 @@
if (ret != NO_ERROR) {
return 0;
}
+ if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
+ return 0;
+ }
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
- audio_config_t config;
- memset(&config, 0, sizeof(config));
- config.sample_rate = sampleRate;
- config.channel_mask = channelMask;
- config.format = format;
+ audio_config_t config, proposed;
+ memset(&proposed, 0, sizeof(proposed));
+ proposed.sample_rate = sampleRate;
+ proposed.channel_mask = channelMask;
+ proposed.format = format;
audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
- size_t size = dev->get_input_buffer_size(dev, &config);
+ size_t frames;
+ for (;;) {
+ // Note: config is currently a const parameter for get_input_buffer_size()
+ // but we use a copy from proposed in case config changes from the call.
+ config = proposed;
+ frames = dev->get_input_buffer_size(dev, &config);
+ if (frames != 0) {
+ break; // hal success, config is the result
+ }
+ // change one parameter of the configuration each iteration to a more "common" value
+ // to see if the device will support it.
+ if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) {
+ proposed.format = AUDIO_FORMAT_PCM_16_BIT;
+ } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as
+ proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw?
+ } else {
+ ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, "
+ "format %#x, channelMask 0x%X",
+ sampleRate, format, channelMask);
+ break; // retries failed, break out of loop with frames == 0.
+ }
+ }
mHardwareStatus = AUDIO_HW_IDLE;
- return size;
+ if (frames > 0 && config.sample_rate != sampleRate) {
+ frames = destinationFramesPossible(frames, sampleRate, config.sample_rate);
+ }
+ return frames; // may be converted to bytes at the Java level.
}
uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
@@ -1419,9 +1448,8 @@
goto Exit;
}
- // we don't yet support anything other than 16-bit PCM
- if (!(audio_is_valid_format(format) &&
- audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) {
+ // we don't yet support anything other than linear PCM
+ if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
ALOGE("openRecord() invalid format %#x", format);
lStatus = BAD_VALUE;
goto Exit;
@@ -2002,11 +2030,11 @@
status, address.string());
// If the input could not be opened with the requested parameters and we can handle the
- // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
- // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
+ // conversion internally, try to open again with the proposed parameters.
if (status == BAD_VALUE &&
- config->format == halconfig.format && halconfig.format == AUDIO_FORMAT_PCM_16_BIT &&
- (halconfig.sample_rate <= 2 * config->sample_rate) &&
+ audio_is_linear_pcm(config->format) &&
+ audio_is_linear_pcm(halconfig.format) &&
+ (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
(audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) &&
(audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) {
// FIXME describe the change proposed by HAL (save old values so we can log them here)